From 825b22e5961a47ba7387d3ee6729956d6bb3b495 Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Thu, 7 Mar 2019 11:34:53 +0900 Subject: ASoC: rsnd: src: fix compiler warnings compiler complains about following declarations sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern1[] = { ^~~~~ sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern2[] = { ^~~~~ sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsisr_table[] = { ^~~~~ sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan288888[] = { ^~~~~ sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan244888[] = { ^~~~~ sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan222222[] = { ^~~~~ This patch moves the 'static' keyword to the front of the declaration to fix the compiler warnings Fixes: linux-next commit 7674bec4fc09 ("ASoC: rsnd: update BSDSR/BSDISR handling") Signed-off-by: Jiada Wang Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index db81e066b92e..79716725f67b 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -135,7 +135,7 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, return rate; } -const static u32 bsdsr_table_pattern1[] = { +static const u32 bsdsr_table_pattern1[] = { 0x01800000, /* 6 - 1/6 */ 0x01000000, /* 6 - 1/4 */ 0x00c00000, /* 6 - 1/3 */ @@ -144,7 +144,7 @@ const static u32 bsdsr_table_pattern1[] = { 0x00400000, /* 6 - 1 */ }; -const static u32 bsdsr_table_pattern2[] = { +static const u32 bsdsr_table_pattern2[] = { 0x02400000, /* 6 - 1/6 */ 0x01800000, /* 6 - 1/4 */ 0x01200000, /* 6 - 1/3 */ @@ -153,7 +153,7 @@ const static u32 bsdsr_table_pattern2[] = { 0x00600000, /* 6 - 1 */ }; -const static u32 bsisr_table[] = { +static const u32 bsisr_table[] = { 0x00100060, /* 6 - 1/6 */ 0x00100040, /* 6 - 1/4 */ 0x00100030, /* 6 - 1/3 */ @@ -162,7 +162,7 @@ const static u32 bsisr_table[] = { 0x00100020, /* 6 - 1 */ }; -const static u32 chan288888[] = { +static const u32 chan288888[] = { 0x00000006, /* 1 to 2 */ 0x000001fe, /* 1 to 8 */ 0x000001fe, /* 1 to 8 */ @@ -171,7 +171,7 @@ const static u32 chan288888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan244888[] = { +static const u32 chan244888[] = { 0x00000006, /* 1 to 2 */ 0x0000001e, /* 1 to 4 */ 0x0000001e, /* 1 to 4 */ @@ -180,7 +180,7 @@ const static u32 chan244888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan222222[] = { +static const u32 chan222222[] = { 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ -- cgit v1.2.3-59-g8ed1b From 8ca5104715cfd14254ea5aecc390ae583b707607 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 7 Mar 2019 11:11:30 +0100 Subject: ASoC: davinci-mcasp: Fix clang warning without CONFIG_PM Building with clang shows a variable that is only used by the suspend/resume functions but defined outside of their #ifdef block: sound/soc/ti/davinci-mcasp.c:48:12: error: variable 'context_regs' is not needed and will not be emitted We commonly fix these by marking the PM functions as __maybe_unused, but here that would grow the davinci_mcasp structure, so instead add another #ifdef here. Fixes: 1cc0c054f380 ("ASoC: davinci-mcasp: Convert the context save/restore to use array") Signed-off-by: Arnd Bergmann Acked-by: Peter Ujfalusi Reviewed-by: Nathan Chancellor Signed-off-by: Mark Brown --- sound/soc/ti/davinci-mcasp.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index a3a67a8f0f54..9fbc759fdefe 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -45,6 +45,7 @@ #define MCASP_MAX_AFIFO_DEPTH 64 +#ifdef CONFIG_PM static u32 context_regs[] = { DAVINCI_MCASP_TXFMCTL_REG, DAVINCI_MCASP_RXFMCTL_REG, @@ -68,6 +69,7 @@ struct davinci_mcasp_context { u32 *xrsr_regs; /* for serializer configuration */ bool pm_state; }; +#endif struct davinci_mcasp_ruledata { struct davinci_mcasp *mcasp; -- cgit v1.2.3-59-g8ed1b From 686174a0989b1c33698b26523a1e4ccdcc84cf22 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 4 Mar 2019 21:30:52 +0100 Subject: ASoC: qcom: add i2c dependency for SND_SOC_SDM845 We must not select SND_SOC_MAX98927 unless CONFIG_I2C is also enabled: WARNING: unmet direct dependencies detected for SND_SOC_MAX98927 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=n] Selected by [y]: - SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && MFD_CROS_EC [=y] Signed-off-by: Arnd Bergmann Acked-by: Cheng-Yi Chiang Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 75ceb04d8bf0..b1764af858ba 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -98,7 +98,7 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR && MFD_CROS_EC + depends on QCOM_APR && MFD_CROS_EC && I2C select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 -- cgit v1.2.3-59-g8ed1b From 52e7306ca2db565edc1503a74477f3547bc9e408 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 11 Mar 2019 16:26:28 +0100 Subject: ASoC: stm32: i2s: change trigger traces Update traces to log capture/playback stream start/stop. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 47c334de6b09..783b33497435 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -593,7 +593,8 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* Enable i2s */ - dev_dbg(cpu_dai->dev, "start I2S\n"); + dev_dbg(cpu_dai->dev, "start I2S %s\n", + playback_flg ? "playback" : "capture"); cfg1_mask = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN; regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG, @@ -638,6 +639,9 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dev_dbg(cpu_dai->dev, "stop I2S %s\n", + playback_flg ? "playback" : "capture"); + if (playback_flg) regmap_update_bits(i2s->regmap, STM32_I2S_IER_REG, I2S_IER_UDRIE, @@ -654,8 +658,6 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, break; } - dev_dbg(cpu_dai->dev, "stop I2S\n"); - ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG, I2S_CR1_SPE, 0); if (ret < 0) { -- cgit v1.2.3-59-g8ed1b From c2dc8b2c114c22d7a04cf26a7c7d04faaa757143 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 11 Mar 2019 16:26:29 +0100 Subject: ASoC: stm32: i2s: improve channel capabilities handling Use alsa snd_pcm_hw_constraint_single service to manage channels restriction. This provides better status on driver limitations, to the application. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 783b33497435..06bbbef74b3a 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -496,12 +496,6 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, unsigned int fthlv; int ret; - if ((params_channels(params) == 1) && - ((i2s->fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_DSP_A)) { - dev_err(cpu_dai->dev, "Mono mode supported only by DSP_A\n"); - return -EINVAL; - } - switch (format) { case 16: cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_16); @@ -551,6 +545,10 @@ static int stm32_i2s_startup(struct snd_pcm_substream *substream, i2s->substream = substream; spin_unlock_irqrestore(&i2s->irq_lock, flags); + if ((i2s->fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_DSP_A) + snd_pcm_hw_constraint_single(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 2); + ret = clk_prepare_enable(i2s->i2sclk); if (ret < 0) { dev_err(cpu_dai->dev, "Failed to enable clock: %d\n", ret); -- cgit v1.2.3-59-g8ed1b From fef050c88ce1cbec1ab9a0d981b5331a5ba40f15 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 11 Mar 2019 16:26:30 +0100 Subject: ASoC: stm32: i2s: use default dai name Use default DAI name based on dev_name function. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 06bbbef74b3a..c18e068c1a0d 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -179,7 +179,6 @@ enum i2s_datlen { I2S_I2SMOD_DATLEN_32, }; -#define STM32_I2S_DAI_NAME_SIZE 20 #define STM32_I2S_FIFO_SIZE 16 #define STM32_I2S_IS_MASTER(x) ((x)->ms_flg == I2S_MS_MASTER) @@ -202,7 +201,6 @@ enum i2s_datlen { * @phys_addr: I2S registers physical base address * @lock_fd: lock to manage race conditions in full duplex mode * @irq_lock: prevent race condition with IRQ - * @dais_name: DAI name * @mclk_rate: master clock frequency (Hz) * @fmt: DAI protocol * @refcount: keep count of opened streams on I2S @@ -224,7 +222,6 @@ struct stm32_i2s_data { dma_addr_t phys_addr; spinlock_t lock_fd; /* Manage race conditions for full duplex */ spinlock_t irq_lock; /* used to prevent race condition with IRQ */ - char dais_name[STM32_I2S_DAI_NAME_SIZE]; unsigned int mclk_rate; unsigned int fmt; int refcount; @@ -771,12 +768,8 @@ static int stm32_i2s_dais_init(struct platform_device *pdev, if (!dai_ptr) return -ENOMEM; - snprintf(i2s->dais_name, STM32_I2S_DAI_NAME_SIZE, - "%s", dev_name(&pdev->dev)); - dai_ptr->probe = stm32_i2s_dai_probe; dai_ptr->ops = &stm32_i2s_pcm_dai_ops; - dai_ptr->name = i2s->dais_name; dai_ptr->id = 1; stm32_i2s_dai_init(&dai_ptr->playback, "playback"); stm32_i2s_dai_init(&dai_ptr->capture, "capture"); -- cgit v1.2.3-59-g8ed1b From 642aafea8889b712fe8e57aaa706d6c05d295059 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 7 Mar 2019 16:16:08 +0100 Subject: ASoC: ti: remove compat dma probing After running into a link error: sound/soc/ti/edma-pcm.o:(.rodata+0x18): undefined reference to `edma_filter_fn' I checked all users of this, and they have new-style 'dma_slave_map' tables, so none of them should still need it. Removing the associated lines simplifies the code and avoids the build-time dependency on the respective dmaengine drivers. Acked-by: Peter Ujfalusi Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/ti/edma-pcm.c | 5 +---- sound/soc/ti/sdma-pcm.c | 9 +++------ 2 files changed, 4 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/ti/edma-pcm.c b/sound/soc/ti/edma-pcm.c index 59e588abe54b..fdffb801b185 100644 --- a/sound/soc/ti/edma-pcm.c +++ b/sound/soc/ti/edma-pcm.c @@ -23,7 +23,6 @@ #include #include #include -#include #include "edma-pcm.h" @@ -43,14 +42,12 @@ static const struct snd_pcm_hardware edma_pcm_hardware = { static const struct snd_dmaengine_pcm_config edma_dmaengine_pcm_config = { .pcm_hardware = &edma_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = edma_filter_fn, .prealloc_buffer_size = 128 * 1024, }; int edma_pcm_platform_register(struct device *dev) { - return devm_snd_dmaengine_pcm_register(dev, &edma_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_COMPAT); + return devm_snd_dmaengine_pcm_register(dev, &edma_dmaengine_pcm_config, 0); } EXPORT_SYMBOL_GPL(edma_pcm_platform_register); diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c index 21a9c2499d48..a236350beb10 100644 --- a/sound/soc/ti/sdma-pcm.c +++ b/sound/soc/ti/sdma-pcm.c @@ -11,7 +11,6 @@ #include #include #include -#include #include "sdma-pcm.h" @@ -31,7 +30,6 @@ static const struct snd_pcm_hardware sdma_pcm_hardware = { static const struct snd_dmaengine_pcm_config sdma_dmaengine_pcm_config = { .pcm_hardware = &sdma_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = omap_dma_filter_fn, .prealloc_buffer_size = 128 * 1024, }; @@ -39,13 +37,12 @@ int sdma_pcm_platform_register(struct device *dev, char *txdmachan, char *rxdmachan) { struct snd_dmaengine_pcm_config *config; - unsigned int flags = SND_DMAENGINE_PCM_FLAG_COMPAT; + unsigned int flags = 0; /* Standard names for the directions: 'tx' and 'rx' */ if (!txdmachan && !rxdmachan) return devm_snd_dmaengine_pcm_register(dev, - &sdma_dmaengine_pcm_config, - flags); + &sdma_dmaengine_pcm_config, 0); config = devm_kzalloc(dev, sizeof(*config), GFP_KERNEL); if (!config) @@ -65,7 +62,7 @@ int sdma_pcm_platform_register(struct device *dev, config->chan_names[0] = txdmachan; config->chan_names[1] = rxdmachan; - return devm_snd_dmaengine_pcm_register(dev, config, flags); + return devm_snd_dmaengine_pcm_register(dev, config, 0); } EXPORT_SYMBOL_GPL(sdma_pcm_platform_register); -- cgit v1.2.3-59-g8ed1b From c9fbc1c02ec881c8bb037f87efb706d22589fdab Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 7 Mar 2019 11:10:49 +0100 Subject: ASoC: ams-delta: remove duplicate 'const' clang points out that SOC_ENUM_SINGLE_EXT_DECL() contains a 'const' modifier already, so adding another one does not make it more const: sound/soc/ti/ams-delta.c:203:14: error: duplicate 'const' declaration specifier [-Werror,-Wduplicate-decl-specifier] static const SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum, ^ include/sound/soc.h:351:2: note: expanded from macro 'SOC_ENUM_SINGLE_EXT_DECL' const struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts) Signed-off-by: Arnd Bergmann Reviewed-by: Peter Ujfalusi Reviewed-by: Nathan Chancellor Signed-off-by: Mark Brown --- sound/soc/ti/ams-delta.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 4dce494dfbd3..b9611db14c86 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -200,7 +200,7 @@ static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol, return 0; } -static const SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum, +static SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum, ams_delta_audio_mode); static const struct snd_kcontrol_new ams_delta_audio_controls[] = { -- cgit v1.2.3-59-g8ed1b From 7d7b25d05ef1c5a1a9320190e1eeb55534847558 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Mon, 4 Mar 2019 21:30:50 +0100 Subject: ASoC: ti: fix davinci_mcasp_probe dependencies The SND_SOC_DAVINCI_MCASP driver can use either edma or sdma as a back-end, and it takes the presence of the respective dma engine drivers in the configuration as an indication to which ones should be built. However, this is flawed in multiple ways: - With CONFIG_TI_EDMA=m and CONFIG_SND_SOC_DAVINCI_MCASP=y, is enabled as =m, and we get a link error: sound/soc/ti/davinci-mcasp.o: In function `davinci_mcasp_probe': davinci-mcasp.c:(.text+0x930): undefined reference to `edma_pcm_platform_register' - When CONFIG_SND_SOC_DAVINCI_MCASP=m has already been selected by another driver, the same link error appears even if CONFIG_TI_EDMA is disabled There are possibly other issues here, but it seems that the only reasonable solution is to always build both SND_SOC_TI_EDMA_PCM and SND_SOC_TI_SDMA_PCM as a dependency here. Both are fairly small and do not have any other compile-time dependencies, so the cost is very small, and makes the configuration stage much more consistent. Fixes: f2055e145f29 ("ASoC: ti: Merge davinci and omap directories") Signed-off-by: Arnd Bergmann Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/ti/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 4bf3c15d4e51..ee7c202c69b7 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -21,8 +21,8 @@ config SND_SOC_DAVINCI_ASP config SND_SOC_DAVINCI_MCASP tristate "Multichannel Audio Serial Port (McASP) support" - select SND_SOC_TI_EDMA_PCM if TI_EDMA - select SND_SOC_TI_SDMA_PCM if DMA_OMAP + select SND_SOC_TI_EDMA_PCM + select SND_SOC_TI_SDMA_PCM help Say Y or M here if you want to have support for McASP IP found in various Texas Instruments SoCs like: -- cgit v1.2.3-59-g8ed1b From 7bf7d0554890c53478ec04d3d4bfced0c881f6c8 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Thu, 7 Mar 2019 22:44:55 +0100 Subject: ASoC: axi-i2s: let both capture and playback be optional Both the capture and playback channels are optional in the axi_i2s IP block. Reflect this in the driver by enabling only the channel(s) that have a DMA. Signed-off-by: Luca Ceresoli Signed-off-by: Mark Brown --- sound/soc/adi/axi-i2s.c | 68 ++++++++++++++++++++++++++++++++++--------------- 1 file changed, 48 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 4c23381727a1..273c543e8ff3 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -43,6 +43,9 @@ struct axi_i2s { struct clk *clk; struct clk *clk_ref; + bool has_capture; + bool has_playback; + struct snd_soc_dai_driver dai_driver; struct snd_dmaengine_dai_dma_data capture_dma_data; @@ -136,8 +139,10 @@ static int axi_i2s_dai_probe(struct snd_soc_dai *dai) { struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); - snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, - &i2s->capture_dma_data); + snd_soc_dai_init_dma_data( + dai, + i2s->has_playback ? &i2s->playback_dma_data : NULL, + i2s->has_capture ? &i2s->capture_dma_data : NULL); return 0; } @@ -151,18 +156,6 @@ static const struct snd_soc_dai_ops axi_i2s_dai_ops = { static struct snd_soc_dai_driver axi_i2s_dai = { .probe = axi_i2s_dai_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_KNOT, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_KNOT, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE, - }, .ops = &axi_i2s_dai_ops, .symmetric_rates = 1, }; @@ -178,6 +171,19 @@ static const struct regmap_config axi_i2s_regmap_config = { .max_register = AXI_I2S_REG_STATUS, }; +static void axi_i2s_parse_of(struct axi_i2s *i2s, const struct device_node *np) +{ + struct property *dma_names; + const char *dma_name; + + of_property_for_each_string(np, "dma-names", dma_names, dma_name) { + if (strcmp(dma_name, "rx") == 0) + i2s->has_capture = true; + if (strcmp(dma_name, "tx") == 0) + i2s->has_playback = true; + } +} + static int axi_i2s_probe(struct platform_device *pdev) { struct resource *res; @@ -191,6 +197,8 @@ static int axi_i2s_probe(struct platform_device *pdev) platform_set_drvdata(pdev, i2s); + axi_i2s_parse_of(i2s, pdev->dev.of_node); + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(base)) @@ -213,13 +221,29 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) return ret; - i2s->playback_dma_data.addr = res->start + AXI_I2S_REG_TX_FIFO; - i2s->playback_dma_data.addr_width = 4; - i2s->playback_dma_data.maxburst = 1; + if (i2s->has_playback) { + axi_i2s_dai.playback.channels_min = 2; + axi_i2s_dai.playback.channels_max = 2; + axi_i2s_dai.playback.rates = SNDRV_PCM_RATE_KNOT; + axi_i2s_dai.playback.formats = + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE; + + i2s->playback_dma_data.addr = res->start + AXI_I2S_REG_TX_FIFO; + i2s->playback_dma_data.addr_width = 4; + i2s->playback_dma_data.maxburst = 1; + } + + if (i2s->has_capture) { + axi_i2s_dai.capture.channels_min = 2; + axi_i2s_dai.capture.channels_max = 2; + axi_i2s_dai.capture.rates = SNDRV_PCM_RATE_KNOT; + axi_i2s_dai.capture.formats = + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE; - i2s->capture_dma_data.addr = res->start + AXI_I2S_REG_RX_FIFO; - i2s->capture_dma_data.addr_width = 4; - i2s->capture_dma_data.maxburst = 1; + i2s->capture_dma_data.addr = res->start + AXI_I2S_REG_RX_FIFO; + i2s->capture_dma_data.addr_width = 4; + i2s->capture_dma_data.maxburst = 1; + } i2s->ratnum.num = clk_get_rate(i2s->clk_ref) / 2 / AXI_I2S_BITS_PER_FRAME; i2s->ratnum.den_step = 1; @@ -240,6 +264,10 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) goto err_clk_disable; + dev_info(&pdev->dev, "probed, capture %s, playback %s\n", + i2s->has_capture ? "enabled" : "disabled", + i2s->has_playback ? "enabled" : "disabled"); + return 0; err_clk_disable: -- cgit v1.2.3-59-g8ed1b From b517229ca2f7836125be58997808f2803f9ebc86 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Wed, 13 Mar 2019 14:47:18 +0800 Subject: ASoC: nau8810: use 64-bit arithmetic instead of 32-bit Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index bfd74b86c9d2..e7fd0b2645ea 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -505,7 +505,7 @@ static int nau88l0_calc_pll(unsigned int pll_in, f2_max = 0; scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { - f2 = 256 * fs * 4 * nau8810_mclk_scaler[i] / 10; + f2 = 256ULL * fs * 4 * nau8810_mclk_scaler[i] / 10; if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && f2_max < f2) { f2_max = f2; -- cgit v1.2.3-59-g8ed1b From 709a9b8a1f4ff6c34c0a6e6fefb31cfb23b19bf5 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Wed, 13 Mar 2019 15:04:16 +0800 Subject: ASoC: nau8810: fix the typo of function name Correct the typo at the function name. Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8810.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index e7fd0b2645ea..39512a78a556 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -493,7 +493,7 @@ static int nau8810_set_sysclk(struct snd_soc_dai *dai, return 0; } -static int nau88l0_calc_pll(unsigned int pll_in, +static int nau8810_calc_pll(unsigned int pll_in, unsigned int fs, struct nau8810_pll *pll_param) { u64 f2, f2_max, pll_ratio; @@ -542,7 +542,7 @@ static int nau8810_set_pll(struct snd_soc_dai *codec_dai, int pll_id, int ret, fs; fs = freq_out / 256; - ret = nau88l0_calc_pll(freq_in, fs, pll_param); + ret = nau8810_calc_pll(freq_in, fs, pll_param); if (ret < 0) { dev_err(nau8810->dev, "Unsupported input clock %d\n", freq_in); return ret; -- cgit v1.2.3-59-g8ed1b From 20b83421e8eea43e28cb7aeea3e3f865f86aa69d Mon Sep 17 00:00:00 2001 From: John Hsu Date: Wed, 13 Mar 2019 16:52:01 +0800 Subject: ASoC: nau8810: automatic selecting BCLK in I2S master mode The driver will select correct BCLK automatically according to BCLK and FS information in I2S master mode. Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8810.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 39512a78a556..5d300b790f78 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -667,6 +667,24 @@ static int nau8810_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct nau8810 *nau8810 = snd_soc_component_get_drvdata(component); int val_len = 0, val_rate = 0, ret = 0; + unsigned int ctrl_val, bclk_fs, bclk_div; + + /* Select BCLK configuration if the codec as master. */ + regmap_read(nau8810->regmap, NAU8810_REG_CLOCK, &ctrl_val); + if (ctrl_val & NAU8810_CLKIO_MASTER) { + /* get the bclk and fs ratio */ + bclk_fs = snd_soc_params_to_bclk(params) / params_rate(params); + if (bclk_fs <= 32) + bclk_div = NAU8810_BCLKDIV_8; + else if (bclk_fs <= 64) + bclk_div = NAU8810_BCLKDIV_4; + else if (bclk_fs <= 128) + bclk_div = NAU8810_BCLKDIV_2; + else + return -EINVAL; + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_BCLKSEL_MASK, bclk_div); + } switch (params_width(params)) { case 16: -- cgit v1.2.3-59-g8ed1b From 3789debfc379ac84b90e2db32d7a6b66d28a2c7c Mon Sep 17 00:00:00 2001 From: Bard liao Date: Wed, 13 Mar 2019 21:49:43 +0800 Subject: ASoC: topology: create tlv before soc_tplg_init_kcontrol Component driver may want to use tlv data. Create tlv before soc_tplg_init_kcontrol so component driver can use the tlv data in the control_load ops. Signed-off-by: Bard liao Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 25fca7055464..fbee4720b834 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -894,19 +894,20 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, continue; } + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc, &mc->hdr); + /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc, (struct snd_soc_tplg_ctl_hdr *) mc); if (err < 0) { dev_err(tplg->dev, "ASoC: failed to init %s\n", mc->hdr.name); + soc_tplg_free_tlv(tplg, &kc); kfree(sm); continue; } - /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, &mc->hdr); - /* register control here */ err = soc_tplg_add_kcontrol(tplg, &kc, &sm->dobj.control.kcontrol); @@ -1324,18 +1325,19 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( continue; } + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); + /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc[i], (struct snd_soc_tplg_ctl_hdr *)mc); if (err < 0) { dev_err(tplg->dev, "ASoC: failed to init %s\n", mc->hdr.name); + soc_tplg_free_tlv(tplg, &kc[i]); kfree(sm); continue; } - - /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); } return kc; -- cgit v1.2.3-59-g8ed1b From 2944d29d7c89573f9aa47083d105ff0243d5a125 Mon Sep 17 00:00:00 2001 From: Mylène Josserand Date: Mon, 18 Mar 2019 11:39:38 +0100 Subject: ASoC: simple-amplifier: Make gpio property optional MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Some amplifier may not have a GPIO to control the power, but instead simply rely on the regulator to power up and down the amplifier. In order to support those setups, let's make the GPIO optional. Signed-off-by: Mylène Josserand Signed-off-by: Maxime Ripard Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/simple-amplifier.txt | 2 +- sound/soc/codecs/simple-amplifier.c | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt index 7182ac4f1e65..b1b097cc9b68 100644 --- a/Documentation/devicetree/bindings/sound/simple-amplifier.txt +++ b/Documentation/devicetree/bindings/sound/simple-amplifier.txt @@ -2,9 +2,9 @@ Simple Amplifier Audio Driver Required properties: - compatible : "dioo,dio2125" or "simple-audio-amplifier" -- enable-gpios : the gpio connected to the enable pin of the simple amplifier Optional properties: +- enable-gpios : the gpio connected to the enable pin of the simple amplifier - VCC-supply : power supply for the device, as covered in Documentation/devicetree/bindings/regulator/regulator.txt diff --git a/sound/soc/codecs/simple-amplifier.c b/sound/soc/codecs/simple-amplifier.c index c07e8a80b4b7..351aa55c384e 100644 --- a/sound/soc/codecs/simple-amplifier.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -89,7 +89,8 @@ static int simple_amp_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, priv); - priv->gpiod_enable = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); + priv->gpiod_enable = devm_gpiod_get_optional(dev, "enable", + GPIOD_OUT_LOW); if (IS_ERR(priv->gpiod_enable)) { err = PTR_ERR(priv->gpiod_enable); if (err != -EPROBE_DEFER) -- cgit v1.2.3-59-g8ed1b From bbfaa7d36c1eb465f120f2a3dfe25c1fe022195d Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang Date: Thu, 7 Mar 2019 07:51:09 +0800 Subject: ASoC: dpcm: prevent snd_soc_dpcm use after free The dpcm get from fe_clients/be_clients may be free before use Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [] notify_die+0x68/0xb8 [] die+0x118/0x2a8 [] __do_kernel_fault+0x13c/0x14c [] do_translation_fault+0x64/0xa0 [] do_mem_abort+0x4c/0xd0 [] el1_da+0x24/0x40 [] dpcm_be_dai_trigger+0x29c/0x47c [] dpcm_fe_dai_do_trigger+0x94/0x26c [] dpcm_fe_dai_trigger+0x3c/0x44 [] snd_pcm_do_stop+0x50/0x5c [] snd_pcm_action+0xb4/0x13c [] snd_pcm_drop+0xa0/0x128 [] snd_pcm_common_ioctl+0x9d8/0x30f0 [] snd_pcm_ioctl_compat+0x29c/0x2f14 [] compat_SyS_ioctl+0x128/0x244 [] el0_svc_naked+0x34/0x38 [] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 33 ++++++++++++++++++++++++++------- 3 files changed, 29 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index eb7db605955b..1e2be35ed36f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1083,6 +1083,8 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + spinlock_t dpcm_lock; + bool instantiated; bool topology_shortname_created; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 93d316d5bf8e..d05bf9322fa9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2819,6 +2819,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0d5ec68a1e50..42f2e06452b2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1228,8 +1228,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->fe = fe; be->dpcm[stream].runtime = fe->dpcm[stream].runtime; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; + spin_lock(&fe->card->dpcm_lock); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + spin_unlock(&fe->card->dpcm_lock); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, @@ -1294,8 +1296,10 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) #ifdef CONFIG_DEBUG_FS debugfs_remove(dpcm->debugfs_state); #endif + spin_lock(&fe->card->dpcm_lock); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); + spin_unlock(&fe->card->dpcm_lock); kfree(dpcm); } } @@ -1548,9 +1552,11 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + spin_unlock(&fe->card->dpcm_lock); } static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, @@ -2640,11 +2646,13 @@ close: dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } + spin_unlock(&fe->card->dpcm_lock); return ret; } @@ -3221,7 +3229,9 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3230,12 +3240,15 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, state = dpcm->fe->dpcm[stream].state; if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) - return 0; + state == SND_SOC_DPCM_STATE_SUSPEND) { + ret = 0; + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to free/stop this BE DAI */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); @@ -3248,7 +3261,9 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3258,12 +3273,15 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) - return 0; + state == SND_SOC_DPCM_STATE_PREPARE) { + ret = 0; + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to change hw_params */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); @@ -3329,6 +3347,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -3349,7 +3368,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - + spin_unlock(&fe->card->dpcm_lock); out: return offset; } -- cgit v1.2.3-59-g8ed1b From f68c2a682d444296403d1aa6e73b3de0681fdcea Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 6 Mar 2019 17:48:08 +0100 Subject: ASoC: stm32: spdifrx: add power management Add suspend and resume sleep callbacks to STM32 SPDIFRX driver, to support system low power modes. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_spdifrx.c | 42 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 39 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 373df4f24be1..b4c3d983e195 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include @@ -471,6 +472,8 @@ static int stm32_spdifrx_get_ctrl_data(struct stm32_spdifrx_data *spdifrx) memset(spdifrx->cs, 0, SPDIFRX_CS_BYTES_NB); memset(spdifrx->ub, 0, SPDIFRX_UB_BYTES_NB); + pinctrl_pm_select_default_state(&spdifrx->pdev->dev); + ret = stm32_spdifrx_dma_ctrl_start(spdifrx); if (ret < 0) return ret; @@ -502,6 +505,7 @@ static int stm32_spdifrx_get_ctrl_data(struct stm32_spdifrx_data *spdifrx) end: clk_disable_unprepare(spdifrx->kclk); + pinctrl_pm_select_sleep_state(&spdifrx->pdev->dev); return ret; } @@ -611,10 +615,15 @@ static bool stm32_spdifrx_readable_reg(struct device *dev, unsigned int reg) static bool stm32_spdifrx_volatile_reg(struct device *dev, unsigned int reg) { - if (reg == STM32_SPDIFRX_DR) + switch (reg) { + case STM32_SPDIFRX_DR: + case STM32_SPDIFRX_CSR: + case STM32_SPDIFRX_SR: + case STM32_SPDIFRX_DIR: return true; - - return false; + default: + return false; + } } static bool stm32_spdifrx_writeable_reg(struct device *dev, unsigned int reg) @@ -638,6 +647,7 @@ static const struct regmap_config stm32_h7_spdifrx_regmap_conf = { .volatile_reg = stm32_spdifrx_volatile_reg, .writeable_reg = stm32_spdifrx_writeable_reg, .fast_io = true, + .cache_type = REGCACHE_FLAT, }; static irqreturn_t stm32_spdifrx_isr(int irq, void *devid) @@ -983,10 +993,36 @@ static int stm32_spdifrx_remove(struct platform_device *pdev) MODULE_DEVICE_TABLE(of, stm32_spdifrx_ids); +#ifdef CONFIG_PM_SLEEP +static int stm32_spdifrx_suspend(struct device *dev) +{ + struct stm32_spdifrx_data *spdifrx = dev_get_drvdata(dev); + + regcache_cache_only(spdifrx->regmap, true); + regcache_mark_dirty(spdifrx->regmap); + + return 0; +} + +static int stm32_spdifrx_resume(struct device *dev) +{ + struct stm32_spdifrx_data *spdifrx = dev_get_drvdata(dev); + + regcache_cache_only(spdifrx->regmap, false); + + return regcache_sync(spdifrx->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops stm32_spdifrx_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(stm32_spdifrx_suspend, stm32_spdifrx_resume) +}; + static struct platform_driver stm32_spdifrx_driver = { .driver = { .name = "st,stm32-spdifrx", .of_match_table = stm32_spdifrx_ids, + .pm = &stm32_spdifrx_pm_ops, }, .probe = stm32_spdifrx_probe, .remove = stm32_spdifrx_remove, -- cgit v1.2.3-59-g8ed1b From 8a68a509ae6b5d7d18c6bfc88553ca7761029ada Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 5 Mar 2019 12:38:59 +0100 Subject: ASoC: rt5651: Add support for active-high jack detect Some boards use a jack-receptacle with a switch which reports the jack-inserted status as active-high, rather then the standard active-low reporting most jacks use. This commit adds support for it. This is activated by a boolean "realtek,jack-detect-not-inverted" device-property. The not-inverted in the device-property name, rather then active-high, was chosen to keep the device-property naming consistent with the rt5640 codec driver. Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5651.txt | 5 +++ sound/soc/codecs/rt5651.c | 47 ++++++++++++++++++---- sound/soc/codecs/rt5651.h | 1 + 3 files changed, 46 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/rt5651.txt b/Documentation/devicetree/bindings/sound/rt5651.txt index a41199a5cd79..56e736a1cba9 100644 --- a/Documentation/devicetree/bindings/sound/rt5651.txt +++ b/Documentation/devicetree/bindings/sound/rt5651.txt @@ -22,6 +22,11 @@ Optional properties: 2: Use JD1_2 pin for jack-detect 3: Use JD2 pin for jack-detect +- realtek,jack-detect-not-inverted + bool. Normal jack-detect switches give an inverted (active-low) signal, + set this bool in the rare case you've a jack-detect switch which is not + inverted. + - realtek,over-current-threshold-microamp u32, micbias over-current detection threshold in µA, valid values are 600, 1500 and 2000µA. diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 29b2d60076b0..cb8252ff31cb 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1645,7 +1645,10 @@ static bool rt5651_jack_inserted(struct snd_soc_component *component) break; } - return val == 0; + if (rt5651->jd_active_high) + return val != 0; + else + return val == 0; } /* Jack detect and button-press timings */ @@ -1868,20 +1871,47 @@ static void rt5651_enable_jack_detect(struct snd_soc_component *component, case RT5651_JD1_1: snd_soc_component_update_bits(component, RT5651_JD_CTRL2, RT5651_JD_TRG_SEL_MASK, RT5651_JD_TRG_SEL_JD1_1); - snd_soc_component_update_bits(component, RT5651_IRQ_CTRL1, - RT5651_JD1_1_IRQ_EN, RT5651_JD1_1_IRQ_EN); + /* active-low is normal, set inv flag for active-high */ + if (rt5651->jd_active_high) + snd_soc_component_update_bits(component, + RT5651_IRQ_CTRL1, + RT5651_JD1_1_IRQ_EN | RT5651_JD1_1_INV, + RT5651_JD1_1_IRQ_EN | RT5651_JD1_1_INV); + else + snd_soc_component_update_bits(component, + RT5651_IRQ_CTRL1, + RT5651_JD1_1_IRQ_EN | RT5651_JD1_1_INV, + RT5651_JD1_1_IRQ_EN); break; case RT5651_JD1_2: snd_soc_component_update_bits(component, RT5651_JD_CTRL2, RT5651_JD_TRG_SEL_MASK, RT5651_JD_TRG_SEL_JD1_2); - snd_soc_component_update_bits(component, RT5651_IRQ_CTRL1, - RT5651_JD1_2_IRQ_EN, RT5651_JD1_2_IRQ_EN); + /* active-low is normal, set inv flag for active-high */ + if (rt5651->jd_active_high) + snd_soc_component_update_bits(component, + RT5651_IRQ_CTRL1, + RT5651_JD1_2_IRQ_EN | RT5651_JD1_2_INV, + RT5651_JD1_2_IRQ_EN | RT5651_JD1_2_INV); + else + snd_soc_component_update_bits(component, + RT5651_IRQ_CTRL1, + RT5651_JD1_2_IRQ_EN | RT5651_JD1_2_INV, + RT5651_JD1_2_IRQ_EN); break; case RT5651_JD2: snd_soc_component_update_bits(component, RT5651_JD_CTRL2, RT5651_JD_TRG_SEL_MASK, RT5651_JD_TRG_SEL_JD2); - snd_soc_component_update_bits(component, RT5651_IRQ_CTRL1, - RT5651_JD2_IRQ_EN, RT5651_JD2_IRQ_EN); + /* active-low is normal, set inv flag for active-high */ + if (rt5651->jd_active_high) + snd_soc_component_update_bits(component, + RT5651_IRQ_CTRL1, + RT5651_JD2_IRQ_EN | RT5651_JD2_INV, + RT5651_JD2_IRQ_EN | RT5651_JD2_INV); + else + snd_soc_component_update_bits(component, + RT5651_IRQ_CTRL1, + RT5651_JD2_IRQ_EN | RT5651_JD2_INV, + RT5651_JD2_IRQ_EN); break; default: dev_err(component->dev, "Currently only JD1_1 / JD1_2 / JD2 are supported\n"); @@ -1986,6 +2016,9 @@ static void rt5651_apply_properties(struct snd_soc_component *component) "realtek,jack-detect-source", &val) == 0) rt5651->jd_src = val; + if (device_property_read_bool(component->dev, "realtek,jack-detect-not-inverted")) + rt5651->jd_active_high = true; + /* * Testing on various boards has shown that good defaults for the OVCD * threshold and scale-factor are 2000µA and 0.75. For an effective diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h index 41fcb8b5eb40..05b0f6f8b95d 100644 --- a/sound/soc/codecs/rt5651.h +++ b/sound/soc/codecs/rt5651.h @@ -2083,6 +2083,7 @@ struct rt5651_priv { int release_count; int poll_count; unsigned int jd_src; + bool jd_active_high; unsigned int ovcd_th; unsigned int ovcd_sf; -- cgit v1.2.3-59-g8ed1b From a0cb2d4357e483b7bca3c06f790d8f5d4cc20d84 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 5 Mar 2019 12:39:00 +0100 Subject: ASoC: Intel: bytcr_rt5651: Add BYT_RT5651_JD_NOT_INV quirk Add BYT_RT5651_JD_NOT_INV quirk for devices with an inverted (active-high instead of the normal active-low) jack-detect switch. And add a quirk for the Complet Electro Serv MY8307 tablet which has an inverted jack-detect switch (and a mono-speaker). Signed-off-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index b0a4d297176e..4ed59f41ee83 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -79,14 +79,15 @@ enum { #define BYT_RT5651_SSP0_AIF2 BIT(21) #define BYT_RT5651_HP_LR_SWAPPED BIT(22) #define BYT_RT5651_MONO_SPEAKER BIT(23) +#define BYT_RT5651_JD_NOT_INV BIT(24) #define BYT_RT5651_DEFAULT_QUIRKS (BYT_RT5651_MCLK_EN | \ BYT_RT5651_JD1_1 | \ BYT_RT5651_OVCD_TH_2000UA | \ BYT_RT5651_OVCD_SF_0P75) -/* jack-detect-source + dmic-en + ovcd-th + -sf + terminating empty entry */ -#define MAX_NO_PROPS 5 +/* jack-detect-source + inv + dmic-en + ovcd-th + -sf + terminating entry */ +#define MAX_NO_PROPS 6 struct byt_rt5651_private { struct clk *mclk; @@ -137,6 +138,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF2 enabled\n"); if (byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) dev_info(dev, "quirk MONO_SPEAKER enabled\n"); + if (byt_rt5651_quirk & BYT_RT5651_JD_NOT_INV) + dev_info(dev, "quirk JD_NOT_INV enabled\n"); } #define BYT_CODEC_DAI1 "rt5651-aif1" @@ -414,6 +417,18 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_HP_LR_SWAPPED | BYT_RT5651_MONO_SPEAKER), }, + { + /* Complet Electro Serv MY8307 */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Complet Electro Serv"), + DMI_MATCH(DMI_PRODUCT_NAME, "MY8307"), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_MONO_SPEAKER | + BYT_RT5651_JD_NOT_INV), + }, { /* I.T.Works TW701, Ployer Momo7w and Trekstor ST70416-6 * (these all use the same mainboard) */ @@ -525,6 +540,9 @@ static int byt_rt5651_add_codec_device_props(struct device *i2c_dev) if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN) props[cnt++] = PROPERTY_ENTRY_BOOL("realtek,dmic-en"); + if (byt_rt5651_quirk & BYT_RT5651_JD_NOT_INV) + props[cnt++] = PROPERTY_ENTRY_BOOL("realtek,jack-detect-not-inverted"); + return device_add_properties(i2c_dev, props); } -- cgit v1.2.3-59-g8ed1b From 35033d8cb71b4dd4fa8d8d28a534af1ee921b9a0 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 17 Mar 2019 16:50:24 +0900 Subject: ALSA: firewire-motu: add support MOTU 8pre FireWire This commit adds support for MOTU 8pre FireWire, which was shipped 2007 and nowadays already discontinued. Userspace applications can transmit and receive PCM frames and MIDI messages for this model via ALSA PCM interface and RawMidi/Sequencer interfaces. Like the other models of MOTU FireWire series, this model has many quirks in its CIP. At first, data channels for two pairs of optical interfaces. At lower sampling transmission frequency, i.e. 44.1 and 48.0 kHz, one pair is available for ADAT data, thus 8 data chunks are transferred by CIP. At middle sampling transmission frequency, i.e. 88.2 and 96.0 kHz, two pairs are available to keep 8 chunks for ADAT data, thus CIP still includes 8 data chunks. Apart from data chunks for optical interface, CIP includes fixed number of data chunks. In tx stream, two chunks for status message, eight chunks for samples from analog 1-8 input, two chunks for mix-return. In rx stream, two chunks for control message, two chunks for main 1-2 output, two chunks for phone 1-2 output, two chunks for dummy 1-2. CIP header in tx stream includes quirks for its dbs and dbc fields. The value of dbs field is fixed to 0x13, against its actual size. The value of dbc field is firstly updated to 0x07 from zero, then it's incremented continuously according to actual number of data h blocks. Finally, the model has own bits to disable frame fetch. This commit uses several options to absorb the above quirks. $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 0410b57d bus_info_length 4, crc_length 16, crc 46461 404 31333934 bus_name "1394" 408 20001000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 1 (4) 40c 0001f200 company_id 0001f2 | 410 00083dfb device_id 0000083dfb | EUI-64 0001f20000083dfb root directory ----------------------------------------------------------------- 414 0004c65c directory_length 4, crc 50780 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 8d000006 --> eui-64 leaf at 438 424 d1000001 --> unit directory at 428 unit directory at 428 ----------------------------------------------------------------- 428 0003991c directory_length 3, crc 39196 42c 120001f2 specifier id 430 1300000f version 434 17103800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 00022681 leaf_length 2, crc 9857 43c 0001f200 company_id 0001f2 | 440 00083dfb device_id 0000083dfb | EUI-64 0001f20000083dfb Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/motu/amdtp-motu.c | 6 +++++ sound/firewire/motu/motu-protocol-v2.c | 43 ++++++++++++++++++++++++---------- sound/firewire/motu/motu.c | 15 ++++++++++++ sound/firewire/motu/motu.h | 1 + 4 files changed, 53 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 6c9b743ea74b..cb0c967dea63 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -412,6 +412,12 @@ int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, CIP_HEADER_WITHOUT_EOH; fmt = CIP_FMT_MOTU_TX_V3; } + + if (protocol == &snd_motu_protocol_v2) { + // 8pre has some quirks. + flags |= CIP_WRONG_DBS | + CIP_SKIP_DBC_ZERO_CHECK; + } } else { process_data_blocks = process_rx_data_blocks; flags |= CIP_DBC_IS_END_EVENT; diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index 453fc29fade7..848fffe7387e 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -15,6 +15,8 @@ #define V2_CLOCK_SRC_SHIFT 0 #define V2_CLOCK_TRAVELER_FETCH_DISABLE 0x04000000 #define V2_CLOCK_TRAVELER_FETCH_ENABLE 0x03000000 +#define V2_CLOCK_8PRE_FETCH_DISABLE 0x02000000 +#define V2_CLOCK_8PRE_FETCH_ENABLE 0x00000000 #define V2_IN_OUT_CONF_OFFSET 0x0c04 #define V2_OPT_OUT_IFACE_MASK 0x00000c00 @@ -132,20 +134,31 @@ static int v2_switch_fetching_mode(struct snd_motu *motu, bool enable) u32 data; int err = 0; - if (motu->spec == &snd_motu_spec_traveler) { + if (motu->spec == &snd_motu_spec_traveler || + motu->spec == &snd_motu_spec_8pre) { err = snd_motu_transaction_read(motu, V2_CLOCK_STATUS_OFFSET, ®, sizeof(reg)); if (err < 0) return err; data = be32_to_cpu(reg); - data &= ~(V2_CLOCK_TRAVELER_FETCH_DISABLE | - V2_CLOCK_TRAVELER_FETCH_ENABLE); - - if (enable) - data |= V2_CLOCK_TRAVELER_FETCH_ENABLE; - else - data |= V2_CLOCK_TRAVELER_FETCH_DISABLE; + if (motu->spec == &snd_motu_spec_traveler) { + data &= ~(V2_CLOCK_TRAVELER_FETCH_DISABLE | + V2_CLOCK_TRAVELER_FETCH_ENABLE); + + if (enable) + data |= V2_CLOCK_TRAVELER_FETCH_ENABLE; + else + data |= V2_CLOCK_TRAVELER_FETCH_DISABLE; + } else if (motu->spec == &snd_motu_spec_8pre) { + data &= ~(V2_CLOCK_8PRE_FETCH_DISABLE | + V2_CLOCK_8PRE_FETCH_ENABLE); + + if (enable) + data |= V2_CLOCK_8PRE_FETCH_DISABLE; + else + data |= V2_CLOCK_8PRE_FETCH_ENABLE; + } reg = cpu_to_be32(data); err = snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET, @@ -220,10 +233,16 @@ static void calculate_differed_part(struct snd_motu_packet_format *formats, * interfaces. */ data = (data & mask) >> shift; - if ((flags & SND_MOTU_SPEC_HAS_OPT_IFACE_A) && - data == V2_OPT_IFACE_MODE_ADAT) { - pcm_chunks[0] += 8; - pcm_chunks[1] += 4; + if (data == V2_OPT_IFACE_MODE_ADAT) { + if (flags & SND_MOTU_SPEC_HAS_OPT_IFACE_A) { + pcm_chunks[0] += 8; + pcm_chunks[1] += 4; + } + // 8pre has two sets of optical interface and doesn't reduce + // chunks for ADAT signals. + if (flags & SND_MOTU_SPEC_HAS_OPT_IFACE_B) { + pcm_chunks[1] += 4; + } } /* At mode x4, no data chunks are supported in this part. */ diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 513291ba0ab0..201539d4488c 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -203,6 +203,20 @@ const struct snd_motu_spec snd_motu_spec_traveler = { .analog_out_ports = 8, }; +const struct snd_motu_spec snd_motu_spec_8pre = { + .name = "8pre", + .protocol = &snd_motu_protocol_v2, + // In tx, use coax chunks for mix-return 1/2. In rx, use coax chunks for + // dummy 1/2. + .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | + SND_MOTU_SPEC_HAS_OPT_IFACE_A | + SND_MOTU_SPEC_HAS_OPT_IFACE_B | + SND_MOTU_SPEC_RX_MIDI_2ND_Q | + SND_MOTU_SPEC_TX_MIDI_2ND_Q, + .analog_in_ports = 8, + .analog_out_ports = 2, +}; + static const struct snd_motu_spec motu_828mk3 = { .name = "828mk3", .protocol = &snd_motu_protocol_v3, @@ -248,6 +262,7 @@ static const struct snd_motu_spec motu_audio_express = { static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000003, &motu_828mk2), SND_MOTU_DEV_ENTRY(0x000009, &snd_motu_spec_traveler), + SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3), /* FireWire only. */ SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3), /* Hybrid. */ SND_MOTU_DEV_ENTRY(0x000033, &motu_audio_express), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index fd5327d30ab1..1cd112be7dad 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -130,6 +130,7 @@ extern const struct snd_motu_protocol snd_motu_protocol_v2; extern const struct snd_motu_protocol snd_motu_protocol_v3; extern const struct snd_motu_spec snd_motu_spec_traveler; +extern const struct snd_motu_spec snd_motu_spec_8pre; int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, -- cgit v1.2.3-59-g8ed1b From cc4f8e91c4ed04a9a972c6e10c158d4b71be3f2f Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 17 Mar 2019 20:25:06 +0900 Subject: ALSA: firewire-lib: use 8 byte header for IR context to get isochronous cycle In kernel API of Linux FireWire subsystem, handlers of isochronous receive (IR) context can get context headers as an argument of callback. When 4 byte header is used, the context header includes isochronous packet header for each packet. When 8 byte header is used, it includes isochronous cycle as well. ALSA IEC 61883-1/6 engine uses 4 byte header, and computes isochronous cycle from the cycle of interrupt. The usage of 8 byte header can obsolete the computation. Furthermore, this change works well for a case that a series of packet in one interrupt includes skipped isochronous cycle, This commit uses 8 byte header to handle isochronous cycle. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 44 +++++++++++++++++++------------------------ 1 file changed, 19 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 3ada55ed5381..43f28b813386 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -56,8 +56,9 @@ #define INTERRUPT_INTERVAL 16 #define QUEUE_LENGTH 48 -#define IN_PACKET_HEADER_SIZE 4 +#define IR_HEADER_SIZE 8 // For header and timestamp. #define OUT_PACKET_HEADER_SIZE 0 +#define HEADER_TSTAMP_MASK 0x0000ffff static void pcm_period_tasklet(unsigned long data); @@ -456,7 +457,7 @@ static inline int queue_out_packet(struct amdtp_stream *s, static inline int queue_in_packet(struct amdtp_stream *s) { - return queue_packet(s, IN_PACKET_HEADER_SIZE, s->max_payload_length); + return queue_packet(s, IR_HEADER_SIZE, s->max_payload_length); } static int handle_out_packet(struct amdtp_stream *s, @@ -701,13 +702,6 @@ static inline u32 increment_cycle_count(u32 cycle, unsigned int addend) return cycle; } -static inline u32 decrement_cycle_count(u32 cycle, unsigned int subtrahend) -{ - if (cycle < subtrahend) - cycle += 8 * CYCLES_PER_SECOND; - return cycle - subtrahend; -} - static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) @@ -745,29 +739,26 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, struct amdtp_stream *s = private_data; unsigned int i, packets; unsigned int payload_length, max_payload_length; - __be32 *headers = header; - u32 cycle; + __be32 *ctx_header = header; if (s->packet_index < 0) return; /* The number of packets in buffer */ - packets = header_length / IN_PACKET_HEADER_SIZE; - - cycle = compute_cycle_count(tstamp); - - /* Align to actual cycle count for the last packet. */ - cycle = decrement_cycle_count(cycle, packets); + packets = header_length / IR_HEADER_SIZE; /* For buffer-over-run prevention. */ max_payload_length = s->max_payload_length; for (i = 0; i < packets; i++) { - cycle = increment_cycle_count(cycle, 1); + u32 iso_header = be32_to_cpu(ctx_header[0]); + unsigned int cycle; + + tstamp = be32_to_cpu(ctx_header[1]) & HEADER_TSTAMP_MASK; + cycle = compute_cycle_count(tstamp); /* The number of bytes in this packet */ - payload_length = - (be32_to_cpu(headers[i]) >> ISO_DATA_LENGTH_SHIFT); + payload_length = iso_header >> ISO_DATA_LENGTH_SHIFT; if (payload_length > max_payload_length) { dev_err(&s->unit->device, "Detect jumbo payload: %04x %04x\n", @@ -777,6 +768,8 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, if (s->handle_packet(s, payload_length, cycle, i) < 0) break; + + ctx_header += IR_HEADER_SIZE / sizeof(__be32); } /* Queueing error or detecting invalid payload. */ @@ -797,6 +790,7 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, void *header, void *private_data) { struct amdtp_stream *s = private_data; + __be32 *ctx_header = header; u32 cycle; unsigned int packets; @@ -807,11 +801,10 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, s->callbacked = true; wake_up(&s->callback_wait); - cycle = compute_cycle_count(tstamp); - if (s->direction == AMDTP_IN_STREAM) { - packets = header_length / IN_PACKET_HEADER_SIZE; - cycle = decrement_cycle_count(cycle, packets); + tstamp = be32_to_cpu(ctx_header[1]) & HEADER_TSTAMP_MASK; + cycle = compute_cycle_count(tstamp); + context->callback.sc = in_stream_callback; if (s->flags & CIP_NO_HEADER) s->handle_packet = handle_in_packet_without_header; @@ -819,6 +812,7 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, s->handle_packet = handle_in_packet; } else { packets = header_length / 4; + cycle = compute_cycle_count(tstamp); cycle = increment_cycle_count(cycle, QUEUE_LENGTH - packets); context->callback.sc = out_stream_callback; if (s->flags & CIP_NO_HEADER) @@ -880,7 +874,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) if (s->direction == AMDTP_IN_STREAM) { dir = DMA_FROM_DEVICE; type = FW_ISO_CONTEXT_RECEIVE; - header_size = IN_PACKET_HEADER_SIZE; + header_size = IR_HEADER_SIZE; } else { dir = DMA_TO_DEVICE; type = FW_ISO_CONTEXT_TRANSMIT; -- cgit v1.2.3-59-g8ed1b From cdc444721b0eea6817d1003add2f99414dc8028d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 18 Mar 2019 13:50:08 +0900 Subject: ASoC: simple-card: rename simple_get_conversion() to simple_parse_convert() use same naming rule Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7147bba45a2a..5238c912cecd 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -172,9 +172,9 @@ static int simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static void simple_get_conversion(struct device *dev, - struct device_node *np, - struct asoc_simple_card_data *adata) +static void simple_parse_convert(struct device *dev, + struct device_node *np, + struct asoc_simple_card_data *adata) { struct device_node *top = dev->of_node; struct device_node *node = of_get_parent(np); @@ -295,7 +295,7 @@ static int simple_dai_link_of_dpcm(struct simple_priv *priv, "prefix"); } - simple_get_conversion(dev, np, &dai_props->adata); + simple_parse_convert(dev, np, &dai_props->adata); asoc_simple_card_canonicalize_platform(dai_link); @@ -471,7 +471,7 @@ static int simple_for_each_link(struct simple_priv *priv, /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); for_each_child_of_node(node, np) - simple_get_conversion(dev, np, &adata); + simple_parse_convert(dev, np, &adata); /* loop for all CPU/Codec node */ for_each_child_of_node(node, np) { -- cgit v1.2.3-59-g8ed1b From d708d35ef084535ba1a06d0c0f858e18a5edef5a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 18 Mar 2019 13:50:17 +0900 Subject: ASoC: simple-card: add simple_parse_mclk_fs() It is parsing mclk_fs at many places, but it should be same operation. This patch adds simple_parse_mclk_fs() and parse it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 33 ++++++++++++++++++++++----------- 1 file changed, 22 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5238c912cecd..b792638a11c8 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -187,6 +187,26 @@ static void simple_parse_convert(struct device *dev, of_node_put(node); } +static void simple_parse_mclk_fs(struct device_node *top, + struct device_node *cpu, + struct device_node *codec, + struct simple_dai_props *props, + char *prefix) +{ + struct device_node *node = of_get_parent(cpu); + char prop[128]; + + snprintf(prop, sizeof(prop), "%smclk-fs", PREFIX); + of_property_read_u32(top, prop, &props->mclk_fs); + + snprintf(prop, sizeof(prop), "%smclk-fs", prefix); + of_property_read_u32(node, prop, &props->mclk_fs); + of_property_read_u32(cpu, prop, &props->mclk_fs); + of_property_read_u32(codec, prop, &props->mclk_fs); + + of_node_put(node); +} + static int simple_dai_link_of_dpcm(struct simple_priv *priv, struct device_node *np, struct device_node *codec, @@ -200,7 +220,6 @@ static int simple_dai_link_of_dpcm(struct simple_priv *priv, struct snd_soc_dai_link_component *codecs = dai_link->codecs; struct device_node *top = dev->of_node; struct device_node *node = of_get_parent(np); - char prop[128]; char *prefix = ""; int ret; @@ -296,6 +315,7 @@ static int simple_dai_link_of_dpcm(struct simple_priv *priv, } simple_parse_convert(dev, np, &dai_props->adata); + simple_parse_mclk_fs(top, np, codec, dai_props, prefix); asoc_simple_card_canonicalize_platform(dai_link); @@ -303,11 +323,6 @@ static int simple_dai_link_of_dpcm(struct simple_priv *priv, if (ret) return ret; - snprintf(prop, sizeof(prop), "%smclk-fs", prefix); - of_property_read_u32(top, PREFIX "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(node, prop, &dai_props->mclk_fs); - of_property_read_u32(np, prop, &dai_props->mclk_fs); - ret = asoc_simple_card_parse_daifmt(dev, node, codec, prefix, &dai_link->dai_fmt); if (ret < 0) @@ -372,11 +387,7 @@ static int simple_dai_link_of(struct simple_priv *priv, if (ret < 0) goto dai_link_of_err; - snprintf(prop, sizeof(prop), "%smclk-fs", prefix); - of_property_read_u32(top, PREFIX "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(node, prop, &dai_props->mclk_fs); - of_property_read_u32(cpu, prop, &dai_props->mclk_fs); - of_property_read_u32(codec, prop, &dai_props->mclk_fs); + simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); ret = asoc_simple_card_parse_cpu(cpu, dai_link, DAI, CELL, &single_cpu); -- cgit v1.2.3-59-g8ed1b From d2bf008ad11abc52d6e084bc109be2ee34a74a14 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 18 Mar 2019 13:50:25 +0900 Subject: ASoC: audio-graph: rename graph_get_conversion() to graph_parse_convert() use same naming rule, and this patch add missing of_node_put() on it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index bb12351330e8..733ba1e89dd6 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -176,9 +176,9 @@ static int graph_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static void graph_get_conversion(struct device *dev, - struct device_node *ep, - struct asoc_simple_card_data *adata) +static void graph_parse_convert(struct device *dev, + struct device_node *ep, + struct asoc_simple_card_data *adata) { struct device_node *top = dev->of_node; struct device_node *port = of_get_parent(ep); @@ -190,6 +190,10 @@ static void graph_get_conversion(struct device *dev, asoc_simple_card_parse_convert(dev, ports, NULL, adata); asoc_simple_card_parse_convert(dev, port, NULL, adata); asoc_simple_card_parse_convert(dev, ep, NULL, adata); + + of_node_put(port); + of_node_put(ports); + of_node_put(node); } static int graph_dai_link_of_dpcm(struct graph_priv *priv, @@ -227,7 +231,7 @@ static int graph_dai_link_of_dpcm(struct graph_priv *priv, of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); - graph_get_conversion(dev, ep, &dai_props->adata); + graph_parse_convert(dev, ep, &dai_props->adata); of_node_put(ports); of_node_put(port); @@ -462,8 +466,8 @@ static int graph_for_each_link(struct graph_priv *priv, /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); - graph_get_conversion(dev, codec_ep, &adata); - graph_get_conversion(dev, cpu_ep, &adata); + graph_parse_convert(dev, codec_ep, &adata); + graph_parse_convert(dev, cpu_ep, &adata); /* * It is DPCM -- cgit v1.2.3-59-g8ed1b From 4346a745df4107d7ac61846f60087992e9d6e1b8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 18 Mar 2019 13:50:31 +0900 Subject: ASoC: audio-graph: add graph_parse_mclk_fs() It is parsing mclk_fs at many places, but it should be same operation. This patch adds graph_parse_mclk_fs() and parse it. This patch also renames similar function graph_get_conversion() to graph_parse_convert(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 50 +++++++++++++++++------------------- 1 file changed, 23 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 733ba1e89dd6..7dd772a895e2 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -196,6 +196,24 @@ static void graph_parse_convert(struct device *dev, of_node_put(node); } +static void graph_parse_mclk_fs(struct device_node *top, + struct device_node *ep, + struct graph_dai_props *props) +{ + struct device_node *port = of_get_parent(ep); + struct device_node *ports = of_get_parent(port); + struct device_node *node = of_graph_get_port_parent(ep); + + of_property_read_u32(top, "mclk-fs", &props->mclk_fs); + of_property_read_u32(ports, "mclk-fs", &props->mclk_fs); + of_property_read_u32(port, "mclk-fs", &props->mclk_fs); + of_property_read_u32(ep, "mclk-fs", &props->mclk_fs); + + of_node_put(port); + of_node_put(ports); + of_node_put(node); +} + static int graph_dai_link_of_dpcm(struct graph_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, @@ -226,13 +244,6 @@ static int graph_dai_link_of_dpcm(struct graph_priv *priv, dev_dbg(dev, "link_of DPCM (%pOF)\n", ep); - of_property_read_u32(top, "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(ports, "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(port, "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(ep, "mclk-fs", &dai_props->mclk_fs); - - graph_parse_convert(dev, ep, &dai_props->adata); - of_node_put(ports); of_node_put(port); of_node_put(node); @@ -311,6 +322,9 @@ static int graph_dai_link_of_dpcm(struct graph_priv *priv, "prefix"); } + graph_parse_convert(dev, ep, &dai_props->adata); + graph_parse_mclk_fs(top, ep, dai_props); + asoc_simple_card_canonicalize_platform(dai_link); ret = asoc_simple_card_of_parse_tdm(ep, dai); @@ -339,10 +353,6 @@ static int graph_dai_link_of(struct graph_priv *priv, struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); struct graph_dai_props *dai_props = graph_priv_to_props(priv, li->link); struct device_node *top = dev->of_node; - struct device_node *cpu_port; - struct device_node *cpu_ports; - struct device_node *codec_port; - struct device_node *codec_ports; struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; int ret; @@ -351,11 +361,6 @@ static int graph_dai_link_of(struct graph_priv *priv, if (!li->cpu) return 0; - cpu_port = of_get_parent(cpu_ep); - cpu_ports = of_get_parent(cpu_port); - codec_port = of_get_parent(codec_ep); - codec_ports = of_get_parent(codec_port); - dev_dbg(dev, "link_of (%pOF)\n", cpu_ep); li->link++; @@ -366,17 +371,8 @@ static int graph_dai_link_of(struct graph_priv *priv, dai_props->codec_dai = &priv->dais[li->dais++]; /* Factor to mclk, used in hw_params() */ - of_property_read_u32(top, "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(cpu_ports, "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(codec_ports, "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(cpu_port, "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(codec_port, "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(cpu_ep, "mclk-fs", &dai_props->mclk_fs); - of_property_read_u32(codec_ep, "mclk-fs", &dai_props->mclk_fs); - of_node_put(cpu_port); - of_node_put(cpu_ports); - of_node_put(codec_port); - of_node_put(codec_ports); + graph_parse_mclk_fs(top, cpu_ep, dai_props); + graph_parse_mclk_fs(top, codec_ep, dai_props); ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, NULL, &dai_link->dai_fmt); -- cgit v1.2.3-59-g8ed1b From 7b6f68a4df4df3a58e854d24da991640bd4d9590 Mon Sep 17 00:00:00 2001 From: Bard liao Date: Tue, 5 Mar 2019 23:57:52 +0800 Subject: ASoC: topology: free stream_name of dai_drv The stream_name is allocated by kstrdup. We have to free it when the dai is removed or return from error. Signed-off-by: Bard liao Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index fbee4720b834..1be60cea2bf5 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -535,6 +535,8 @@ static void remove_dai(struct snd_soc_component *comp, if (dai->driver == dai_drv) dai->driver = NULL; + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); kfree(dai_drv->name); list_del(&dobj->list); kfree(dai_drv); @@ -1808,6 +1810,9 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + kfree(dai_drv->name); kfree(dai_drv); return ret; } -- cgit v1.2.3-59-g8ed1b From b3718b8f7e9b4afdfd281d35233b078585aef2e9 Mon Sep 17 00:00:00 2001 From: Bard liao Date: Tue, 5 Mar 2019 23:57:53 +0800 Subject: ASoC: topology: free link string in error Some strings are allocated by kstrdup, but not freed when error happened. Signed-off-by: Bard liao Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 1be60cea2bf5..c6a30d000f59 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1883,6 +1883,9 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, ret = soc_tplg_dai_link_load(tplg, link, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n"); + kfree(link->name); + kfree(link->stream_name); + kfree(link->cpu_dai_name); kfree(link); return ret; } -- cgit v1.2.3-59-g8ed1b From 2b0ee49f129cf3e9885f8d83439024e8116b865c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Mar 2019 11:52:08 +0000 Subject: ASoC: wm_adsp: Don't export wm_adsp2_lock This function is not presently called from outside the adsp code and nor should it be, as such stop exporting it. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 61 +++++++++++++++++++++++----------------------- sound/soc/codecs/wm_adsp.h | 1 - 2 files changed, 30 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5608ed5decca..a9542a56b19e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2622,6 +2622,36 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } +static int wm_adsp2_lock(struct wm_adsp *dsp, unsigned int lock_regions) +{ + struct regmap *regmap = dsp->regmap; + unsigned int code0, code1, lock_reg; + + if (!(lock_regions & WM_ADSP2_REGION_ALL)) + return 0; + + lock_regions &= WM_ADSP2_REGION_ALL; + lock_reg = dsp->base + ADSP2_LOCK_REGION_1_LOCK_REGION_0; + + while (lock_regions) { + code0 = code1 = 0; + if (lock_regions & BIT(0)) { + code0 = ADSP2_LOCK_CODE_0; + code1 = ADSP2_LOCK_CODE_1; + } + if (lock_regions & BIT(1)) { + code0 |= ADSP2_LOCK_CODE_0 << ADSP2_LOCK_REGION_SHIFT; + code1 |= ADSP2_LOCK_CODE_1 << ADSP2_LOCK_REGION_SHIFT; + } + regmap_write(regmap, lock_reg, code0); + regmap_write(regmap, lock_reg, code1); + lock_regions >>= 2; + lock_reg += 2; + } + + return 0; +} + static void wm_adsp2_boot_work(struct work_struct *work) { struct wm_adsp *dsp = container_of(work, @@ -3890,37 +3920,6 @@ int wm_adsp_compr_copy(struct snd_compr_stream *stream, char __user *buf, } EXPORT_SYMBOL_GPL(wm_adsp_compr_copy); -int wm_adsp2_lock(struct wm_adsp *dsp, unsigned int lock_regions) -{ - struct regmap *regmap = dsp->regmap; - unsigned int code0, code1, lock_reg; - - if (!(lock_regions & WM_ADSP2_REGION_ALL)) - return 0; - - lock_regions &= WM_ADSP2_REGION_ALL; - lock_reg = dsp->base + ADSP2_LOCK_REGION_1_LOCK_REGION_0; - - while (lock_regions) { - code0 = code1 = 0; - if (lock_regions & BIT(0)) { - code0 = ADSP2_LOCK_CODE_0; - code1 = ADSP2_LOCK_CODE_1; - } - if (lock_regions & BIT(1)) { - code0 |= ADSP2_LOCK_CODE_0 << ADSP2_LOCK_REGION_SHIFT; - code1 |= ADSP2_LOCK_CODE_1 << ADSP2_LOCK_REGION_SHIFT; - } - regmap_write(regmap, lock_reg, code0); - regmap_write(regmap, lock_reg, code1); - lock_regions >>= 2; - lock_reg += 2; - } - - return 0; -} -EXPORT_SYMBOL_GPL(wm_adsp2_lock); - static void wm_adsp_fatal_error(struct wm_adsp *dsp) { struct wm_adsp_compr *compr; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 8f09b4419a91..65ccf120dd5b 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -141,7 +141,6 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event, unsigned int freq); -int wm_adsp2_lock(struct wm_adsp *adsp, unsigned int regions); irqreturn_t wm_adsp2_bus_error(struct wm_adsp *adsp); int wm_adsp2_event(struct snd_soc_dapm_widget *w, -- cgit v1.2.3-59-g8ed1b From b9070df451668e0c317d420d8331568b4d7ba6df Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 19 Mar 2019 11:52:09 +0000 Subject: ASoC: wm_adsp: Move wm_adsp2_set_dspclk to CODEC drivers The original wm_adsp2_early_event took an additional frequency argument for clocking control so could not be used directly as a DAPM callback. But this setup could equally be done by the codec driver function wrapping wm_adsp2_early event. In preparation for adding support for new core types wm_adsp2_set_dspclk has been exported, and the freq argument removed so that it can be used directly as a DAPM callback. Signed-off-by: Richard Fitzgerald Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l24.c | 4 +++- sound/soc/codecs/wm5102.c | 4 +++- sound/soc/codecs/wm5110.c | 4 +++- sound/soc/codecs/wm_adsp.c | 32 +++++++++++++------------------- sound/soc/codecs/wm_adsp.h | 6 ++++-- 5 files changed, 26 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index b16832a6a9af..e056d871fafb 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -75,7 +75,9 @@ static int cs47l24_adsp_power_ev(struct snd_soc_dapm_widget *w, v = (v & ARIZONA_SYSCLK_FREQ_MASK) >> ARIZONA_SYSCLK_FREQ_SHIFT; - return wm_adsp2_early_event(w, kcontrol, event, v); + wm_adsp2_set_dspclk(w, v); + + return wm_adsp2_early_event(w, kcontrol, event); } static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 4466e195b66d..c972591f1cc4 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -646,6 +646,8 @@ static int wm5102_adsp_power_ev(struct snd_soc_dapm_widget *w, return ret; } } + + wm_adsp2_set_dspclk(w, v); break; case SND_SOC_DAPM_POST_PMD: @@ -659,7 +661,7 @@ static int wm5102_adsp_power_ev(struct snd_soc_dapm_widget *w, break; } - return wm_adsp2_early_event(w, kcontrol, event, v); + return wm_adsp2_early_event(w, kcontrol, event); } static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index b25877fa529d..c543b73e2f20 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -211,7 +211,9 @@ static int wm5110_adsp_power_ev(struct snd_soc_dapm_widget *w, v = (v & ARIZONA_SYSCLK_FREQ_MASK) >> ARIZONA_SYSCLK_FREQ_SHIFT; - return wm_adsp2_early_event(w, kcontrol, event, v); + wm_adsp2_set_dspclk(w, v); + + return wm_adsp2_early_event(w, kcontrol, event); } static const struct reg_sequence wm5110_no_dre_left_enable[] = { diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a9542a56b19e..8800ffcc4f8c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2715,26 +2715,22 @@ err_mutex: mutex_unlock(&dsp->pwr_lock); } -static void wm_adsp2_set_dspclk(struct wm_adsp *dsp, unsigned int freq) +int wm_adsp2_set_dspclk(struct snd_soc_dapm_widget *w, unsigned int freq) { + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsp = &dsps[w->shift]; int ret; - switch (dsp->rev) { - case 0: - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CLOCKING, - ADSP2_CLK_SEL_MASK, - freq << ADSP2_CLK_SEL_SHIFT); - if (ret) { - adsp_err(dsp, "Failed to set clock rate: %d\n", ret); - return; - } - break; - default: - /* clock is handled by parent codec driver */ - break; - } + ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CLOCKING, + ADSP2_CLK_SEL_MASK, + freq << ADSP2_CLK_SEL_SHIFT); + if (ret) + adsp_err(dsp, "Failed to set clock rate: %d\n", ret); + + return ret; } +EXPORT_SYMBOL_GPL(wm_adsp2_set_dspclk); int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2792,8 +2788,7 @@ static void wm_adsp_stop_watchdog(struct wm_adsp *dsp) } int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event, - unsigned int freq) + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); @@ -2802,7 +2797,6 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - wm_adsp2_set_dspclk(dsp, freq); queue_work(system_unbound_wq, &dsp->boot_work); break; case SND_SOC_DAPM_PRE_PMD: diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 65ccf120dd5b..ac1bec3b2248 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -137,15 +137,17 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp int wm_adsp2_component_remove(struct wm_adsp *dsp, struct snd_soc_component *component); int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); + int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event, - unsigned int freq); + struct snd_kcontrol *kcontrol, int event); irqreturn_t wm_adsp2_bus_error(struct wm_adsp *adsp); int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int wm_adsp2_set_dspclk(struct snd_soc_dapm_widget *w, unsigned int freq); + int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, -- cgit v1.2.3-59-g8ed1b From 4049ce86fc9b0512d4f624bb8b63c8f513fb66d1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Mar 2019 11:52:10 +0000 Subject: ASoC: wm_adsp: Refactor firmware status reading In preparation for further additions refactor the reading of the firmware status. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 43 ++++++++++++++++++------------------------- 1 file changed, 18 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 8800ffcc4f8c..fb252762f23c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -788,49 +788,42 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, } } -static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) +static void wm_adsp_read_fw_status(struct wm_adsp *dsp, + int noffs, unsigned int *offs) { - unsigned int scratch[4]; - unsigned int addr = dsp->base + ADSP2_SCRATCH0; unsigned int i; int ret; - for (i = 0; i < ARRAY_SIZE(scratch); ++i) { - ret = regmap_read(dsp->regmap, addr + i, &scratch[i]); + for (i = 0; i < noffs; ++i) { + ret = regmap_read(dsp->regmap, dsp->base + offs[i], &offs[i]); if (ret) { adsp_err(dsp, "Failed to read SCRATCH%u: %d\n", i, ret); return; } } +} + +static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) +{ + unsigned int offs[] = { + ADSP2_SCRATCH0, ADSP2_SCRATCH1, ADSP2_SCRATCH2, ADSP2_SCRATCH3, + }; + + wm_adsp_read_fw_status(dsp, ARRAY_SIZE(offs), offs); adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", - scratch[0], scratch[1], scratch[2], scratch[3]); + offs[0], offs[1], offs[2], offs[3]); } static void wm_adsp2v2_show_fw_status(struct wm_adsp *dsp) { - unsigned int scratch[2]; - int ret; + unsigned int offs[] = { ADSP2V2_SCRATCH0_1, ADSP2V2_SCRATCH2_3 }; - ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH0_1, - &scratch[0]); - if (ret) { - adsp_err(dsp, "Failed to read SCRATCH0_1: %d\n", ret); - return; - } - - ret = regmap_read(dsp->regmap, dsp->base + ADSP2V2_SCRATCH2_3, - &scratch[1]); - if (ret) { - adsp_err(dsp, "Failed to read SCRATCH2_3: %d\n", ret); - return; - } + wm_adsp_read_fw_status(dsp, ARRAY_SIZE(offs), offs); adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", - scratch[0] & 0xFFFF, - scratch[0] >> 16, - scratch[1] & 0xFFFF, - scratch[1] >> 16); + offs[0] & 0xFFFF, offs[0] >> 16, + offs[1] & 0xFFFF, offs[1] >> 16); } static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext) -- cgit v1.2.3-59-g8ed1b From a5dcb24d70ffbb4ea47b8eefad1158d033b9dec9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Mar 2019 11:52:11 +0000 Subject: ASoC: wm_adsp: Factor out parsing of firmware ID header There is no need to duplicate this code for both ADSP1 and 2 as the handling is exactly the same. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index fb252762f23c..acb57dd2c2ad 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1982,6 +1982,18 @@ static void wm_adsp_free_alg_regions(struct wm_adsp *dsp) } } +static void wmfw_parse_id_header(struct wm_adsp *dsp, + struct wmfw_id_hdr *fw, int nalgs) +{ + dsp->fw_id = be32_to_cpu(fw->id); + dsp->fw_id_version = be32_to_cpu(fw->ver); + + adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", + dsp->fw_id, (dsp->fw_id_version & 0xff0000) >> 16, + (dsp->fw_id_version & 0xff00) >> 8, dsp->fw_id_version & 0xff, + nalgs); +} + static int wm_adsp1_setup_algs(struct wm_adsp *dsp) { struct wmfw_adsp1_id_hdr adsp1_id; @@ -2005,13 +2017,8 @@ static int wm_adsp1_setup_algs(struct wm_adsp *dsp) } n_algs = be32_to_cpu(adsp1_id.n_algs); - dsp->fw_id = be32_to_cpu(adsp1_id.fw.id); - adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", - dsp->fw_id, - (be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16, - (be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8, - be32_to_cpu(adsp1_id.fw.ver) & 0xff, - n_algs); + + wmfw_parse_id_header(dsp, &adsp1_id.fw, n_algs); alg_region = wm_adsp_create_region(dsp, WMFW_ADSP1_ZM, adsp1_id.fw.id, adsp1_id.zm); @@ -2111,14 +2118,8 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp) } n_algs = be32_to_cpu(adsp2_id.n_algs); - dsp->fw_id = be32_to_cpu(adsp2_id.fw.id); - dsp->fw_id_version = be32_to_cpu(adsp2_id.fw.ver); - adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", - dsp->fw_id, - (dsp->fw_id_version & 0xff0000) >> 16, - (dsp->fw_id_version & 0xff00) >> 8, - dsp->fw_id_version & 0xff, - n_algs); + + wmfw_parse_id_header(dsp, &adsp2_id.fw, n_algs); alg_region = wm_adsp_create_region(dsp, WMFW_ADSP2_XM, adsp2_id.fw.id, adsp2_id.xm); -- cgit v1.2.3-59-g8ed1b From 4e08d50d1fb6144df4b0b5c75a17edd344bf3d1b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Mar 2019 11:52:12 +0000 Subject: ASoC: wm_adsp: Factor out DSP specific operations In preparation for the addition of more types of DSP core refactor the handling of DSP specific operations such as starting the memory or enabling the core into a set of callbacks. This should make it easier to add new core types and allow for more code reuse between them. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm_adsp.c | 383 ++++++++++++++++++++++++++++----------------- sound/soc/codecs/wm_adsp.h | 35 ++++- 5 files changed, 271 insertions(+), 153 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index e056d871fafb..eebbf02e1c39 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -77,7 +77,7 @@ static int cs47l24_adsp_power_ev(struct snd_soc_dapm_widget *w, wm_adsp2_set_dspclk(w, v); - return wm_adsp2_early_event(w, kcontrol, event); + return wm_adsp_early_event(w, kcontrol, event); } static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c972591f1cc4..b32e8313954d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -661,7 +661,7 @@ static int wm5102_adsp_power_ev(struct snd_soc_dapm_widget *w, break; } - return wm_adsp2_early_event(w, kcontrol, event); + return wm_adsp_early_event(w, kcontrol, event); } static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c543b73e2f20..1f500cc8d96a 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -213,7 +213,7 @@ static int wm5110_adsp_power_ev(struct snd_soc_dapm_widget *w, wm_adsp2_set_dspclk(w, v); - return wm_adsp2_early_event(w, kcontrol, event); + return wm_adsp_early_event(w, kcontrol, event); } static const struct reg_sequence wm5110_no_dre_left_enable[] = { diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index acb57dd2c2ad..eccc640d6184 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -227,6 +227,9 @@ */ #define WM_ADSP_FW_EVENT_SHUTDOWN 0x000001 +struct wm_adsp_ops wm_adsp1_ops; +struct wm_adsp_ops wm_adsp2_ops[]; + struct wm_adsp_buf { struct list_head list; void *buf; @@ -1640,6 +1643,52 @@ static int wm_adsp_parse_coeff(struct wm_adsp *dsp, return 0; } +static unsigned int wm_adsp1_parse_sizes(struct wm_adsp *dsp, + const char * const file, + unsigned int pos, + const struct firmware *firmware) +{ + const struct wmfw_adsp1_sizes *adsp1_sizes; + + adsp1_sizes = (void *)&firmware->data[pos]; + + adsp_dbg(dsp, "%s: %d DM, %d PM, %d ZM\n", file, + le32_to_cpu(adsp1_sizes->dm), le32_to_cpu(adsp1_sizes->pm), + le32_to_cpu(adsp1_sizes->zm)); + + return pos + sizeof(*adsp1_sizes); +} + +static unsigned int wm_adsp2_parse_sizes(struct wm_adsp *dsp, + const char * const file, + unsigned int pos, + const struct firmware *firmware) +{ + const struct wmfw_adsp2_sizes *adsp2_sizes; + + adsp2_sizes = (void *)&firmware->data[pos]; + + adsp_dbg(dsp, "%s: %d XM, %d YM %d PM, %d ZM\n", file, + le32_to_cpu(adsp2_sizes->xm), le32_to_cpu(adsp2_sizes->ym), + le32_to_cpu(adsp2_sizes->pm), le32_to_cpu(adsp2_sizes->zm)); + + return pos + sizeof(*adsp2_sizes); +} + +static bool wm_adsp_validate_version(struct wm_adsp *dsp, unsigned int version) +{ + switch (version) { + case 0: + adsp_warn(dsp, "Deprecated file format %d\n", version); + return true; + case 1: + case 2: + return true; + default: + return false; + } +} + static int wm_adsp_load(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -1648,7 +1697,6 @@ static int wm_adsp_load(struct wm_adsp *dsp) unsigned int pos = 0; const struct wmfw_header *header; const struct wmfw_adsp1_sizes *adsp1_sizes; - const struct wmfw_adsp2_sizes *adsp2_sizes; const struct wmfw_footer *footer; const struct wmfw_region *region; const struct wm_adsp_region *mem; @@ -1657,7 +1705,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) struct wm_adsp_buf *buf; unsigned int reg; int regions = 0; - int ret, offset, type, sizes; + int ret, offset, type; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -1688,15 +1736,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) goto out_fw; } - switch (header->ver) { - case 0: - adsp_warn(dsp, "%s: Depreciated file format %d\n", - file, header->ver); - break; - case 1: - case 2: - break; - default: + if (!dsp->ops->validate_version(dsp, header->ver)) { adsp_err(dsp, "%s: unknown file format %d\n", file, header->ver); goto out_fw; @@ -1711,39 +1751,13 @@ static int wm_adsp_load(struct wm_adsp *dsp) goto out_fw; } - switch (dsp->type) { - case WMFW_ADSP1: - pos = sizeof(*header) + sizeof(*adsp1_sizes) + sizeof(*footer); - adsp1_sizes = (void *)&(header[1]); - footer = (void *)&(adsp1_sizes[1]); - sizes = sizeof(*adsp1_sizes); + pos = sizeof(*header); + pos = dsp->ops->parse_sizes(dsp, file, pos, firmware); - adsp_dbg(dsp, "%s: %d DM, %d PM, %d ZM\n", - file, le32_to_cpu(adsp1_sizes->dm), - le32_to_cpu(adsp1_sizes->pm), - le32_to_cpu(adsp1_sizes->zm)); - break; - - case WMFW_ADSP2: - pos = sizeof(*header) + sizeof(*adsp2_sizes) + sizeof(*footer); - adsp2_sizes = (void *)&(header[1]); - footer = (void *)&(adsp2_sizes[1]); - sizes = sizeof(*adsp2_sizes); - - adsp_dbg(dsp, "%s: %d XM, %d YM %d PM, %d ZM\n", - file, le32_to_cpu(adsp2_sizes->xm), - le32_to_cpu(adsp2_sizes->ym), - le32_to_cpu(adsp2_sizes->pm), - le32_to_cpu(adsp2_sizes->zm)); - break; + footer = (void *)&firmware->data[pos]; + pos += sizeof(*footer); - default: - WARN(1, "Unknown DSP type"); - goto out_fw; - } - - if (le32_to_cpu(header->len) != sizeof(*header) + - sizes + sizeof(*footer)) { + if (le32_to_cpu(header->len) != pos) { adsp_err(dsp, "%s: unexpected header length %d\n", file, le32_to_cpu(header->len)); goto out_fw; @@ -2458,6 +2472,8 @@ static int wm_adsp_common_init(struct wm_adsp *dsp) int wm_adsp1_init(struct wm_adsp *dsp) { + dsp->ops = &wm_adsp1_ops; + return wm_adsp_common_init(dsp); } EXPORT_SYMBOL_GPL(wm_adsp1_init); @@ -2577,23 +2593,11 @@ err_mutex: } EXPORT_SYMBOL_GPL(wm_adsp1_event); -static int wm_adsp2_ena(struct wm_adsp *dsp) +static int wm_adsp2v2_enable_core(struct wm_adsp *dsp) { unsigned int val; int ret, count; - switch (dsp->rev) { - case 0: - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA, ADSP2_SYS_ENA); - if (ret != 0) - return ret; - break; - default: - break; - } - /* Wait for the RAM to start, should be near instantaneous */ for (count = 0; count < 10; ++count) { ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1, &val); @@ -2616,6 +2620,18 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return 0; } +static int wm_adsp2_enable_core(struct wm_adsp *dsp) +{ + int ret; + + ret = regmap_update_bits_async(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA, ADSP2_SYS_ENA); + if (ret != 0) + return ret; + + return wm_adsp2v2_enable_core(dsp); +} + static int wm_adsp2_lock(struct wm_adsp *dsp, unsigned int lock_regions) { struct regmap *regmap = dsp->regmap; @@ -2646,7 +2662,36 @@ static int wm_adsp2_lock(struct wm_adsp *dsp, unsigned int lock_regions) return 0; } -static void wm_adsp2_boot_work(struct work_struct *work) +static int wm_adsp2_enable_memory(struct wm_adsp *dsp) +{ + return regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_MEM_ENA, ADSP2_MEM_ENA); +} + +static void wm_adsp2_disable_memory(struct wm_adsp *dsp) +{ + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_MEM_ENA, 0); +} + +static void wm_adsp2_disable_core(struct wm_adsp *dsp) +{ + regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0); + regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_1, 0); + regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_2, 0); + + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA, 0); +} + +static void wm_adsp2v2_disable_core(struct wm_adsp *dsp) +{ + regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0); + regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_1, 0); + regmap_write(dsp->regmap, dsp->base + ADSP2V2_WDMA_CONFIG_2, 0); +} + +static void wm_adsp_boot_work(struct work_struct *work) { struct wm_adsp *dsp = container_of(work, struct wm_adsp, @@ -2655,20 +2700,23 @@ static void wm_adsp2_boot_work(struct work_struct *work) mutex_lock(&dsp->pwr_lock); - ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_MEM_ENA, ADSP2_MEM_ENA); - if (ret != 0) - goto err_mutex; + if (dsp->ops->enable_memory) { + ret = dsp->ops->enable_memory(dsp); + if (ret != 0) + goto err_mutex; + } - ret = wm_adsp2_ena(dsp); - if (ret != 0) - goto err_mem; + if (dsp->ops->enable_core) { + ret = dsp->ops->enable_core(dsp); + if (ret != 0) + goto err_mem; + } ret = wm_adsp_load(dsp); if (ret != 0) goto err_ena; - ret = wm_adsp2_setup_algs(dsp); + ret = dsp->ops->setup_algs(dsp); if (ret != 0) goto err_ena; @@ -2681,17 +2729,8 @@ static void wm_adsp2_boot_work(struct work_struct *work) if (ret != 0) goto err_ena; - switch (dsp->rev) { - case 0: - /* Turn DSP back off until we are ready to run */ - ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA, 0); - if (ret != 0) - goto err_ena; - break; - default: - break; - } + if (dsp->ops->disable_core) + dsp->ops->disable_core(dsp); dsp->booted = true; @@ -2700,11 +2739,11 @@ static void wm_adsp2_boot_work(struct work_struct *work) return; err_ena: - regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0); + if (dsp->ops->disable_core) + dsp->ops->disable_core(dsp); err_mem: - regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_MEM_ENA, 0); + if (dsp->ops->disable_memory) + dsp->ops->disable_memory(dsp); err_mutex: mutex_unlock(&dsp->pwr_lock); } @@ -2771,18 +2810,12 @@ EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); static void wm_adsp_stop_watchdog(struct wm_adsp *dsp) { - switch (dsp->rev) { - case 0: - case 1: - return; - default: - regmap_update_bits(dsp->regmap, dsp->base + ADSP2_WATCHDOG, - ADSP2_WDT_ENA_MASK, 0); - } + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_WATCHDOG, + ADSP2_WDT_ENA_MASK, 0); } -int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +int wm_adsp_early_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); @@ -2803,8 +2836,8 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, dsp->booted = false; - regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_MEM_ENA, 0); + if (dsp->ops->disable_memory) + dsp->ops->disable_memory(dsp); list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; @@ -2821,10 +2854,23 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, return 0; } -EXPORT_SYMBOL_GPL(wm_adsp2_early_event); +EXPORT_SYMBOL_GPL(wm_adsp_early_event); -int wm_adsp2_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int wm_adsp2_start_core(struct wm_adsp *dsp) +{ + return regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_CORE_ENA | ADSP2_START, + ADSP2_CORE_ENA | ADSP2_START); +} + +static void wm_adsp2_stop_core(struct wm_adsp *dsp) +{ + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_CORE_ENA | ADSP2_START, 0); +} + +int wm_adsp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); @@ -2842,23 +2888,31 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, goto err; } - ret = wm_adsp2_ena(dsp); - if (ret != 0) - goto err; + if (dsp->ops->enable_core) { + ret = dsp->ops->enable_core(dsp); + if (ret != 0) + goto err; + } /* Sync set controls */ ret = wm_coeff_sync_controls(dsp); if (ret != 0) goto err; - wm_adsp2_lock(dsp, dsp->lock_regions); + if (dsp->ops->lock_memory) { + ret = dsp->ops->lock_memory(dsp, dsp->lock_regions); + if (ret != 0) { + adsp_err(dsp, "Error configuring MPU: %d\n", + ret); + goto err; + } + } - ret = regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA | ADSP2_START, - ADSP2_CORE_ENA | ADSP2_START); - if (ret != 0) - goto err; + if (dsp->ops->start_core) { + ret = dsp->ops->start_core(dsp); + if (ret != 0) + goto err; + } if (wm_adsp_fw[dsp->fw].num_caps != 0) { ret = wm_adsp_buffer_init(dsp); @@ -2869,56 +2923,27 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, dsp->running = true; mutex_unlock(&dsp->pwr_lock); - break; case SND_SOC_DAPM_PRE_PMD: /* Tell the firmware to cleanup */ wm_adsp_signal_event_controls(dsp, WM_ADSP_FW_EVENT_SHUTDOWN); - wm_adsp_stop_watchdog(dsp); + if (dsp->ops->stop_watchdog) + dsp->ops->stop_watchdog(dsp); /* Log firmware state, it can be useful for analysis */ - switch (dsp->rev) { - case 0: - wm_adsp2_show_fw_status(dsp); - break; - default: - wm_adsp2v2_show_fw_status(dsp); - break; - } + if (dsp->ops->show_fw_status) + dsp->ops->show_fw_status(dsp); mutex_lock(&dsp->pwr_lock); dsp->running = false; - regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA | ADSP2_START, 0); - - /* Make sure DMAs are quiesced */ - switch (dsp->rev) { - case 0: - regmap_write(dsp->regmap, - dsp->base + ADSP2_RDMA_CONFIG_1, 0); - regmap_write(dsp->regmap, - dsp->base + ADSP2_WDMA_CONFIG_1, 0); - regmap_write(dsp->regmap, - dsp->base + ADSP2_WDMA_CONFIG_2, 0); - - regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA, 0); - break; - default: - regmap_write(dsp->regmap, - dsp->base + ADSP2_RDMA_CONFIG_1, 0); - regmap_write(dsp->regmap, - dsp->base + ADSP2_WDMA_CONFIG_1, 0); - regmap_write(dsp->regmap, - dsp->base + ADSP2V2_WDMA_CONFIG_2, 0); - break; - } + if (dsp->ops->stop_core) + dsp->ops->stop_core(dsp); + if (dsp->ops->disable_core) + dsp->ops->disable_core(dsp); if (wm_adsp_fw[dsp->fw].num_caps != 0) wm_adsp_buffer_free(dsp); @@ -2936,12 +2961,14 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, return 0; err: - regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA | ADSP2_CORE_ENA | ADSP2_START, 0); + if (dsp->ops->stop_core) + dsp->ops->stop_core(dsp); + if (dsp->ops->disable_core) + dsp->ops->disable_core(dsp); mutex_unlock(&dsp->pwr_lock); return ret; } -EXPORT_SYMBOL_GPL(wm_adsp2_event); +EXPORT_SYMBOL_GPL(wm_adsp_event); int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *component) { @@ -2987,12 +3014,18 @@ int wm_adsp2_init(struct wm_adsp *dsp) "Failed to clear memory retention: %d\n", ret); return ret; } + + dsp->ops = &wm_adsp2_ops[0]; + break; + case 1: + dsp->ops = &wm_adsp2_ops[1]; break; default: + dsp->ops = &wm_adsp2_ops[2]; break; } - INIT_WORK(&dsp->boot_work, wm_adsp2_boot_work); + INIT_WORK(&dsp->boot_work, wm_adsp_boot_work); return 0; } @@ -3988,4 +4021,64 @@ error: } EXPORT_SYMBOL_GPL(wm_adsp2_bus_error); +struct wm_adsp_ops wm_adsp1_ops = { + .validate_version = wm_adsp_validate_version, + .parse_sizes = wm_adsp1_parse_sizes, +}; + +struct wm_adsp_ops wm_adsp2_ops[] = { + { + .parse_sizes = wm_adsp2_parse_sizes, + .validate_version = wm_adsp_validate_version, + .setup_algs = wm_adsp2_setup_algs, + + .show_fw_status = wm_adsp2_show_fw_status, + + .enable_memory = wm_adsp2_enable_memory, + .disable_memory = wm_adsp2_disable_memory, + + .enable_core = wm_adsp2_enable_core, + .disable_core = wm_adsp2_disable_core, + + .start_core = wm_adsp2_start_core, + .stop_core = wm_adsp2_stop_core, + + }, + { + .parse_sizes = wm_adsp2_parse_sizes, + .validate_version = wm_adsp_validate_version, + .setup_algs = wm_adsp2_setup_algs, + + .show_fw_status = wm_adsp2v2_show_fw_status, + + .enable_memory = wm_adsp2_enable_memory, + .disable_memory = wm_adsp2_disable_memory, + .lock_memory = wm_adsp2_lock, + + .enable_core = wm_adsp2v2_enable_core, + .disable_core = wm_adsp2v2_disable_core, + + .start_core = wm_adsp2_start_core, + .stop_core = wm_adsp2_stop_core, + }, + { + .parse_sizes = wm_adsp2_parse_sizes, + .validate_version = wm_adsp_validate_version, + .setup_algs = wm_adsp2_setup_algs, + + .show_fw_status = wm_adsp2v2_show_fw_status, + .stop_watchdog = wm_adsp_stop_watchdog, + + .enable_memory = wm_adsp2_enable_memory, + .disable_memory = wm_adsp2_disable_memory, + .lock_memory = wm_adsp2_lock, + + .enable_core = wm_adsp2v2_enable_core, + .disable_core = wm_adsp2v2_disable_core, + + .start_core = wm_adsp2_start_core, + .stop_core = wm_adsp2_stop_core, + }, +}; + MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index ac1bec3b2248..c75a671c19a1 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -54,6 +54,7 @@ struct wm_adsp_alg_region { struct wm_adsp_compr; struct wm_adsp_compr_buf; +struct wm_adsp_ops; struct wm_adsp { const char *part; @@ -66,6 +67,8 @@ struct wm_adsp { struct regmap *regmap; struct snd_soc_component *component; + struct wm_adsp_ops *ops; + unsigned int base; unsigned int sysclk_reg; unsigned int sysclk_mask; @@ -106,6 +109,28 @@ struct wm_adsp { }; +struct wm_adsp_ops { + bool (*validate_version)(struct wm_adsp *dsp, unsigned int version); + unsigned int (*parse_sizes)(struct wm_adsp *dsp, + const char * const file, + unsigned int pos, + const struct firmware *firmware); + int (*setup_algs)(struct wm_adsp *dsp); + + void (*show_fw_status)(struct wm_adsp *dsp); + void (*stop_watchdog)(struct wm_adsp *dsp); + + int (*enable_memory)(struct wm_adsp *dsp); + void (*disable_memory)(struct wm_adsp *dsp); + int (*lock_memory)(struct wm_adsp *dsp, unsigned int lock_regions); + + int (*enable_core)(struct wm_adsp *dsp); + void (*disable_core)(struct wm_adsp *dsp); + + int (*start_core)(struct wm_adsp *dsp); + void (*stop_core)(struct wm_adsp *dsp); +}; + #define WM_ADSP1(wname, num) \ SND_SOC_DAPM_PGA_E(wname, SND_SOC_NOPM, num, 0, NULL, 0, \ wm_adsp1_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) @@ -121,7 +146,7 @@ struct wm_adsp { .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD, \ .subseq = 100, /* Ensure we run after SYSCLK supply widget */ }, \ { .id = snd_soc_dapm_out_drv, .name = wname, \ - .reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \ + .reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp_event, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } #define WM_ADSP_FW_CONTROL(dspname, num) \ @@ -138,13 +163,13 @@ int wm_adsp2_component_remove(struct wm_adsp *dsp, struct snd_soc_component *com int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event); +int wm_adsp_early_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); irqreturn_t wm_adsp2_bus_error(struct wm_adsp *adsp); -int wm_adsp2_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event); +int wm_adsp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); int wm_adsp2_set_dspclk(struct snd_soc_dapm_widget *w, unsigned int freq); -- cgit v1.2.3-59-g8ed1b From ed1666f6867059e1ed4e2e565a1c87c5580f1b38 Mon Sep 17 00:00:00 2001 From: Pi-Hsun Shih Date: Wed, 6 Mar 2019 17:09:02 +0800 Subject: ASoC: mediatek: Fix UBSAN warning. In sound/soc/mediatek/common/mtk-afe-fe-dai.c, when xxx_reg is -1, it's a no-op to call mtk_regmap_update_bits, but since both xxx_reg and xxx_shift are set to -1, the (1 << xxx_shift) in the argument would trigger a UBSAN warning. Fix the warning by setting those xxx_shift to 0 instead. Note that since the code explicitly checks .mono_shift >= 0 and .fs_shift >= 0 before using them in '<<' operator, those two members are not set to 0. Signed-off-by: Pi-Hsun Shih Signed-off-by: Mark Brown --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 9 ++++----- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 13 ------------- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 16 ---------------- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 16 ---------------- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 20 -------------------- 5 files changed, 4 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index cf4978be062f..3830e582e188 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -197,11 +197,10 @@ int mtk_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - if (memif->data->enable_shift >= 0) - mtk_regmap_update_bits(afe->regmap, - memif->data->enable_reg, - 1 << memif->data->enable_shift, - 1 << memif->data->enable_shift); + mtk_regmap_update_bits(afe->regmap, + memif->data->enable_reg, + 1 << memif->data->enable_shift, + 1 << memif->data->enable_shift); /* set irq counter */ mtk_regmap_update_bits(afe->regmap, irq_data->irq_cnt_reg, diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 968fba4d7533..7064a9fd6f74 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -994,7 +994,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 6, .msb_reg = -1, - .msb_shift = -1, }, { .name = "DL2", @@ -1013,7 +1012,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 7, .msb_reg = -1, - .msb_shift = -1, }, { .name = "DL3", @@ -1032,7 +1030,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 8, .msb_reg = -1, - .msb_shift = -1, }, { .name = "DL4", @@ -1051,7 +1048,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 9, .msb_reg = -1, - .msb_shift = -1, }, { .name = "DL5", @@ -1070,7 +1066,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 10, .msb_reg = -1, - .msb_shift = -1, }, { .name = "DLM", @@ -1089,7 +1084,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 12, .msb_reg = -1, - .msb_shift = -1, }, { .name = "UL1", @@ -1108,7 +1102,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 0, .msb_reg = -1, - .msb_shift = -1, }, { .name = "UL2", @@ -1127,7 +1120,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 1, .msb_reg = -1, - .msb_shift = -1, }, { .name = "UL3", @@ -1146,7 +1138,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 2, .msb_reg = -1, - .msb_shift = -1, }, { .name = "UL4", @@ -1165,7 +1156,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 3, .msb_reg = -1, - .msb_shift = -1, }, { .name = "UL5", @@ -1184,7 +1174,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 4, .msb_reg = -1, - .msb_shift = -1, }, { .name = "DLBT", @@ -1203,7 +1192,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 13, .msb_reg = -1, - .msb_shift = -1, }, { .name = "ULBT", @@ -1222,7 +1210,6 @@ static const struct mtk_base_memif_data memif_data[MT2701_MEMIF_NUM] = { .agent_disable_reg = AUDIO_TOP_CON5, .agent_disable_shift = 16, .msb_reg = -1, - .msb_shift = -1, }, }; diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index bff7d71d0742..08a6532da322 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -401,9 +401,7 @@ static const struct mtk_base_memif_data memif_data[MT6797_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = DL1_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT6797_MEMIF_DL2] = { .name = "DL2", @@ -420,9 +418,7 @@ static const struct mtk_base_memif_data memif_data[MT6797_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = DL2_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT6797_MEMIF_DL3] = { .name = "DL3", @@ -439,9 +435,7 @@ static const struct mtk_base_memif_data memif_data[MT6797_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = DL3_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT6797_MEMIF_VUL] = { .name = "VUL", @@ -458,9 +452,7 @@ static const struct mtk_base_memif_data memif_data[MT6797_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = VUL_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT6797_MEMIF_AWB] = { .name = "AWB", @@ -477,9 +469,7 @@ static const struct mtk_base_memif_data memif_data[MT6797_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = AWB_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT6797_MEMIF_VUL12] = { .name = "VUL12", @@ -496,9 +486,7 @@ static const struct mtk_base_memif_data memif_data[MT6797_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = VUL_DATA2_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT6797_MEMIF_DAI] = { .name = "DAI", @@ -515,9 +503,7 @@ static const struct mtk_base_memif_data memif_data[MT6797_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = DAI_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT6797_MEMIF_MOD_DAI] = { .name = "MOD_DAI", @@ -534,9 +520,7 @@ static const struct mtk_base_memif_data memif_data[MT6797_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = MOD_DAI_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 166aed28330d..0382896c162e 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -714,13 +714,11 @@ static const struct mtk_base_memif_data memif_data[MT8173_AFE_MEMIF_NUM] = { .mono_reg = AFE_DAC_CON1, .mono_shift = 21, .hd_reg = -1, - .hd_shift = -1, .enable_reg = AFE_DAC_CON0, .enable_shift = 1, .msb_reg = AFE_MEMIF_MSB, .msb_shift = 0, .agent_disable_reg = -1, - .agent_disable_shift = -1, }, { .name = "DL2", .id = MT8173_AFE_MEMIF_DL2, @@ -732,13 +730,11 @@ static const struct mtk_base_memif_data memif_data[MT8173_AFE_MEMIF_NUM] = { .mono_reg = AFE_DAC_CON1, .mono_shift = 22, .hd_reg = -1, - .hd_shift = -1, .enable_reg = AFE_DAC_CON0, .enable_shift = 2, .msb_reg = AFE_MEMIF_MSB, .msb_shift = 1, .agent_disable_reg = -1, - .agent_disable_shift = -1, }, { .name = "VUL", .id = MT8173_AFE_MEMIF_VUL, @@ -750,13 +746,11 @@ static const struct mtk_base_memif_data memif_data[MT8173_AFE_MEMIF_NUM] = { .mono_reg = AFE_DAC_CON1, .mono_shift = 27, .hd_reg = -1, - .hd_shift = -1, .enable_reg = AFE_DAC_CON0, .enable_shift = 3, .msb_reg = AFE_MEMIF_MSB, .msb_shift = 6, .agent_disable_reg = -1, - .agent_disable_shift = -1, }, { .name = "DAI", .id = MT8173_AFE_MEMIF_DAI, @@ -768,13 +762,11 @@ static const struct mtk_base_memif_data memif_data[MT8173_AFE_MEMIF_NUM] = { .mono_reg = -1, .mono_shift = -1, .hd_reg = -1, - .hd_shift = -1, .enable_reg = AFE_DAC_CON0, .enable_shift = 4, .msb_reg = AFE_MEMIF_MSB, .msb_shift = 5, .agent_disable_reg = -1, - .agent_disable_shift = -1, }, { .name = "AWB", .id = MT8173_AFE_MEMIF_AWB, @@ -786,13 +778,11 @@ static const struct mtk_base_memif_data memif_data[MT8173_AFE_MEMIF_NUM] = { .mono_reg = AFE_DAC_CON1, .mono_shift = 24, .hd_reg = -1, - .hd_shift = -1, .enable_reg = AFE_DAC_CON0, .enable_shift = 6, .msb_reg = AFE_MEMIF_MSB, .msb_shift = 3, .agent_disable_reg = -1, - .agent_disable_shift = -1, }, { .name = "MOD_DAI", .id = MT8173_AFE_MEMIF_MOD_DAI, @@ -804,13 +794,11 @@ static const struct mtk_base_memif_data memif_data[MT8173_AFE_MEMIF_NUM] = { .mono_reg = AFE_DAC_CON1, .mono_shift = 30, .hd_reg = -1, - .hd_shift = -1, .enable_reg = AFE_DAC_CON0, .enable_shift = 7, .msb_reg = AFE_MEMIF_MSB, .msb_shift = 4, .agent_disable_reg = -1, - .agent_disable_shift = -1, }, { .name = "HDMI", .id = MT8173_AFE_MEMIF_HDMI, @@ -822,13 +810,10 @@ static const struct mtk_base_memif_data memif_data[MT8173_AFE_MEMIF_NUM] = { .mono_reg = -1, .mono_shift = -1, .hd_reg = -1, - .hd_shift = -1, .enable_reg = -1, - .enable_shift = -1, .msb_reg = AFE_MEMIF_MSB, .msb_shift = 8, .agent_disable_reg = -1, - .agent_disable_shift = -1, }, }; @@ -914,7 +899,6 @@ static const struct mtk_base_irq_data irq_data[MT8173_AFE_IRQ_NUM] = { .irq_en_reg = AFE_IRQ_MCU_CON, .irq_en_shift = 12, .irq_fs_reg = -1, - .irq_fs_shift = -1, .irq_fs_maskbit = -1, .irq_clr_reg = AFE_IRQ_CLR, .irq_clr_shift = 4, diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 4e045dd305a7..43be51bf0329 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -421,9 +421,7 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = DL1_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT8183_MEMIF_DL2] = { .name = "DL2", @@ -440,9 +438,7 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = DL2_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT8183_MEMIF_DL3] = { .name = "DL3", @@ -459,9 +455,7 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = DL3_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT8183_MEMIF_VUL2] = { .name = "VUL2", @@ -478,9 +472,7 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = VUL2_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT8183_MEMIF_AWB] = { .name = "AWB", @@ -497,9 +489,7 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = AWB_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT8183_MEMIF_AWB2] = { .name = "AWB2", @@ -516,9 +506,7 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = AWB2_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT8183_MEMIF_VUL12] = { .name = "VUL12", @@ -535,9 +523,7 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = VUL12_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT8183_MEMIF_MOD_DAI] = { .name = "MOD_DAI", @@ -554,9 +540,7 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = MOD_DAI_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, [MT8183_MEMIF_HDMI] = { .name = "HDMI", @@ -569,13 +553,10 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .mono_reg = -1, .mono_shift = -1, .enable_reg = -1, /* control in tdm for sync start */ - .enable_shift = -1, .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = HDMI_HD_SFT, .agent_disable_reg = -1, - .agent_disable_shift = -1, .msb_reg = -1, - .msb_shift = -1, }, }; @@ -690,7 +671,6 @@ static const struct mtk_base_irq_data irq_data[MT8183_IRQ_NUM] = { .irq_cnt_shift = 0, .irq_cnt_maskbit = 0x3ffff, .irq_fs_reg = -1, - .irq_fs_shift = -1, .irq_fs_maskbit = -1, .irq_en_reg = AFE_IRQ_MCU_CON0, .irq_en_shift = IRQ8_MCU_ON_SFT, -- cgit v1.2.3-59-g8ed1b From c6839641532f95aca2246f7c116565b86b8ce878 Mon Sep 17 00:00:00 2001 From: Pi-Hsun Shih Date: Wed, 6 Mar 2019 17:09:03 +0800 Subject: ASoC: mediatek: Reduce repititive code on mtk_regmap_update_bits. Change the signature of mtk_regmap_update_bits to also take a shift, and warn when reg >= 0 but shift < 0. This reduce the code repetition on the calling side, and prevent future UBSAN warning when some of the xxx_shift and xxx_reg are both set to -1. Signed-off-by: Pi-Hsun Shih Signed-off-by: Mark Brown --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 46 ++++++++++++------------------ 1 file changed, 18 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index 3830e582e188..fded11d14cde 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -18,11 +18,11 @@ static int mtk_regmap_update_bits(struct regmap *map, int reg, unsigned int mask, - unsigned int val) + unsigned int val, int shift) { - if (reg < 0) + if (reg < 0 || WARN_ON_ONCE(shift < 0)) return 0; - return regmap_update_bits(map, reg, mask, val); + return regmap_update_bits(map, reg, mask << shift, val << shift); } static int mtk_regmap_write(struct regmap *map, int reg, unsigned int val) @@ -49,8 +49,7 @@ int mtk_afe_fe_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16); /* enable agent */ mtk_regmap_update_bits(afe->regmap, memif->data->agent_disable_reg, - 1 << memif->data->agent_disable_shift, - 0 << memif->data->agent_disable_shift); + 1, 0, memif->data->agent_disable_shift); snd_soc_set_runtime_hwparams(substream, mtk_afe_hardware); @@ -105,8 +104,7 @@ void mtk_afe_fe_shutdown(struct snd_pcm_substream *substream, irq_id = memif->irq_usage; mtk_regmap_update_bits(afe->regmap, memif->data->agent_disable_reg, - 1 << memif->data->agent_disable_shift, - 1 << memif->data->agent_disable_shift); + 1, 1, memif->data->agent_disable_shift); if (!memif->const_irq) { mtk_dynamic_irq_release(afe, irq_id); @@ -144,16 +142,14 @@ int mtk_afe_fe_hw_params(struct snd_pcm_substream *substream, /* set MSB to 33-bit */ mtk_regmap_update_bits(afe->regmap, memif->data->msb_reg, - 1 << memif->data->msb_shift, - msb_at_bit33 << memif->data->msb_shift); + 1, msb_at_bit33, memif->data->msb_shift); /* set channel */ if (memif->data->mono_shift >= 0) { unsigned int mono = (params_channels(params) == 1) ? 1 : 0; mtk_regmap_update_bits(afe->regmap, memif->data->mono_reg, - 1 << memif->data->mono_shift, - mono << memif->data->mono_shift); + 1, mono, memif->data->mono_shift); } /* set rate */ @@ -166,8 +162,8 @@ int mtk_afe_fe_hw_params(struct snd_pcm_substream *substream, return -EINVAL; mtk_regmap_update_bits(afe->regmap, memif->data->fs_reg, - memif->data->fs_maskbit << memif->data->fs_shift, - fs << memif->data->fs_shift); + memif->data->fs_maskbit, fs, + memif->data->fs_shift); return 0; } @@ -199,14 +195,12 @@ int mtk_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: mtk_regmap_update_bits(afe->regmap, memif->data->enable_reg, - 1 << memif->data->enable_shift, - 1 << memif->data->enable_shift); + 1, 1, memif->data->enable_shift); /* set irq counter */ mtk_regmap_update_bits(afe->regmap, irq_data->irq_cnt_reg, - irq_data->irq_cnt_maskbit - << irq_data->irq_cnt_shift, - counter << irq_data->irq_cnt_shift); + irq_data->irq_cnt_maskbit, counter, + irq_data->irq_cnt_shift); /* set irq fs */ fs = afe->irq_fs(substream, runtime->rate); @@ -215,24 +209,21 @@ int mtk_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, return -EINVAL; mtk_regmap_update_bits(afe->regmap, irq_data->irq_fs_reg, - irq_data->irq_fs_maskbit - << irq_data->irq_fs_shift, - fs << irq_data->irq_fs_shift); + irq_data->irq_fs_maskbit, fs, + irq_data->irq_fs_shift); /* enable interrupt */ mtk_regmap_update_bits(afe->regmap, irq_data->irq_en_reg, - 1 << irq_data->irq_en_shift, - 1 << irq_data->irq_en_shift); + 1, 1, irq_data->irq_en_shift); return 0; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: mtk_regmap_update_bits(afe->regmap, memif->data->enable_reg, - 1 << memif->data->enable_shift, 0); + 1, 0, memif->data->enable_shift); /* disable interrupt */ mtk_regmap_update_bits(afe->regmap, irq_data->irq_en_reg, - 1 << irq_data->irq_en_shift, - 0 << irq_data->irq_en_shift); + 1, 0, irq_data->irq_en_shift); /* and clear pending IRQ */ mtk_regmap_write(afe->regmap, irq_data->irq_clr_reg, 1 << irq_data->irq_clr_shift); @@ -269,8 +260,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, } mtk_regmap_update_bits(afe->regmap, memif->data->hd_reg, - 1 << memif->data->hd_shift, - hd_audio << memif->data->hd_shift); + 1, hd_audio, memif->data->hd_shift); return 0; } -- cgit v1.2.3-59-g8ed1b From 170b1e123f385e7d60e992bc7fb1cc2897520769 Mon Sep 17 00:00:00 2001 From: Wen Shi Date: Tue, 19 Mar 2019 11:52:13 +0000 Subject: ASoC: wm_adsp: Add support for new Halo core DSPs The Halo core is a new generation of audio DSP architecture from Cirrus Logic. A new iteration of the WMFW file format (v3) is also added, for this new architecture. Currently this format is not supported on the old ADSP2 architecture however support may be added for it in the future. Signed-off-by: Wen Shi Signed-off-by: Piotr Stankiewicz Signed-off-by: Richard Fitzgerald Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 354 +++++++++++++++++++++++++++++++++++++++++---- sound/soc/codecs/wm_adsp.h | 8 + sound/soc/codecs/wmfw.h | 30 ++++ 3 files changed, 363 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index eccc640d6184..bd31056db4f5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -227,8 +227,54 @@ */ #define WM_ADSP_FW_EVENT_SHUTDOWN 0x000001 +/* + * HALO core + */ +#define HALO_SCRATCH1 0x005c0 +#define HALO_SCRATCH2 0x005c8 +#define HALO_SCRATCH3 0x005d0 +#define HALO_SCRATCH4 0x005d8 +#define HALO_CCM_CORE_CONTROL 0x41000 +#define HALO_CORE_SOFT_RESET 0x00010 + +/* + * HALO MPU banks + */ +#define HALO_MPU_XMEM_ACCESS_0 0x43000 +#define HALO_MPU_YMEM_ACCESS_0 0x43004 +#define HALO_MPU_WINDOW_ACCESS_0 0x43008 +#define HALO_MPU_XREG_ACCESS_0 0x4300C +#define HALO_MPU_YREG_ACCESS_0 0x43014 +#define HALO_MPU_XMEM_ACCESS_1 0x43018 +#define HALO_MPU_YMEM_ACCESS_1 0x4301C +#define HALO_MPU_WINDOW_ACCESS_1 0x43020 +#define HALO_MPU_XREG_ACCESS_1 0x43024 +#define HALO_MPU_YREG_ACCESS_1 0x4302C +#define HALO_MPU_XMEM_ACCESS_2 0x43030 +#define HALO_MPU_YMEM_ACCESS_2 0x43034 +#define HALO_MPU_WINDOW_ACCESS_2 0x43038 +#define HALO_MPU_XREG_ACCESS_2 0x4303C +#define HALO_MPU_YREG_ACCESS_2 0x43044 +#define HALO_MPU_XMEM_ACCESS_3 0x43048 +#define HALO_MPU_YMEM_ACCESS_3 0x4304C +#define HALO_MPU_WINDOW_ACCESS_3 0x43050 +#define HALO_MPU_XREG_ACCESS_3 0x43054 +#define HALO_MPU_YREG_ACCESS_3 0x4305C +#define HALO_MPU_LOCK_CONFIG 0x43140 + +/* + * HALO_CCM_CORE_CONTROL + */ +#define HALO_CORE_EN 0x00000001 + +/* + * HALO_CORE_SOFT_RESET + */ +#define HALO_CORE_SOFT_RESET_MASK 0x00000001 + struct wm_adsp_ops wm_adsp1_ops; struct wm_adsp_ops wm_adsp2_ops[]; +struct wm_adsp_ops wm_halo_ops; struct wm_adsp_buf { struct list_head list; @@ -309,6 +355,12 @@ struct wm_adsp_system_config_xm_hdr { __be32 build_job_number; }; +struct wm_halo_system_config_xm_hdr { + __be32 halo_heartbeat; + __be32 build_job_name[3]; + __be32 build_job_number; +}; + struct wm_adsp_alg_xm_struct { __be32 magic; __be32 smoothing; @@ -535,12 +587,18 @@ static const char *wm_adsp_mem_region_name(unsigned int type) switch (type) { case WMFW_ADSP1_PM: return "PM"; + case WMFW_HALO_PM_PACKED: + return "PM_PACKED"; case WMFW_ADSP1_DM: return "DM"; case WMFW_ADSP2_XM: return "XM"; + case WMFW_HALO_XM_PACKED: + return "XM_PACKED"; case WMFW_ADSP2_YM: return "YM"; + case WMFW_HALO_YM_PACKED: + return "YM_PACKED"; case WMFW_ADSP1_ZM: return "ZM"; default: @@ -772,17 +830,12 @@ static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, unsigned int offset) { - if (WARN_ON(!mem)) - return offset; switch (mem->type) { case WMFW_ADSP1_PM: return mem->base + (offset * 3); case WMFW_ADSP1_DM: - return mem->base + (offset * 2); case WMFW_ADSP2_XM: - return mem->base + (offset * 2); case WMFW_ADSP2_YM: - return mem->base + (offset * 2); case WMFW_ADSP1_ZM: return mem->base + (offset * 2); default: @@ -791,6 +844,24 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, } } +static unsigned int wm_halo_region_to_reg(struct wm_adsp_region const *mem, + unsigned int offset) +{ + switch (mem->type) { + case WMFW_ADSP2_XM: + case WMFW_ADSP2_YM: + return mem->base + (offset * 4); + case WMFW_HALO_XM_PACKED: + case WMFW_HALO_YM_PACKED: + return (mem->base + (offset * 3)) & ~0x3; + case WMFW_HALO_PM_PACKED: + return mem->base + (offset * 5); + default: + WARN(1, "Unknown memory region type"); + return offset; + } +} + static void wm_adsp_read_fw_status(struct wm_adsp *dsp, int noffs, unsigned int *offs) { @@ -829,6 +900,18 @@ static void wm_adsp2v2_show_fw_status(struct wm_adsp *dsp) offs[1] & 0xFFFF, offs[1] >> 16); } +static void wm_halo_show_fw_status(struct wm_adsp *dsp) +{ + unsigned int offs[] = { + HALO_SCRATCH1, HALO_SCRATCH2, HALO_SCRATCH3, HALO_SCRATCH4, + }; + + wm_adsp_read_fw_status(dsp, ARRAY_SIZE(offs), offs); + + adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", + offs[0], offs[1], offs[2], offs[3]); +} + static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext) { return container_of(ext, struct wm_coeff_ctl, bytes_ext); @@ -847,7 +930,7 @@ static int wm_coeff_base_reg(struct wm_coeff_ctl *ctl, unsigned int *reg) return -EINVAL; } - *reg = wm_adsp_region_to_reg(mem, ctl->alg_region.base + ctl->offset); + *reg = dsp->ops->region_to_reg(mem, ctl->alg_region.base + ctl->offset); return 0; } @@ -1335,28 +1418,33 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, case 1: snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x", dsp->name, region_name, alg_region->alg); + subname = NULL; /* don't append subname */ break; - default: + case 2: ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s%c %.12s %x", dsp->name, *region_name, wm_adsp_fw_text[dsp->fw], alg_region->alg); + break; + default: + ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "%s %.12s %x", dsp->name, + wm_adsp_fw_text[dsp->fw], alg_region->alg); + break; + } - /* Truncate the subname from the start if it is too long */ - if (subname) { - int avail = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret - 2; - int skip = 0; + if (subname) { + int avail = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret - 2; + int skip = 0; - if (dsp->component->name_prefix) - avail -= strlen(dsp->component->name_prefix) + 1; + if (dsp->component->name_prefix) + avail -= strlen(dsp->component->name_prefix) + 1; - if (subname_len > avail) - skip = subname_len - avail; + /* Truncate the subname from the start if it is too long */ + if (subname_len > avail) + skip = subname_len - avail; - snprintf(name + ret, - SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret, " %.*s", - subname_len - skip, subname + skip); - } - break; + snprintf(name + ret, SNDRV_CTL_ELEM_ID_NAME_MAXLEN - ret, + " %.*s", subname_len - skip, subname + skip); } list_for_each_entry(ctl, &dsp->ctl_list, list) { @@ -1689,6 +1777,16 @@ static bool wm_adsp_validate_version(struct wm_adsp *dsp, unsigned int version) } } +static bool wm_halo_validate_version(struct wm_adsp *dsp, unsigned int version) +{ + switch (version) { + case 3: + return true; + default: + return false; + } +} + static int wm_adsp_load(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -1774,7 +1872,6 @@ static int wm_adsp_load(struct wm_adsp *dsp) text = NULL; offset = le32_to_cpu(region->offset) & 0xffffff; type = be32_to_cpu(region->type) & 0xff; - mem = wm_adsp_find_region(dsp, type); switch (type) { case WMFW_NAME_TEXT: @@ -1802,8 +1899,17 @@ static int wm_adsp_load(struct wm_adsp *dsp) case WMFW_ADSP2_XM: case WMFW_ADSP2_YM: case WMFW_ADSP1_ZM: + case WMFW_HALO_PM_PACKED: + case WMFW_HALO_XM_PACKED: + case WMFW_HALO_YM_PACKED: + mem = wm_adsp_find_region(dsp, type); + if (!mem) { + adsp_err(dsp, "No region of type: %x\n", type); + goto out_fw; + } + region_name = wm_adsp_mem_region_name(type); - reg = wm_adsp_region_to_reg(mem, offset); + reg = dsp->ops->region_to_reg(mem, offset); break; default: adsp_warn(dsp, @@ -1916,7 +2022,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, } /* Read the terminator first to validate the length */ - reg = wm_adsp_region_to_reg(mem, pos + len); + reg = dsp->ops->region_to_reg(mem, pos + len); ret = regmap_raw_read(dsp->regmap, reg, &val, sizeof(val)); if (ret != 0) { @@ -1936,7 +2042,7 @@ static void *wm_adsp_read_algs(struct wm_adsp *dsp, size_t n_algs, if (!alg) return ERR_PTR(-ENOMEM); - reg = wm_adsp_region_to_reg(mem, pos); + reg = dsp->ops->region_to_reg(mem, pos); ret = regmap_raw_read(dsp->regmap, reg, alg, len); if (ret != 0) { @@ -2008,6 +2114,35 @@ static void wmfw_parse_id_header(struct wm_adsp *dsp, nalgs); } +static void wmfw_v3_parse_id_header(struct wm_adsp *dsp, + struct wmfw_v3_id_hdr *fw, int nalgs) +{ + dsp->fw_id = be32_to_cpu(fw->id); + dsp->fw_id_version = be32_to_cpu(fw->ver); + dsp->fw_vendor_id = be32_to_cpu(fw->vendor_id); + + adsp_info(dsp, "Firmware: %x vendor: 0x%x v%d.%d.%d, %zu algorithms\n", + dsp->fw_id, dsp->fw_vendor_id, + (dsp->fw_id_version & 0xff0000) >> 16, + (dsp->fw_id_version & 0xff00) >> 8, dsp->fw_id_version & 0xff, + nalgs); +} + +static int wm_adsp_create_regions(struct wm_adsp *dsp, __be32 id, int nregions, + int *type, __be32 *base) +{ + struct wm_adsp_alg_region *alg_region; + int i; + + for (i = 0; i < nregions; i++) { + alg_region = wm_adsp_create_region(dsp, type[i], id, base[i]); + if (IS_ERR(alg_region)) + return PTR_ERR(alg_region); + } + + return 0; +} + static int wm_adsp1_setup_algs(struct wm_adsp *dsp) { struct wmfw_adsp1_id_hdr adsp1_id; @@ -2238,6 +2373,78 @@ out: return ret; } +static int wm_halo_create_regions(struct wm_adsp *dsp, __be32 id, + __be32 xm_base, __be32 ym_base) +{ + int types[] = { + WMFW_ADSP2_XM, WMFW_HALO_XM_PACKED, + WMFW_ADSP2_YM, WMFW_HALO_YM_PACKED + }; + __be32 bases[] = { xm_base, xm_base, ym_base, ym_base }; + + return wm_adsp_create_regions(dsp, id, ARRAY_SIZE(types), types, bases); +} + +static int wm_halo_setup_algs(struct wm_adsp *dsp) +{ + struct wmfw_halo_id_hdr halo_id; + struct wmfw_halo_alg_hdr *halo_alg; + const struct wm_adsp_region *mem; + unsigned int pos, len; + size_t n_algs; + int i, ret; + + mem = wm_adsp_find_region(dsp, WMFW_ADSP2_XM); + if (WARN_ON(!mem)) + return -EINVAL; + + ret = regmap_raw_read(dsp->regmap, mem->base, &halo_id, + sizeof(halo_id)); + if (ret != 0) { + adsp_err(dsp, "Failed to read algorithm info: %d\n", + ret); + return ret; + } + + n_algs = be32_to_cpu(halo_id.n_algs); + + wmfw_v3_parse_id_header(dsp, &halo_id.fw, n_algs); + + ret = wm_halo_create_regions(dsp, halo_id.fw.id, + halo_id.ym_base, halo_id.ym_base); + if (ret) + return ret; + + /* Calculate offset and length in DSP words */ + pos = sizeof(halo_id) / sizeof(u32); + len = (sizeof(*halo_alg) * n_algs) / sizeof(u32); + + halo_alg = wm_adsp_read_algs(dsp, n_algs, mem, pos, len); + if (IS_ERR(halo_alg)) + return PTR_ERR(halo_alg); + + for (i = 0; i < n_algs; i++) { + adsp_info(dsp, + "%d: ID %x v%d.%d.%d XM@%x YM@%x\n", + i, be32_to_cpu(halo_alg[i].alg.id), + (be32_to_cpu(halo_alg[i].alg.ver) & 0xff0000) >> 16, + (be32_to_cpu(halo_alg[i].alg.ver) & 0xff00) >> 8, + be32_to_cpu(halo_alg[i].alg.ver) & 0xff, + be32_to_cpu(halo_alg[i].xm_base), + be32_to_cpu(halo_alg[i].ym_base)); + + ret = wm_halo_create_regions(dsp, halo_alg[i].alg.id, + halo_alg[i].xm_base, + halo_alg[i].ym_base); + if (ret) + goto out; + } + +out: + kfree(halo_alg); + return ret; +} + static int wm_adsp_load_coeff(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -2332,7 +2539,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) adsp_err(dsp, "No ZM\n"); break; } - reg = wm_adsp_region_to_reg(mem, 0); + reg = dsp->ops->region_to_reg(mem, 0); } else { region_name = "register"; @@ -2344,6 +2551,9 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) case WMFW_ADSP1_ZM: case WMFW_ADSP2_XM: case WMFW_ADSP2_YM: + case WMFW_HALO_XM_PACKED: + case WMFW_HALO_YM_PACKED: + case WMFW_HALO_PM_PACKED: adsp_dbg(dsp, "%s.%d: %d bytes in %x for %x\n", file, blocks, le32_to_cpu(blk->len), type, le32_to_cpu(blk->id)); @@ -2358,7 +2568,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) le32_to_cpu(blk->id)); if (alg_region) { reg = alg_region->base; - reg = wm_adsp_region_to_reg(mem, reg); + reg = dsp->ops->region_to_reg(mem, reg); reg += offset; } else { adsp_err(dsp, "No %x for algorithm %x\n", @@ -2748,6 +2958,37 @@ err_mutex: mutex_unlock(&dsp->pwr_lock); } +static int wm_halo_configure_mpu(struct wm_adsp *dsp, unsigned int lock_regions) +{ + struct reg_sequence config[] = { + { dsp->base + HALO_MPU_LOCK_CONFIG, 0x5555 }, + { dsp->base + HALO_MPU_LOCK_CONFIG, 0xAAAA }, + { dsp->base + HALO_MPU_XMEM_ACCESS_0, 0xFFFFFFFF }, + { dsp->base + HALO_MPU_YMEM_ACCESS_0, 0xFFFFFFFF }, + { dsp->base + HALO_MPU_WINDOW_ACCESS_0, lock_regions }, + { dsp->base + HALO_MPU_XREG_ACCESS_0, lock_regions }, + { dsp->base + HALO_MPU_YREG_ACCESS_0, lock_regions }, + { dsp->base + HALO_MPU_XMEM_ACCESS_1, 0xFFFFFFFF }, + { dsp->base + HALO_MPU_YMEM_ACCESS_1, 0xFFFFFFFF }, + { dsp->base + HALO_MPU_WINDOW_ACCESS_1, lock_regions }, + { dsp->base + HALO_MPU_XREG_ACCESS_1, lock_regions }, + { dsp->base + HALO_MPU_YREG_ACCESS_1, lock_regions }, + { dsp->base + HALO_MPU_XMEM_ACCESS_2, 0xFFFFFFFF }, + { dsp->base + HALO_MPU_YMEM_ACCESS_2, 0xFFFFFFFF }, + { dsp->base + HALO_MPU_WINDOW_ACCESS_2, lock_regions }, + { dsp->base + HALO_MPU_XREG_ACCESS_2, lock_regions }, + { dsp->base + HALO_MPU_YREG_ACCESS_2, lock_regions }, + { dsp->base + HALO_MPU_XMEM_ACCESS_3, 0xFFFFFFFF }, + { dsp->base + HALO_MPU_YMEM_ACCESS_3, 0xFFFFFFFF }, + { dsp->base + HALO_MPU_WINDOW_ACCESS_3, lock_regions }, + { dsp->base + HALO_MPU_XREG_ACCESS_3, lock_regions }, + { dsp->base + HALO_MPU_YREG_ACCESS_3, lock_regions }, + { dsp->base + HALO_MPU_LOCK_CONFIG, 0 }, + }; + + return regmap_multi_reg_write(dsp->regmap, config, ARRAY_SIZE(config)); +} + int wm_adsp2_set_dspclk(struct snd_soc_dapm_widget *w, unsigned int freq) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); @@ -2970,6 +3211,23 @@ err: } EXPORT_SYMBOL_GPL(wm_adsp_event); +static int wm_halo_start_core(struct wm_adsp *dsp) +{ + return regmap_update_bits(dsp->regmap, + dsp->base + HALO_CCM_CORE_CONTROL, + HALO_CORE_EN, HALO_CORE_EN); +} + +static void wm_halo_stop_core(struct wm_adsp *dsp) +{ + regmap_update_bits(dsp->regmap, dsp->base + HALO_CCM_CORE_CONTROL, + HALO_CORE_EN, 0); + + /* reset halo core with CORE_SOFT_REEST */ + regmap_update_bits(dsp->regmap, dsp->base + HALO_CORE_SOFT_RESET, + HALO_CORE_SOFT_RESET_MASK, 1); +} + int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *component) { char preload[32]; @@ -3031,6 +3289,22 @@ int wm_adsp2_init(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_init); +int wm_halo_init(struct wm_adsp *dsp) +{ + int ret; + + ret = wm_adsp_common_init(dsp); + if (ret) + return ret; + + dsp->ops = &wm_halo_ops; + + INIT_WORK(&dsp->boot_work, wm_adsp_boot_work); + + return 0; +} +EXPORT_SYMBOL_GPL(wm_halo_init); + void wm_adsp2_remove(struct wm_adsp *dsp) { struct wm_coeff_ctl *ctl; @@ -3275,7 +3549,7 @@ static int wm_adsp_read_data_block(struct wm_adsp *dsp, int mem_type, if (!mem) return -EINVAL; - reg = wm_adsp_region_to_reg(mem, mem_addr); + reg = dsp->ops->region_to_reg(mem, mem_addr); ret = regmap_raw_read(dsp->regmap, reg, data, sizeof(*data) * num_words); @@ -3303,7 +3577,7 @@ static int wm_adsp_write_data_word(struct wm_adsp *dsp, int mem_type, if (!mem) return -EINVAL; - reg = wm_adsp_region_to_reg(mem, mem_addr); + reg = dsp->ops->region_to_reg(mem, mem_addr); data = cpu_to_be32(data & 0x00ffffffu); @@ -3414,7 +3688,7 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp) return -ENOMEM; alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); - xmalg = sizeof(struct wm_adsp_system_config_xm_hdr) / sizeof(__be32); + xmalg = dsp->ops->sys_config_size / sizeof(__be32); addr = alg_region->base + xmalg + ALG_XM_FIELD(magic); ret = wm_adsp_read_data_word(dsp, WMFW_ADSP2_XM, addr, &magic); @@ -4024,13 +4298,16 @@ EXPORT_SYMBOL_GPL(wm_adsp2_bus_error); struct wm_adsp_ops wm_adsp1_ops = { .validate_version = wm_adsp_validate_version, .parse_sizes = wm_adsp1_parse_sizes, + .region_to_reg = wm_adsp_region_to_reg, }; struct wm_adsp_ops wm_adsp2_ops[] = { { + .sys_config_size = sizeof(struct wm_adsp_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, .validate_version = wm_adsp_validate_version, .setup_algs = wm_adsp2_setup_algs, + .region_to_reg = wm_adsp_region_to_reg, .show_fw_status = wm_adsp2_show_fw_status, @@ -4045,9 +4322,11 @@ struct wm_adsp_ops wm_adsp2_ops[] = { }, { + .sys_config_size = sizeof(struct wm_adsp_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, .validate_version = wm_adsp_validate_version, .setup_algs = wm_adsp2_setup_algs, + .region_to_reg = wm_adsp_region_to_reg, .show_fw_status = wm_adsp2v2_show_fw_status, @@ -4062,9 +4341,11 @@ struct wm_adsp_ops wm_adsp2_ops[] = { .stop_core = wm_adsp2_stop_core, }, { + .sys_config_size = sizeof(struct wm_adsp_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, .validate_version = wm_adsp_validate_version, .setup_algs = wm_adsp2_setup_algs, + .region_to_reg = wm_adsp_region_to_reg, .show_fw_status = wm_adsp2v2_show_fw_status, .stop_watchdog = wm_adsp_stop_watchdog, @@ -4081,4 +4362,19 @@ struct wm_adsp_ops wm_adsp2_ops[] = { }, }; +struct wm_adsp_ops wm_halo_ops = { + .sys_config_size = sizeof(struct wm_halo_system_config_xm_hdr), + .parse_sizes = wm_adsp2_parse_sizes, + .validate_version = wm_halo_validate_version, + .setup_algs = wm_halo_setup_algs, + .region_to_reg = wm_halo_region_to_reg, + + .show_fw_status = wm_halo_show_fw_status, + + .lock_memory = wm_halo_configure_mpu, + + .start_core = wm_halo_start_core, + .stop_core = wm_halo_stop_core, +}; + MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index c75a671c19a1..e7f1fde2b54b 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -70,6 +70,7 @@ struct wm_adsp { struct wm_adsp_ops *ops; unsigned int base; + unsigned int base_sysinfo; unsigned int sysclk_reg; unsigned int sysclk_mask; unsigned int sysclk_shift; @@ -78,6 +79,7 @@ struct wm_adsp { unsigned int fw_id; unsigned int fw_id_version; + unsigned int fw_vendor_id; const struct wm_adsp_region *mem; int num_mems; @@ -110,12 +112,16 @@ struct wm_adsp { }; struct wm_adsp_ops { + unsigned int sys_config_size; + bool (*validate_version)(struct wm_adsp *dsp, unsigned int version); unsigned int (*parse_sizes)(struct wm_adsp *dsp, const char * const file, unsigned int pos, const struct firmware *firmware); int (*setup_algs)(struct wm_adsp *dsp); + unsigned int (*region_to_reg)(struct wm_adsp_region const *mem, + unsigned int offset); void (*show_fw_status)(struct wm_adsp *dsp); void (*stop_watchdog)(struct wm_adsp *dsp); @@ -160,6 +166,8 @@ int wm_adsp2_init(struct wm_adsp *dsp); void wm_adsp2_remove(struct wm_adsp *dsp); int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *component); int wm_adsp2_component_remove(struct wm_adsp *dsp, struct snd_soc_component *component); +int wm_halo_init(struct wm_adsp *dsp); + int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); diff --git a/sound/soc/codecs/wmfw.h b/sound/soc/codecs/wmfw.h index 0c3f50acb8b1..14b2d1a2fc59 100644 --- a/sound/soc/codecs/wmfw.h +++ b/sound/soc/codecs/wmfw.h @@ -73,6 +73,14 @@ struct wmfw_id_hdr { __be32 ver; } __packed; +struct wmfw_v3_id_hdr { + __be32 core_id; + __be32 block_rev; + __be32 vendor_id; + __be32 id; + __be32 ver; +} __packed; + struct wmfw_adsp1_id_hdr { struct wmfw_id_hdr fw; __be32 zm; @@ -88,6 +96,15 @@ struct wmfw_adsp2_id_hdr { __be32 n_algs; } __packed; +struct wmfw_halo_id_hdr { + struct wmfw_v3_id_hdr fw; + __be32 xm_base; + __be32 xm_size; + __be32 ym_base; + __be32 ym_size; + __be32 n_algs; +} __packed; + struct wmfw_alg_hdr { __be32 id; __be32 ver; @@ -106,6 +123,14 @@ struct wmfw_adsp2_alg_hdr { __be32 ym; } __packed; +struct wmfw_halo_alg_hdr { + struct wmfw_alg_hdr alg; + __be32 xm_base; + __be32 xm_size; + __be32 ym_base; + __be32 ym_size; +} __packed; + struct wmfw_adsp_alg_data { __le32 id; u8 name[WMFW_MAX_ALG_NAME]; @@ -154,6 +179,7 @@ struct wmfw_coeff_item { #define WMFW_ADSP1 1 #define WMFW_ADSP2 2 +#define WMFW_HALO 4 #define WMFW_ABSOLUTE 0xf0 #define WMFW_ALGORITHM_DATA 0xf2 @@ -169,4 +195,8 @@ struct wmfw_coeff_item { #define WMFW_ADSP2_XM 5 #define WMFW_ADSP2_YM 6 +#define WMFW_HALO_PM_PACKED 0x10 +#define WMFW_HALO_XM_PACKED 0x11 +#define WMFW_HALO_YM_PACKED 0x12 + #endif -- cgit v1.2.3-59-g8ed1b From 2ae581380e4da6b1b9eff37cb087e3788ec82c67 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 19 Mar 2019 11:52:14 +0000 Subject: ASoC: wm_adsp: Add HALO MPU fault logging A Halo Core DSP has a memory protection unit that can trap and signal memory access faults. This patch adds a function that dumps the fault information. The interrupt reaches the host via the parent codec interrupt controller so this fault function is exported to be called by the codec driver. Signed-off-by: Richard Fitzgerald Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 84 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 85 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bd31056db4f5..3e74cbb80d3c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -227,6 +227,12 @@ */ #define WM_ADSP_FW_EVENT_SHUTDOWN 0x000001 +/* + * HALO system info + */ +#define HALO_AHBM_WINDOW_DEBUG_0 0x02040 +#define HALO_AHBM_WINDOW_DEBUG_1 0x02044 + /* * HALO core */ @@ -260,8 +266,21 @@ #define HALO_MPU_WINDOW_ACCESS_3 0x43050 #define HALO_MPU_XREG_ACCESS_3 0x43054 #define HALO_MPU_YREG_ACCESS_3 0x4305C +#define HALO_MPU_XM_VIO_ADDR 0x43100 +#define HALO_MPU_XM_VIO_STATUS 0x43104 +#define HALO_MPU_YM_VIO_ADDR 0x43108 +#define HALO_MPU_YM_VIO_STATUS 0x4310C +#define HALO_MPU_PM_VIO_ADDR 0x43110 +#define HALO_MPU_PM_VIO_STATUS 0x43114 #define HALO_MPU_LOCK_CONFIG 0x43140 +/* + * HALO_AHBM_WINDOW_DEBUG_1 + */ +#define HALO_AHBM_CORE_ERR_ADDR_MASK 0x0fffff00 +#define HALO_AHBM_CORE_ERR_ADDR_SHIFT 8 +#define HALO_AHBM_FLAGS_ERR_MASK 0x000000ff + /* * HALO_CCM_CORE_CONTROL */ @@ -272,6 +291,15 @@ */ #define HALO_CORE_SOFT_RESET_MASK 0x00000001 +/* + * HALO_MPU_?M_VIO_STATUS + */ +#define HALO_MPU_VIO_STS_MASK 0x007e0000 +#define HALO_MPU_VIO_STS_SHIFT 17 +#define HALO_MPU_VIO_ERR_WR_MASK 0x00008000 +#define HALO_MPU_VIO_ERR_SRC_MASK 0x00007fff +#define HALO_MPU_VIO_ERR_SRC_SHIFT 0 + struct wm_adsp_ops wm_adsp1_ops; struct wm_adsp_ops wm_adsp2_ops[]; struct wm_adsp_ops wm_halo_ops; @@ -4295,6 +4323,62 @@ error: } EXPORT_SYMBOL_GPL(wm_adsp2_bus_error); +irqreturn_t wm_halo_bus_error(struct wm_adsp *dsp) +{ + struct regmap *regmap = dsp->regmap; + unsigned int fault[6]; + struct reg_sequence clear[] = { + { dsp->base + HALO_MPU_XM_VIO_STATUS, 0x0 }, + { dsp->base + HALO_MPU_YM_VIO_STATUS, 0x0 }, + { dsp->base + HALO_MPU_PM_VIO_STATUS, 0x0 }, + }; + int ret; + + mutex_lock(&dsp->pwr_lock); + + ret = regmap_read(regmap, dsp->base_sysinfo + HALO_AHBM_WINDOW_DEBUG_1, + fault); + if (ret) { + adsp_warn(dsp, "Failed to read AHB DEBUG_1: %d\n", ret); + goto exit_unlock; + } + + adsp_warn(dsp, "AHB: STATUS: 0x%x ADDR: 0x%x\n", + *fault & HALO_AHBM_FLAGS_ERR_MASK, + (*fault & HALO_AHBM_CORE_ERR_ADDR_MASK) >> + HALO_AHBM_CORE_ERR_ADDR_SHIFT); + + ret = regmap_read(regmap, dsp->base_sysinfo + HALO_AHBM_WINDOW_DEBUG_0, + fault); + if (ret) { + adsp_warn(dsp, "Failed to read AHB DEBUG_0: %d\n", ret); + goto exit_unlock; + } + + adsp_warn(dsp, "AHB: SYS_ADDR: 0x%x\n", *fault); + + ret = regmap_bulk_read(regmap, dsp->base + HALO_MPU_XM_VIO_ADDR, + fault, ARRAY_SIZE(fault)); + if (ret) { + adsp_warn(dsp, "Failed to read MPU fault info: %d\n", ret); + goto exit_unlock; + } + + adsp_warn(dsp, "XM: STATUS:0x%x ADDR:0x%x\n", fault[1], fault[0]); + adsp_warn(dsp, "YM: STATUS:0x%x ADDR:0x%x\n", fault[3], fault[2]); + adsp_warn(dsp, "PM: STATUS:0x%x ADDR:0x%x\n", fault[5], fault[4]); + + ret = regmap_multi_reg_write(dsp->regmap, clear, ARRAY_SIZE(clear)); + if (ret) + adsp_warn(dsp, "Failed to clear MPU status: %d\n", ret); + +exit_unlock: + mutex_unlock(&dsp->pwr_lock); + + return IRQ_HANDLED; +} +EXPORT_SYMBOL_GPL(wm_halo_bus_error); + struct wm_adsp_ops wm_adsp1_ops = { .validate_version = wm_adsp_validate_version, .parse_sizes = wm_adsp1_parse_sizes, diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index e7f1fde2b54b..521dccbf3a2e 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -175,6 +175,7 @@ int wm_adsp_early_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); irqreturn_t wm_adsp2_bus_error(struct wm_adsp *adsp); +irqreturn_t wm_halo_bus_error(struct wm_adsp *dsp); int wm_adsp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- cgit v1.2.3-59-g8ed1b From 8bc144f90b6c813b2888f34842eba6a06a588429 Mon Sep 17 00:00:00 2001 From: Stuart Henderson Date: Tue, 19 Mar 2019 11:52:15 +0000 Subject: ASoC: wm_adsp: Add WDT expiry IRQ handling support for Halo core Signed-off-by: Stuart Henderson Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 29 +++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3e74cbb80d3c..644aaf1d2740 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -242,6 +242,7 @@ #define HALO_SCRATCH4 0x005d8 #define HALO_CCM_CORE_CONTROL 0x41000 #define HALO_CORE_SOFT_RESET 0x00010 +#define HALO_WDT_CONTROL 0x47000 /* * HALO MPU banks @@ -291,6 +292,11 @@ */ #define HALO_CORE_SOFT_RESET_MASK 0x00000001 +/* + * HALO_WDT_CONTROL + */ +#define HALO_WDT_EN_MASK 0x00000001 + /* * HALO_MPU_?M_VIO_STATUS */ @@ -3083,6 +3089,12 @@ static void wm_adsp_stop_watchdog(struct wm_adsp *dsp) ADSP2_WDT_ENA_MASK, 0); } +static void wm_halo_stop_watchdog(struct wm_adsp *dsp) +{ + regmap_update_bits(dsp->regmap, dsp->base + HALO_WDT_CONTROL, + HALO_WDT_EN_MASK, 0); +} + int wm_adsp_early_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -4379,6 +4391,22 @@ exit_unlock: } EXPORT_SYMBOL_GPL(wm_halo_bus_error); +irqreturn_t wm_halo_wdt_expire(int irq, void *data) +{ + struct wm_adsp *dsp = data; + + mutex_lock(&dsp->pwr_lock); + + adsp_warn(dsp, "WDT Expiry Fault\n"); + wm_halo_stop_watchdog(dsp); + wm_adsp_fatal_error(dsp); + + mutex_unlock(&dsp->pwr_lock); + + return IRQ_HANDLED; +} +EXPORT_SYMBOL_GPL(wm_halo_wdt_expire); + struct wm_adsp_ops wm_adsp1_ops = { .validate_version = wm_adsp_validate_version, .parse_sizes = wm_adsp1_parse_sizes, @@ -4454,6 +4482,7 @@ struct wm_adsp_ops wm_halo_ops = { .region_to_reg = wm_halo_region_to_reg, .show_fw_status = wm_halo_show_fw_status, + .stop_watchdog = wm_halo_stop_watchdog, .lock_memory = wm_halo_configure_mpu, diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 521dccbf3a2e..3631c9200c5d 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -176,6 +176,7 @@ int wm_adsp_early_event(struct snd_soc_dapm_widget *w, irqreturn_t wm_adsp2_bus_error(struct wm_adsp *adsp); irqreturn_t wm_halo_bus_error(struct wm_adsp *dsp); +irqreturn_t wm_halo_wdt_expire(int irq, void *data); int wm_adsp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); -- cgit v1.2.3-59-g8ed1b From 0f91d24f2c78e24303594395db58a0e983606a8c Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Tue, 19 Mar 2019 15:00:14 +0100 Subject: ASoC: jz4740: Depend on MIPS instead of MACH_JZ4740 Depending on MACH_JZ4740 prevent us from creating a generic kernel that works on more than one MIPS board. Instead, we just depend on MIPS being set. Signed-off-by: Paul Cercueil Signed-off-by: Mark Brown --- sound/soc/jz4740/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig index 1a354a6b6e87..b3f9c41b4319 100644 --- a/sound/soc/jz4740/Kconfig +++ b/sound/soc/jz4740/Kconfig @@ -1,6 +1,6 @@ config SND_JZ4740_SOC tristate "SoC Audio for Ingenic JZ4740 SoC" - depends on MACH_JZ4740 || COMPILE_TEST + depends on MIPS || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- cgit v1.2.3-59-g8ed1b From bf31cbfbe25001036e1e096b1c260bf871766ea5 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Mon, 18 Mar 2019 20:37:44 -0700 Subject: ASoC: tlv320aic32x4: Break out clock setting into separate function Break the clock setting logic out from the main hw_params. It's rather large and unweildy and makes for a large function. This also better enables some of the following changes to the clock tree access in the driver. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 26 ++++++++++++++++++-------- 1 file changed, 18 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 96f1526cb258..1aa8f5aa4225 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -696,17 +696,13 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -static int aic32x4_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int aic32x4_setup_clocks(struct snd_soc_component *component, + unsigned int sample_rate, + unsigned int parent_rate) { - struct snd_soc_component *component = dai->component; - struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); - u8 iface1_reg = 0; - u8 dacsetup_reg = 0; int i; - i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); + i = aic32x4_get_divs(parent_rate, sample_rate); if (i < 0) { printk(KERN_ERR "aic32x4: sampling rate not supported\n"); return i; @@ -763,6 +759,20 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AIC32X4_BCLKN, AIC32X4_BCLK_MASK, aic32x4_divs[i].blck_N); + return 0; +} + +static int aic32x4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); + u8 iface1_reg = 0; + u8 dacsetup_reg = 0; + + aic32x4_setup_clocks(component, params_rate(params), aic32x4->sysclk); + switch (params_width(params)) { case 16: iface1_reg |= (AIC32X4_WORD_LEN_16BITS << -- cgit v1.2.3-59-g8ed1b From f29eec793d4d16a1019abac1ac0702d07adae78e Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Tue, 19 Mar 2019 13:11:02 +0100 Subject: ASoC: samsung: i2s: Don't use register read to determine slave/master mode We can simplify the code by caching the CPU DAI master/slave information rather than reading previously set register bit. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 27 +++++++++++---------------- 1 file changed, 11 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ab471d550d17..9722940da6a4 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -88,12 +88,6 @@ struct samsung_i2s_priv { struct platform_device *pdev; struct platform_device *pdev_sec; - /* Memory mapped SFR region */ - void __iomem *addr; - - /* Spinlock protecting access to the device's registers */ - spinlock_t lock; - /* Lock for cross interface checks */ spinlock_t pcm_lock; @@ -122,6 +116,15 @@ struct samsung_i2s_priv { /* The clock provider's data */ struct clk *clk_table[3]; struct clk_onecell_data clk_data; + + /* Spinlock protecting member fields below */ + spinlock_t lock; + + /* Memory mapped SFR region */ + void __iomem *addr; + + /* A flag indicating the I2S slave mode operation */ + bool slave_mode; }; /* Returns true if this is the 'overlay' stereo DAI */ @@ -130,15 +133,6 @@ static inline bool is_secondary(struct i2s_dai *i2s) return i2s->drv->id == SAMSUNG_I2S_ID_SECONDARY; } -/* If operating in SoC-Slave mode */ -static inline bool is_slave(struct i2s_dai *i2s) -{ - struct samsung_i2s_priv *priv = i2s->priv; - - u32 mod = readl(priv->addr + I2SMOD); - return (mod & (1 << priv->variant_regs->mss_off)) ? true : false; -} - /* If this interface of the controller is transmitting data */ static inline bool tx_active(struct i2s_dai *i2s) { @@ -715,6 +709,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) mod &= ~(sdf_mask | lrp_rlow | mod_slave); mod |= tmp; writel(mod, priv->addr + I2SMOD); + priv->slave_mode = (mod & mod_slave); spin_unlock_irqrestore(&priv->lock, flags); pm_runtime_put(dai->dev); @@ -917,7 +912,7 @@ static int config_setup(struct i2s_dai *i2s) set_rfs(i2s, rfs); /* Don't bother with PSR in Slave mode */ - if (is_slave(i2s)) + if (priv->slave_mode) return 0; if (!(priv->quirks & QUIRK_NO_MUXPSR)) { -- cgit v1.2.3-59-g8ed1b From cd537873729cf939996a6984df080e6a0dfac0fb Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 19 Mar 2019 17:43:09 +0000 Subject: ASoC: wm_adsp: Correct some minor issues The wm_adsp_ops structures should be static and correct two printf specifiers. Fixes: 170b1e123f38 ("ASoC: wm_adsp: Add support for new Halo core DSPs") Fixes: 4e08d50d1fb6 ("ASoC: wm_adsp: Factor out DSP specific operations") Reported-by: kbuild test robot Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 644aaf1d2740..5e615fbddf9a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -306,9 +306,9 @@ #define HALO_MPU_VIO_ERR_SRC_MASK 0x00007fff #define HALO_MPU_VIO_ERR_SRC_SHIFT 0 -struct wm_adsp_ops wm_adsp1_ops; -struct wm_adsp_ops wm_adsp2_ops[]; -struct wm_adsp_ops wm_halo_ops; +static struct wm_adsp_ops wm_adsp1_ops; +static struct wm_adsp_ops wm_adsp2_ops[]; +static struct wm_adsp_ops wm_halo_ops; struct wm_adsp_buf { struct list_head list; @@ -2142,7 +2142,7 @@ static void wmfw_parse_id_header(struct wm_adsp *dsp, dsp->fw_id = be32_to_cpu(fw->id); dsp->fw_id_version = be32_to_cpu(fw->ver); - adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", + adsp_info(dsp, "Firmware: %x v%d.%d.%d, %d algorithms\n", dsp->fw_id, (dsp->fw_id_version & 0xff0000) >> 16, (dsp->fw_id_version & 0xff00) >> 8, dsp->fw_id_version & 0xff, nalgs); @@ -2155,7 +2155,7 @@ static void wmfw_v3_parse_id_header(struct wm_adsp *dsp, dsp->fw_id_version = be32_to_cpu(fw->ver); dsp->fw_vendor_id = be32_to_cpu(fw->vendor_id); - adsp_info(dsp, "Firmware: %x vendor: 0x%x v%d.%d.%d, %zu algorithms\n", + adsp_info(dsp, "Firmware: %x vendor: 0x%x v%d.%d.%d, %d algorithms\n", dsp->fw_id, dsp->fw_vendor_id, (dsp->fw_id_version & 0xff0000) >> 16, (dsp->fw_id_version & 0xff00) >> 8, dsp->fw_id_version & 0xff, @@ -4407,13 +4407,13 @@ irqreturn_t wm_halo_wdt_expire(int irq, void *data) } EXPORT_SYMBOL_GPL(wm_halo_wdt_expire); -struct wm_adsp_ops wm_adsp1_ops = { +static struct wm_adsp_ops wm_adsp1_ops = { .validate_version = wm_adsp_validate_version, .parse_sizes = wm_adsp1_parse_sizes, .region_to_reg = wm_adsp_region_to_reg, }; -struct wm_adsp_ops wm_adsp2_ops[] = { +static struct wm_adsp_ops wm_adsp2_ops[] = { { .sys_config_size = sizeof(struct wm_adsp_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, @@ -4474,7 +4474,7 @@ struct wm_adsp_ops wm_adsp2_ops[] = { }, }; -struct wm_adsp_ops wm_halo_ops = { +static struct wm_adsp_ops wm_halo_ops = { .sys_config_size = sizeof(struct wm_halo_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, .validate_version = wm_halo_validate_version, -- cgit v1.2.3-59-g8ed1b From d151cf89a47666622425ff093ee97e15587acd9a Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 4 Mar 2019 15:52:45 +0100 Subject: ASoC: stm32: dfsdm: add mmap support Enable direct access (mmap) support. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 47901983a6ff..1f20ff1ccb57 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -41,7 +41,7 @@ struct stm32_adfsdm_priv { static const struct snd_pcm_hardware stm32_adfsdm_pcm_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE, + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_PAUSE, .formats = SNDRV_PCM_FMTBIT_S32_LE, .rate_min = 8000, -- cgit v1.2.3-59-g8ed1b From d90ba6c8b53e541913a181638c353cf7a0856256 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 19 Mar 2019 17:49:31 +0000 Subject: ASoC: da7219: Expose BCLK and WCLK control through CCF For the purposes of platforms which use the codec as DAI clock master for the CPU and other codec devices, there is the need to not only expose the clock gating of BCLK and WCLK but also the ability to set those rates without going through the ASoC APIs. To make this possible, the previous CCF implementation in the driver has been extended to separate BCLK and WCLK out. WCLK is the parent clock to BCLK, and is also the clock gate for both. BCLK in HW is a factor/multiplier of WCLK so derives from whatever SR is chosen for WCLK, hence the need to make it a child of WCLK for the purposes of CCF. Enabling/disabling either BCLK or WCLK will result in clocks being ungated/gated accordingly. To simplify matters, these clocks can only be configured if the codec is set as master, otherwise CCF control is disallowed. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- include/sound/da7219.h | 8 +- sound/soc/codecs/da7219.c | 523 +++++++++++++++++++++++++++++++++++----------- sound/soc/codecs/da7219.h | 6 +- 3 files changed, 407 insertions(+), 130 deletions(-) (limited to 'sound') diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16f2d10..4a36954c86c5 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -33,10 +33,16 @@ enum da7219_mic_amp_in_sel { struct da7219_aad_pdata; +enum da7219_dai_clks { + DA7219_DAI_WCLK_IDX = 0, + DA7219_DAI_BCLK_IDX, + DA7219_DAI_NUM_CLKS, +}; + struct da7219_pdata { bool wakeup_source; - const char *dai_clks_name; + const char *dai_clk_names[DA7219_DAI_NUM_CLKS]; /* Mic */ enum da7219_micbias_voltage micbias_lvl; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 121a8190f93e..5f5fa3416af3 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -797,6 +797,7 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + struct clk *bclk = da7219->dai_clks[DA7219_DAI_BCLK_IDX]; u8 pll_ctrl, pll_status; int i = 0, ret; bool srm_lock = false; @@ -805,11 +806,11 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMU: if (da7219->master) { /* Enable DAI clks for master mode */ - if (da7219->dai_clks) { - ret = clk_prepare_enable(da7219->dai_clks); + if (bclk) { + ret = clk_prepare_enable(bclk); if (ret) { dev_err(component->dev, - "Failed to enable dai_clks\n"); + "Failed to enable DAI clks\n"); return ret; } } else { @@ -852,8 +853,8 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, /* Disable DAI clks if in master mode */ if (da7219->master) { - if (da7219->dai_clks) - clk_disable_unprepare(da7219->dai_clks); + if (bclk) + clk_disable_unprepare(bclk); else snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, @@ -1385,17 +1386,50 @@ static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } +static int da7219_set_bclks_per_wclk(struct snd_soc_component *component, + unsigned long factor) +{ + u8 bclks_per_wclk; + + switch (factor) { + case 32: + bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; + break; + case 64: + bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; + break; + case 128: + bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128; + break; + case 256: + bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + bclks_per_wclk); + + return 0; +} + static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + struct clk *wclk = da7219->dai_clks[DA7219_DAI_WCLK_IDX]; + struct clk *bclk = da7219->dai_clks[DA7219_DAI_BCLK_IDX]; unsigned int ch_mask; - u8 dai_bclks_per_wclk, slot_offset; + unsigned long sr, bclk_rate; + u8 slot_offset; u16 offset; __le16 dai_offset; u32 frame_size; + int ret; /* No channels enabled so disable TDM */ if (!tx_mask) { @@ -1432,28 +1466,26 @@ static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, */ if (da7219->master) { frame_size = slots * slot_width; - switch (frame_size) { - case 32: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; - break; - case 64: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; - break; - case 128: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128; - break; - case 256: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256; - break; - default: - dev_err(component->dev, "Invalid frame size %d\n", - frame_size); - return -EINVAL; - } - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, - DA7219_DAI_BCLKS_PER_WCLK_MASK, - dai_bclks_per_wclk); + if (bclk) { + sr = clk_get_rate(wclk); + bclk_rate = sr * frame_size; + ret = clk_set_rate(bclk, bclk_rate); + if (ret) { + dev_err(component->dev, + "Failed to set TDM BCLK rate %lu: %d\n", + bclk_rate, ret); + return ret; + } + } else { + ret = da7219_set_bclks_per_wclk(component, frame_size); + if (ret) { + dev_err(component->dev, + "Failed to set TDM BCLKs per WCLK %d: %d\n", + frame_size, ret); + return ret; + } + } } dai_offset = cpu_to_le16(offset); @@ -1471,44 +1503,12 @@ static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static int da7219_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int da7219_set_sr(struct snd_soc_component *component, + unsigned long rate) { - struct snd_soc_component *component = dai->component; - struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); - u8 dai_ctrl = 0, dai_bclks_per_wclk = 0, fs; - unsigned int channels; - int word_len = params_width(params); - int frame_size; - - switch (word_len) { - case 16: - dai_ctrl |= DA7219_DAI_WORD_LENGTH_S16_LE; - break; - case 20: - dai_ctrl |= DA7219_DAI_WORD_LENGTH_S20_LE; - break; - case 24: - dai_ctrl |= DA7219_DAI_WORD_LENGTH_S24_LE; - break; - case 32: - dai_ctrl |= DA7219_DAI_WORD_LENGTH_S32_LE; - break; - default: - return -EINVAL; - } - - channels = params_channels(params); - if ((channels < 1) || (channels > DA7219_DAI_CH_NUM_MAX)) { - dev_err(component->dev, - "Invalid number of channels, only 1 to %d supported\n", - DA7219_DAI_CH_NUM_MAX); - return -EINVAL; - } - dai_ctrl |= channels << DA7219_DAI_CH_NUM_SHIFT; + u8 fs; - switch (params_rate(params)) { + switch (rate) { case 8000: fs = DA7219_SR_8000; break; @@ -1546,28 +1546,103 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + snd_soc_component_write(component, DA7219_SR, fs); + + return 0; +} + +static int da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + struct clk *wclk = da7219->dai_clks[DA7219_DAI_WCLK_IDX]; + struct clk *bclk = da7219->dai_clks[DA7219_DAI_BCLK_IDX]; + u8 dai_ctrl = 0; + unsigned int channels; + unsigned long sr, bclk_rate; + int word_len = params_width(params); + int frame_size, ret; + + switch (word_len) { + case 16: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S16_LE; + break; + case 20: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S20_LE; + break; + case 24: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S24_LE; + break; + case 32: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S32_LE; + break; + default: + return -EINVAL; + } + + channels = params_channels(params); + if ((channels < 1) || (channels > DA7219_DAI_CH_NUM_MAX)) { + dev_err(component->dev, + "Invalid number of channels, only 1 to %d supported\n", + DA7219_DAI_CH_NUM_MAX); + return -EINVAL; + } + dai_ctrl |= channels << DA7219_DAI_CH_NUM_SHIFT; + + sr = params_rate(params); + if (da7219->master && wclk) { + ret = clk_set_rate(wclk, sr); + if (ret) { + dev_err(component->dev, + "Failed to set WCLK SR %lu: %d\n", sr, ret); + return ret; + } + } else { + ret = da7219_set_sr(component, sr); + if (ret) { + dev_err(component->dev, + "Failed to set SR %lu: %d\n", sr, ret); + return ret; + } + } + /* * If we're master, then we have a limited set of BCLK rates we * support. For slave mode this isn't the case and the codec can detect * the BCLK rate automatically. */ if (da7219->master && !da7219->tdm_en) { - frame_size = word_len * 2; - if (frame_size <= 32) - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; + if ((word_len * DA7219_DAI_CH_NUM_MAX) <= 32) + frame_size = 32; else - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; - - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, - DA7219_DAI_BCLKS_PER_WCLK_MASK, - dai_bclks_per_wclk); + frame_size = 64; + + if (bclk) { + bclk_rate = frame_size * sr; + ret = clk_set_rate(bclk, bclk_rate); + if (ret) { + dev_err(component->dev, + "Failed to set BCLK rate %lu: %d\n", + bclk_rate, ret); + return ret; + } + } else { + ret = da7219_set_bclks_per_wclk(component, frame_size); + if (ret) { + dev_err(component->dev, + "Failed to set BCLKs per WCLK %d: %d\n", + frame_size, ret); + return ret; + } + } } snd_soc_component_update_bits(component, DA7219_DAI_CTRL, DA7219_DAI_WORD_LENGTH_MASK | DA7219_DAI_CH_NUM_MASK, dai_ctrl); - snd_soc_component_write(component, DA7219_SR, fs); return 0; } @@ -1672,11 +1747,14 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone pdata->wakeup_source = device_property_read_bool(dev, "wakeup-source"); - pdata->dai_clks_name = "da7219-dai-clks"; - if (device_property_read_string(dev, "clock-output-names", - &pdata->dai_clks_name)) - dev_warn(dev, "Using default clk name: %s\n", - pdata->dai_clks_name); + pdata->dai_clk_names[DA7219_DAI_WCLK_IDX] = "da7219-dai-wclk"; + pdata->dai_clk_names[DA7219_DAI_BCLK_IDX] = "da7219-dai-bclk"; + if (device_property_read_string_array(dev, "clock-output-names", + pdata->dai_clk_names, + DA7219_DAI_NUM_CLKS) < 0) + dev_warn(dev, "Using default DAI clk names: %s, %s\n", + pdata->dai_clk_names[DA7219_DAI_WCLK_IDX], + pdata->dai_clk_names[DA7219_DAI_BCLK_IDX]); if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0) pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32); @@ -1793,12 +1871,16 @@ static int da7219_handle_supplies(struct snd_soc_component *component) } #ifdef CONFIG_COMMON_CLK -static int da7219_dai_clks_prepare(struct clk_hw *hw) +static int da7219_wclk_prepare(struct clk_hw *hw) { struct da7219_priv *da7219 = - container_of(hw, struct da7219_priv, dai_clks_hw); + container_of(hw, struct da7219_priv, + dai_clks_hw[DA7219_DAI_WCLK_IDX]); struct snd_soc_component *component = da7219->component; + if (!da7219->master) + return -EINVAL; + snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_CLK_EN_MASK, DA7219_DAI_CLK_EN_MASK); @@ -1806,36 +1888,48 @@ static int da7219_dai_clks_prepare(struct clk_hw *hw) return 0; } -static void da7219_dai_clks_unprepare(struct clk_hw *hw) +static void da7219_wclk_unprepare(struct clk_hw *hw) { struct da7219_priv *da7219 = - container_of(hw, struct da7219_priv, dai_clks_hw); + container_of(hw, struct da7219_priv, + dai_clks_hw[DA7219_DAI_WCLK_IDX]); struct snd_soc_component *component = da7219->component; + if (!da7219->master) + return; + snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, DA7219_DAI_CLK_EN_MASK, 0); } -static int da7219_dai_clks_is_prepared(struct clk_hw *hw) +static int da7219_wclk_is_prepared(struct clk_hw *hw) { struct da7219_priv *da7219 = - container_of(hw, struct da7219_priv, dai_clks_hw); + container_of(hw, struct da7219_priv, + dai_clks_hw[DA7219_DAI_WCLK_IDX]); struct snd_soc_component *component = da7219->component; u8 clk_reg; + if (!da7219->master) + return -EINVAL; + clk_reg = snd_soc_component_read32(component, DA7219_DAI_CLK_MODE); return !!(clk_reg & DA7219_DAI_CLK_EN_MASK); } -static unsigned long da7219_dai_clks_recalc_rate(struct clk_hw *hw, - unsigned long parent_rate) +static unsigned long da7219_wclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) { struct da7219_priv *da7219 = - container_of(hw, struct da7219_priv, dai_clks_hw); + container_of(hw, struct da7219_priv, + dai_clks_hw[DA7219_DAI_WCLK_IDX]); struct snd_soc_component *component = da7219->component; u8 fs = snd_soc_component_read32(component, DA7219_SR); + if (!da7219->master) + return 0; + switch (fs & DA7219_SR_MASK) { case DA7219_SR_8000: return 8000; @@ -1864,11 +1958,151 @@ static unsigned long da7219_dai_clks_recalc_rate(struct clk_hw *hw, } } -static const struct clk_ops da7219_dai_clks_ops = { - .prepare = da7219_dai_clks_prepare, - .unprepare = da7219_dai_clks_unprepare, - .is_prepared = da7219_dai_clks_is_prepared, - .recalc_rate = da7219_dai_clks_recalc_rate, +static long da7219_wclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct da7219_priv *da7219 = + container_of(hw, struct da7219_priv, + dai_clks_hw[DA7219_DAI_WCLK_IDX]); + + if (!da7219->master) + return -EINVAL; + + if (rate < 11025) + return 8000; + else if (rate < 12000) + return 11025; + else if (rate < 16000) + return 12000; + else if (rate < 22050) + return 16000; + else if (rate < 24000) + return 22050; + else if (rate < 32000) + return 24000; + else if (rate < 44100) + return 32000; + else if (rate < 48000) + return 44100; + else if (rate < 88200) + return 48000; + else if (rate < 96000) + return 88200; + else + return 96000; +} + +static int da7219_wclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct da7219_priv *da7219 = + container_of(hw, struct da7219_priv, + dai_clks_hw[DA7219_DAI_WCLK_IDX]); + struct snd_soc_component *component = da7219->component; + + if (!da7219->master) + return -EINVAL; + + return da7219_set_sr(component, rate); +} + +static unsigned long da7219_bclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct da7219_priv *da7219 = + container_of(hw, struct da7219_priv, + dai_clks_hw[DA7219_DAI_BCLK_IDX]); + struct snd_soc_component *component = da7219->component; + u8 bclks_per_wclk = snd_soc_component_read32(component, + DA7219_DAI_CLK_MODE); + + if (!da7219->master) + return 0; + + switch (bclks_per_wclk & DA7219_DAI_BCLKS_PER_WCLK_MASK) { + case DA7219_DAI_BCLKS_PER_WCLK_32: + return parent_rate * 32; + case DA7219_DAI_BCLKS_PER_WCLK_64: + return parent_rate * 64; + case DA7219_DAI_BCLKS_PER_WCLK_128: + return parent_rate * 128; + case DA7219_DAI_BCLKS_PER_WCLK_256: + return parent_rate * 256; + default: + return 0; + } +} + +static unsigned long da7219_bclk_get_factor(unsigned long rate, + unsigned long parent_rate) +{ + unsigned long factor; + + factor = rate / parent_rate; + if (factor < 64) + return 32; + else if (factor < 128) + return 64; + else if (factor < 256) + return 128; + else + return 256; +} + +static long da7219_bclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct da7219_priv *da7219 = + container_of(hw, struct da7219_priv, + dai_clks_hw[DA7219_DAI_BCLK_IDX]); + unsigned long factor; + + if (!*parent_rate || !da7219->master) + return -EINVAL; + + /* + * We don't allow changing the parent rate as some BCLK rates can be + * derived from multiple parent WCLK rates (BCLK rates are set as a + * multiplier of WCLK in HW). We just do some rounding down based on the + * parent WCLK rate set and find the appropriate multiplier of BCLK to + * get the rounded down BCLK value. + */ + factor = da7219_bclk_get_factor(rate, *parent_rate); + + return *parent_rate * factor; +} + +static int da7219_bclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct da7219_priv *da7219 = + container_of(hw, struct da7219_priv, + dai_clks_hw[DA7219_DAI_BCLK_IDX]); + struct snd_soc_component *component = da7219->component; + unsigned long factor; + + if (!da7219->master) + return -EINVAL; + + factor = da7219_bclk_get_factor(rate, parent_rate); + + return da7219_set_bclks_per_wclk(component, factor); +} + +static const struct clk_ops da7219_dai_clk_ops[DA7219_DAI_NUM_CLKS] = { + [DA7219_DAI_WCLK_IDX] = { + .prepare = da7219_wclk_prepare, + .unprepare = da7219_wclk_unprepare, + .is_prepared = da7219_wclk_is_prepared, + .recalc_rate = da7219_wclk_recalc_rate, + .round_rate = da7219_wclk_round_rate, + .set_rate = da7219_wclk_set_rate, + }, + [DA7219_DAI_BCLK_IDX] = { + .recalc_rate = da7219_bclk_recalc_rate, + .round_rate = da7219_bclk_round_rate, + .set_rate = da7219_bclk_set_rate, + }, }; static int da7219_register_dai_clks(struct snd_soc_component *component) @@ -1876,47 +2110,81 @@ static int da7219_register_dai_clks(struct snd_soc_component *component) struct device *dev = component->dev; struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); struct da7219_pdata *pdata = da7219->pdata; - struct clk_init_data init = {}; - struct clk *dai_clks; - struct clk_lookup *dai_clks_lookup; const char *parent_name; + int i, ret; - if (da7219->mclk) { - parent_name = __clk_get_name(da7219->mclk); - init.parent_names = &parent_name; - init.num_parents = 1; - } else { - init.parent_names = NULL; - init.num_parents = 0; - } + for (i = 0; i < DA7219_DAI_NUM_CLKS; ++i) { + struct clk_init_data init = {}; + struct clk *dai_clk; + struct clk_lookup *dai_clk_lookup; + struct clk_hw *dai_clk_hw = &da7219->dai_clks_hw[i]; - init.name = pdata->dai_clks_name; - init.ops = &da7219_dai_clks_ops; - init.flags = CLK_GET_RATE_NOCACHE; - da7219->dai_clks_hw.init = &init; + switch (i) { + case DA7219_DAI_WCLK_IDX: + /* + * If we can, make MCLK the parent of WCLK to ensure + * it's enabled as required. + */ + if (da7219->mclk) { + parent_name = __clk_get_name(da7219->mclk); + init.parent_names = &parent_name; + init.num_parents = 1; + } else { + init.parent_names = NULL; + init.num_parents = 0; + } + break; + case DA7219_DAI_BCLK_IDX: + /* Make WCLK the parent of BCLK */ + parent_name = __clk_get_name(da7219->dai_clks[DA7219_DAI_WCLK_IDX]); + init.parent_names = &parent_name; + init.num_parents = 1; + break; + default: + dev_err(dev, "Invalid clock index\n"); + ret = -EINVAL; + goto err; + } - dai_clks = devm_clk_register(dev, &da7219->dai_clks_hw); - if (IS_ERR(dai_clks)) { - dev_warn(dev, "Failed to register DAI clocks: %ld\n", - PTR_ERR(dai_clks)); - return PTR_ERR(dai_clks); - } - da7219->dai_clks = dai_clks; + init.name = pdata->dai_clk_names[i]; + init.ops = &da7219_dai_clk_ops[i]; + init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE; + dai_clk_hw->init = &init; + + dai_clk = devm_clk_register(dev, dai_clk_hw); + if (IS_ERR(dai_clk)) { + dev_warn(dev, "Failed to register %s: %ld\n", + init.name, PTR_ERR(dai_clk)); + ret = PTR_ERR(dai_clk); + goto err; + } + da7219->dai_clks[i] = dai_clk; - /* If we're using DT, then register as provider accordingly */ - if (dev->of_node) { - devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, - &da7219->dai_clks_hw); - } else { - dai_clks_lookup = clkdev_create(dai_clks, pdata->dai_clks_name, - "%s", dev_name(dev)); - if (!dai_clks_lookup) - return -ENOMEM; - else - da7219->dai_clks_lookup = dai_clks_lookup; + /* If we're using DT, then register as provider accordingly */ + if (dev->of_node) { + devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, + dai_clk_hw); + } else { + dai_clk_lookup = clkdev_create(dai_clk, init.name, + "%s", dev_name(dev)); + if (!dai_clk_lookup) { + ret = -ENOMEM; + goto err; + } else { + da7219->dai_clks_lookup[i] = dai_clk_lookup; + } + } } return 0; + +err: + do { + if (da7219->dai_clks_lookup[i]) + clkdev_drop(da7219->dai_clks_lookup[i]); + } while (i-- > 0); + + return ret; } #else static inline int da7219_register_dai_clks(struct snd_soc_component *component) @@ -2080,12 +2348,15 @@ err_disable_reg: static void da7219_remove(struct snd_soc_component *component) { struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + int i; da7219_aad_exit(component); #ifdef CONFIG_COMMON_CLK - if (da7219->dai_clks_lookup) - clkdev_drop(da7219->dai_clks_lookup); + for (i = DA7219_DAI_NUM_CLKS - 1; i >= 0; --i) { + if (da7219->dai_clks_lookup[i]) + clkdev_drop(da7219->dai_clks_lookup[i]); + } #endif /* Supplies */ diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 018819c631fb..f3b180bc986f 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -820,10 +820,10 @@ struct da7219_priv { struct mutex pll_lock; #ifdef CONFIG_COMMON_CLK - struct clk_hw dai_clks_hw; + struct clk_hw dai_clks_hw[DA7219_DAI_NUM_CLKS]; #endif - struct clk_lookup *dai_clks_lookup; - struct clk *dai_clks; + struct clk_lookup *dai_clks_lookup[DA7219_DAI_NUM_CLKS]; + struct clk *dai_clks[DA7219_DAI_NUM_CLKS]; struct clk *mclk; unsigned int mclk_rate; -- cgit v1.2.3-59-g8ed1b From 883149027a989dc121ad277785a4b4a901b7fd31 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 19 Mar 2019 17:49:32 +0000 Subject: ASoC: AMD: Update DA7219 DAI clock name to align with codec updates As the DAI clocks for DA7219 have now been split into BCLK and WCLK, the clock lookup name needs to be udpated here to select BCLK to achieve the same functionality as before with regards to DAI clock gating. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index a5daad973ce5..f37a588ba345 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -72,7 +72,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks"); + da7219_dai_clk = clk_get(component->dev, "da7219-dai-bclk"); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | -- cgit v1.2.3-59-g8ed1b From 0580dde59438686d60762b6da9229ebec693b94f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:54:42 +0900 Subject: ASoC: simple-card-utils: add asoc_simple_debug_info() Current simple-card-utils has dev_dbg(), but people want to add #define DEBUG at simple-card/audio-graph, not simple-card-utils. And, people want to get all information. This patch adds new asoc_simple_debug_info() to indicates information. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 72 +++++++++++++++++++++++++++++++++++ sound/soc/generic/audio-graph-card.c | 2 + sound/soc/generic/simple-card-utils.c | 12 ------ sound/soc/generic/simple-card.c | 2 + 4 files changed, 76 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 7afe45389972..0199ad0bbcd0 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -129,4 +129,76 @@ int asoc_simple_card_init_jack(struct snd_soc_card *card, struct asoc_simple_jack *sjack, int is_hp, char *prefix); + +#ifdef DEBUG +inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv, + char *name, + struct asoc_simple_dai *dai) +{ + struct device *dev = simple_priv_to_dev(priv); + + if (dai->name) + dev_dbg(dev, "%s dai name = %s\n", + name, dai->name); + if (dai->sysclk) + dev_dbg(dev, "%s sysclk = %d\n", + name, dai->sysclk); + + dev_dbg(dev, "%s direction = %s\n", + name, dai->clk_direction ? "OUT" : "IN"); + + if (dai->slots) + dev_dbg(dev, "%s slots = %d\n", name, dai->slots); + if (dai->slot_width) + dev_dbg(dev, "%s slot width = %d\n", name, dai->slot_width); + if (dai->tx_slot_mask) + dev_dbg(dev, "%s tx slot mask = %d\n", name, dai->tx_slot_mask); + if (dai->rx_slot_mask) + dev_dbg(dev, "%s rx slot mask = %d\n", name, dai->rx_slot_mask); + if (dai->clk) + dev_dbg(dev, "%s clk %luHz\n", name, clk_get_rate(dai->clk)); +} + +inline void asoc_simple_debug_info(struct asoc_simple_priv *priv) +{ + struct snd_soc_card *card = simple_priv_to_card(priv); + struct device *dev = simple_priv_to_dev(priv); + + int i; + + if (card->name) + dev_dbg(dev, "Card Name: %s\n", card->name); + + for (i = 0; i < card->num_links; i++) { + struct simple_dai_props *props = simple_priv_to_props(priv, i); + struct snd_soc_dai_link *link = simple_priv_to_link(priv, i); + + dev_dbg(dev, "DAI%d\n", i); + + asoc_simple_debug_dai(priv, "cpu", props->cpu_dai); + asoc_simple_debug_dai(priv, "codec", props->codec_dai); + + if (link->name) + dev_dbg(dev, "dai name = %s\n", link->name); + + dev_dbg(dev, "dai format = %04x\n", link->dai_fmt); + + if (props->adata.convert_rate) + dev_dbg(dev, "convert_rate = %d\n", + props->adata.convert_rate); + if (props->adata.convert_channels) + dev_dbg(dev, "convert_channels = %d\n", + props->adata.convert_channels); + if (props->codec_conf && props->codec_conf->name_prefix) + dev_dbg(dev, "name prefix = %s\n", + props->codec_conf->name_prefix); + if (props->mclk_fs) + dev_dbg(dev, "mclk-fs = %d\n", + props->mclk_fs); + } +} +#else +#define asoc_simple_debug_info(priv) +#endif /* DEBUG */ + #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 7dd772a895e2..2a95355540d1 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -712,6 +712,8 @@ static int graph_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); + asoc_simple_debug_info(priv); + ret = devm_snd_soc_register_card(dev, card); if (ret < 0) goto err; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 5c1424f03620..f5ba5ad300cc 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -49,9 +49,6 @@ void asoc_simple_card_parse_convert(struct device *dev, /* channels transfer */ snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-channels"); of_property_read_u32(np, prop, &data->convert_channels); - - dev_dbg(dev, "convert_rate %d\n", data->convert_rate); - dev_dbg(dev, "convert_channels %d\n", data->convert_channels); } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_convert); @@ -93,8 +90,6 @@ int asoc_simple_card_parse_daifmt(struct device *dev, *retfmt = daifmt; - dev_dbg(dev, "format : %04x\n", daifmt); - return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_daifmt); @@ -116,8 +111,6 @@ int asoc_simple_card_set_dailink_name(struct device *dev, dai_link->name = name; dai_link->stream_name = name; - - dev_dbg(dev, "name : %s\n", name); } return ret; @@ -146,8 +139,6 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, if (!card->name && card->dai_link) card->name = card->dai_link->name; - dev_dbg(card->dev, "Card Name: %s\n", card->name ? card->name : ""); - return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); @@ -211,9 +202,6 @@ int asoc_simple_card_parse_clk(struct device *dev, if (of_property_read_bool(node, "system-clock-direction-out")) simple_dai->clk_direction = SND_SOC_CLOCK_OUT; - dev_dbg(dev, "%s : sysclk = %d, direction %d\n", dai_name, - simple_dai->sysclk, simple_dai->clk_direction); - return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index b792638a11c8..77899624787e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -813,6 +813,8 @@ static int simple_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); + asoc_simple_debug_info(priv); + ret = devm_snd_soc_register_card(dev, card); if (ret < 0) goto err; -- cgit v1.2.3-59-g8ed1b From e59289cda8dec0153fa396864c8ba8092ec3b80d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:54:59 +0900 Subject: ASoC: simple_card_utils: share common priv for simple-card/audio-graph Historically, simple-card/simple-scu-card/audio-graph/audio-graph-scu are similar but different generic sound card. simple-scu-card which was for DPCM was merged into simple-card, and audio-graph-scu which was for DPCM was merged into audio-graph. simple-card is for non OF graph sound card, and audio-graph is for OF graph sound card. And, small detail difference (= function parameter, naming, etc) between simple-card/audio-graph has been unified. So today, the difference between simple-card/audio-graph are just using OF graph style, or not. In other words, there should no difference other than OF graph sytle. simple-card/audio-graph are using own priv today , but we can merge it. This patch merge it at simple_card_utils. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 23 +++++++++ sound/soc/generic/audio-graph-card.c | 96 ++++++++++++++---------------------- sound/soc/generic/simple-card.c | 57 +++++++-------------- 3 files changed, 76 insertions(+), 100 deletions(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 0199ad0bbcd0..c0161ec17572 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -37,6 +37,29 @@ struct asoc_simple_jack { struct snd_soc_jack_gpio gpio; }; +struct asoc_simple_priv { + struct snd_soc_card snd_card; + struct simple_dai_props { + struct asoc_simple_dai *cpu_dai; + struct asoc_simple_dai *codec_dai; + struct snd_soc_dai_link_component codecs; /* single codec */ + struct snd_soc_dai_link_component platforms; + struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; + unsigned int mclk_fs; + } *dai_props; + struct asoc_simple_jack hp_jack; + struct asoc_simple_jack mic_jack; + struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; + struct snd_soc_codec_conf *codec_conf; + struct gpio_desc *pa_gpio; +}; +#define simple_priv_to_card(priv) (&(priv)->snd_card) +#define simple_priv_to_props(priv, i) ((priv)->dai_props + (i)) +#define simple_priv_to_dev(priv) (simple_priv_to_card(priv)->dev) +#define simple_priv_to_link(priv, i) (simple_priv_to_card(priv)->dai_link + (i)) + int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 2a95355540d1..656ca49ba727 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -20,25 +20,6 @@ #include #include -struct graph_priv { - struct snd_soc_card snd_card; - struct graph_dai_props { - struct asoc_simple_dai *cpu_dai; - struct asoc_simple_dai *codec_dai; - struct snd_soc_dai_link_component codecs; /* single codec */ - struct snd_soc_dai_link_component platforms; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; - unsigned int mclk_fs; - } *dai_props; - struct asoc_simple_jack hp_jack; - struct asoc_simple_jack mic_jack; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dais; - struct snd_soc_codec_conf *codec_conf; - struct gpio_desc *pa_gpio; -}; - struct link_info { int dais; /* number of dai */ int link; /* number of link */ @@ -46,11 +27,6 @@ struct link_info { int cpu; /* turn for CPU / Codec */ }; -#define graph_priv_to_card(priv) (&(priv)->snd_card) -#define graph_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) -#define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) - #define PREFIX "audio-graph-card," static int graph_outdrv_event(struct snd_soc_dapm_widget *w, @@ -58,7 +34,7 @@ static int graph_outdrv_event(struct snd_soc_dapm_widget *w, int event) { struct snd_soc_dapm_context *dapm = w->dapm; - struct graph_priv *priv = snd_soc_card_get_drvdata(dapm->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(dapm->card); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -83,8 +59,8 @@ static const struct snd_soc_dapm_widget graph_dapm_widgets[] = { static int graph_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); @@ -101,8 +77,8 @@ static int graph_startup(struct snd_pcm_substream *substream) static void graph_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); asoc_simple_card_clk_disable(dai_props->cpu_dai); @@ -115,8 +91,8 @@ static int graph_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); unsigned int mclk, mclk_fs = 0; int ret = 0; @@ -148,8 +124,8 @@ static const struct snd_soc_ops graph_ops = { static int graph_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret = 0; ret = asoc_simple_card_init_dai(rtd->codec_dai, @@ -168,8 +144,8 @@ static int graph_dai_init(struct snd_soc_pcm_runtime *rtd) static int graph_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); asoc_simple_card_convert_fixup(&dai_props->adata, params); @@ -198,7 +174,7 @@ static void graph_parse_convert(struct device *dev, static void graph_parse_mclk_fs(struct device_node *top, struct device_node *ep, - struct graph_dai_props *props) + struct simple_dai_props *props) { struct device_node *port = of_get_parent(ep); struct device_node *ports = of_get_parent(port); @@ -214,15 +190,15 @@ static void graph_parse_mclk_fs(struct device_node *top, of_node_put(node); } -static int graph_dai_link_of_dpcm(struct graph_priv *priv, +static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li, int dup_codec) { - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, li->link); + struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct device_node *top = dev->of_node; struct device_node *ep = li->cpu ? cpu_ep : codec_ep; struct device_node *port; @@ -344,14 +320,14 @@ static int graph_dai_link_of_dpcm(struct graph_priv *priv, return 0; } -static int graph_dai_link_of(struct graph_priv *priv, +static int graph_dai_link_of(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li) { - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, li->link); + struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct device_node *top = dev->of_node; struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; @@ -420,19 +396,19 @@ static int graph_dai_link_of(struct graph_priv *priv, return 0; } -static int graph_for_each_link(struct graph_priv *priv, +static int graph_for_each_link(struct asoc_simple_priv *priv, struct link_info *li, - int (*func_noml)(struct graph_priv *priv, + int (*func_noml)(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li), - int (*func_dpcm)(struct graph_priv *priv, + int (*func_dpcm)(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li, int dup_codec)) { struct of_phandle_iterator it; - struct device *dev = graph_priv_to_dev(priv); + struct device *dev = simple_priv_to_dev(priv); struct device_node *node = dev->of_node; struct device_node *cpu_port; struct device_node *cpu_ep; @@ -488,9 +464,9 @@ static int graph_for_each_link(struct graph_priv *priv, return 0; } -static int graph_parse_of(struct graph_priv *priv) +static int graph_parse_of(struct asoc_simple_priv *priv) { - struct snd_soc_card *card = graph_priv_to_card(priv); + struct snd_soc_card *card = simple_priv_to_card(priv); struct link_info li; int ret; @@ -526,12 +502,12 @@ static int graph_parse_of(struct graph_priv *priv) return asoc_simple_card_parse_card_name(card, NULL); } -static int graph_count_noml(struct graph_priv *priv, +static int graph_count_noml(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li) { - struct device *dev = graph_priv_to_dev(priv); + struct device *dev = simple_priv_to_dev(priv); li->link += 1; /* 1xCPU-Codec */ li->dais += 2; /* 1xCPU + 1xCodec */ @@ -541,13 +517,13 @@ static int graph_count_noml(struct graph_priv *priv, return 0; } -static int graph_count_dpcm(struct graph_priv *priv, +static int graph_count_dpcm(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li, int dup_codec) { - struct device *dev = graph_priv_to_dev(priv); + struct device *dev = simple_priv_to_dev(priv); li->link++; /* 1xCPU-dummy */ li->dais++; /* 1xCPU */ @@ -563,10 +539,10 @@ static int graph_count_dpcm(struct graph_priv *priv, return 0; } -static void graph_get_dais_count(struct graph_priv *priv, +static void graph_get_dais_count(struct asoc_simple_priv *priv, struct link_info *li) { - struct device *dev = graph_priv_to_dev(priv); + struct device *dev = simple_priv_to_dev(priv); /* * link_num : number of links. @@ -623,7 +599,7 @@ static void graph_get_dais_count(struct graph_priv *priv, static int graph_card_probe(struct snd_soc_card *card) { - struct graph_priv *priv = snd_soc_card_get_drvdata(card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(card); int ret; ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL); @@ -639,9 +615,9 @@ static int graph_card_probe(struct snd_soc_card *card) static int graph_probe(struct platform_device *pdev) { - struct graph_priv *priv; + struct asoc_simple_priv *priv; struct snd_soc_dai_link *dai_link; - struct graph_dai_props *dai_props; + struct simple_dai_props *dai_props; struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; @@ -654,7 +630,7 @@ static int graph_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - card = graph_priv_to_card(priv); + card = simple_priv_to_card(priv); card->owner = THIS_MODULE; card->dev = dev; card->dapm_widgets = graph_dapm_widgets; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 77899624787e..628c19a3aaff 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -15,24 +15,6 @@ #include #include -struct simple_priv { - struct snd_soc_card snd_card; - struct simple_dai_props { - struct asoc_simple_dai *cpu_dai; - struct asoc_simple_dai *codec_dai; - struct snd_soc_dai_link_component codecs; /* single codec */ - struct snd_soc_dai_link_component platforms; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; - unsigned int mclk_fs; - } *dai_props; - struct asoc_simple_jack hp_jack; - struct asoc_simple_jack mic_jack; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dais; - struct snd_soc_codec_conf *codec_conf; -}; - struct link_info { int dais; /* number of dai */ int link; /* number of link */ @@ -40,11 +22,6 @@ struct link_info { int cpu; /* turn for CPU / Codec */ }; -#define simple_priv_to_card(priv) (&(priv)->snd_card) -#define simple_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define simple_priv_to_dev(priv) (simple_priv_to_card(priv)->dev) -#define simple_priv_to_link(priv, i) (simple_priv_to_card(priv)->dai_link + (i)) - #define DAI "sound-dai" #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," @@ -52,7 +29,7 @@ struct link_info { static int simple_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; @@ -71,7 +48,7 @@ static int simple_startup(struct snd_pcm_substream *substream) static void simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); @@ -101,7 +78,7 @@ static int simple_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); unsigned int mclk, mclk_fs = 0; @@ -144,7 +121,7 @@ static const struct snd_soc_ops simple_ops = { static int simple_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; @@ -164,7 +141,7 @@ static int simple_dai_init(struct snd_soc_pcm_runtime *rtd) static int simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); asoc_simple_card_convert_fixup(&dai_props->adata, params); @@ -207,7 +184,7 @@ static void simple_parse_mclk_fs(struct device_node *top, of_node_put(node); } -static int simple_dai_link_of_dpcm(struct simple_priv *priv, +static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, @@ -336,7 +313,7 @@ static int simple_dai_link_of_dpcm(struct simple_priv *priv, return 0; } -static int simple_dai_link_of(struct simple_priv *priv, +static int simple_dai_link_of(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, @@ -438,13 +415,13 @@ dai_link_of_err: return ret; } -static int simple_for_each_link(struct simple_priv *priv, +static int simple_for_each_link(struct asoc_simple_priv *priv, struct link_info *li, - int (*func_noml)(struct simple_priv *priv, + int (*func_noml)(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, bool is_top), - int (*func_dpcm)(struct simple_priv *priv, + int (*func_dpcm)(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, bool is_top)) @@ -513,7 +490,7 @@ static int simple_for_each_link(struct simple_priv *priv, } static int simple_parse_aux_devs(struct device_node *node, - struct simple_priv *priv) + struct asoc_simple_priv *priv) { struct device *dev = simple_priv_to_dev(priv); struct device_node *aux_node; @@ -543,7 +520,7 @@ static int simple_parse_aux_devs(struct device_node *node, return 0; } -static int simple_parse_of(struct simple_priv *priv) +static int simple_parse_of(struct asoc_simple_priv *priv) { struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; @@ -593,7 +570,7 @@ static int simple_parse_of(struct simple_priv *priv) return ret; } -static int simple_count_noml(struct simple_priv *priv, +static int simple_count_noml(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, bool is_top) @@ -605,7 +582,7 @@ static int simple_count_noml(struct simple_priv *priv, return 0; } -static int simple_count_dpcm(struct simple_priv *priv, +static int simple_count_dpcm(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, bool is_top) @@ -618,7 +595,7 @@ static int simple_count_dpcm(struct simple_priv *priv, return 0; } -static void simple_get_dais_count(struct simple_priv *priv, +static void simple_get_dais_count(struct asoc_simple_priv *priv, struct link_info *li) { struct device *dev = simple_priv_to_dev(priv); @@ -687,7 +664,7 @@ static void simple_get_dais_count(struct simple_priv *priv, static int simple_soc_probe(struct snd_soc_card *card) { - struct simple_priv *priv = snd_soc_card_get_drvdata(card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(card); int ret; ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX); @@ -703,7 +680,7 @@ static int simple_soc_probe(struct snd_soc_card *card) static int simple_probe(struct platform_device *pdev) { - struct simple_priv *priv; + struct asoc_simple_priv *priv; struct snd_soc_dai_link *dai_link; struct simple_dai_props *dai_props; struct asoc_simple_dai *dais; -- cgit v1.2.3-59-g8ed1b From f38df5bf0c9cb905fa9d5abc86c3a00128cdbba5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:55:14 +0900 Subject: ASoC: simple-card-utils: share asoc_simple_startup() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph have almost same functions. This patch shares asoc_simple_startup() between in these 2 drivers. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 20 +------------------- sound/soc/generic/simple-card-utils.c | 22 ++++++++++++++++++++-- sound/soc/generic/simple-card.c | 21 +-------------------- 4 files changed, 23 insertions(+), 42 deletions(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index c0161ec17572..0653fb9385d1 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -84,8 +84,8 @@ int asoc_simple_card_parse_clk(struct device *dev, struct asoc_simple_dai *simple_dai, const char *dai_name, struct snd_soc_dai_link_component *dlc); -int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai); void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); +int asoc_simple_startup(struct snd_pcm_substream *substream); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 656ca49ba727..29ae71b427ec 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -56,24 +56,6 @@ static const struct snd_soc_dapm_widget graph_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; -static int graph_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - int ret; - - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); - if (ret) - return ret; - - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); - if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - return ret; -} - static void graph_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -117,7 +99,7 @@ err: } static const struct snd_soc_ops graph_ops = { - .startup = graph_startup, + .startup = asoc_simple_startup, .shutdown = graph_shutdown, .hw_params = graph_hw_params, }; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index f5ba5ad300cc..83ea1afa3c72 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -143,14 +143,13 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); -int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) +static int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) { if (dai) return clk_prepare_enable(dai->clk); return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_clk_enable); void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) { @@ -206,6 +205,25 @@ int asoc_simple_card_parse_clk(struct device *dev, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); +int asoc_simple_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; + + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + if (ret) + return ret; + + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + if (ret) + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + return ret; +} +EXPORT_SYMBOL_GPL(asoc_simple_startup); + int asoc_simple_card_parse_dai(struct device_node *node, struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 628c19a3aaff..d0ea5736889e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -26,25 +26,6 @@ struct link_info { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -static int simple_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = - simple_priv_to_props(priv, rtd->num); - int ret; - - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); - if (ret) - return ret; - - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); - if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - return ret; -} - static void simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -114,7 +95,7 @@ err: } static const struct snd_soc_ops simple_ops = { - .startup = simple_startup, + .startup = asoc_simple_startup, .shutdown = simple_shutdown, .hw_params = simple_hw_params, }; -- cgit v1.2.3-59-g8ed1b From 686911b46fb5a08df142fe22b6c06dc6fbd3ba65 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:55:27 +0900 Subject: ASoC: simple-card-utils: share asoc_simple_shutdown() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph have almost same functions. This patch shares asoc_simple_shutdown() between in these 2 drivers. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 13 +------------ sound/soc/generic/simple-card-utils.c | 16 ++++++++++++++-- sound/soc/generic/simple-card.c | 14 +------------- 4 files changed, 17 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 0653fb9385d1..edda646c3ebe 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -84,8 +84,8 @@ int asoc_simple_card_parse_clk(struct device *dev, struct asoc_simple_dai *simple_dai, const char *dai_name, struct snd_soc_dai_link_component *dlc); -void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); int asoc_simple_startup(struct snd_pcm_substream *substream); +void asoc_simple_shutdown(struct snd_pcm_substream *substream); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 29ae71b427ec..f3577b5bf14c 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -56,17 +56,6 @@ static const struct snd_soc_dapm_widget graph_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; -static void graph_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - asoc_simple_card_clk_disable(dai_props->codec_dai); -} - static int graph_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -100,7 +89,7 @@ err: static const struct snd_soc_ops graph_ops = { .startup = asoc_simple_startup, - .shutdown = graph_shutdown, + .shutdown = asoc_simple_shutdown, .hw_params = graph_hw_params, }; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 83ea1afa3c72..7ede16347ea8 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -151,12 +151,11 @@ static int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) return 0; } -void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) +static void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) { if (dai) clk_disable_unprepare(dai->clk); } -EXPORT_SYMBOL_GPL(asoc_simple_card_clk_disable); int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, @@ -224,6 +223,19 @@ int asoc_simple_startup(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(asoc_simple_startup); +void asoc_simple_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = + simple_priv_to_props(priv, rtd->num); + + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + asoc_simple_card_clk_disable(dai_props->codec_dai); +} +EXPORT_SYMBOL_GPL(asoc_simple_shutdown); + int asoc_simple_card_parse_dai(struct device_node *node, struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d0ea5736889e..bb57c482eac8 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -26,18 +26,6 @@ struct link_info { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -static void simple_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = - simple_priv_to_props(priv, rtd->num); - - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - asoc_simple_card_clk_disable(dai_props->codec_dai); -} - static int simple_set_clk_rate(struct asoc_simple_dai *simple_dai, unsigned long rate) { @@ -96,7 +84,7 @@ err: static const struct snd_soc_ops simple_ops = { .startup = asoc_simple_startup, - .shutdown = simple_shutdown, + .shutdown = asoc_simple_shutdown, .hw_params = simple_hw_params, }; -- cgit v1.2.3-59-g8ed1b From f48dcbb6d47d870cf3a03f453c923dd262158c66 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:55:39 +0900 Subject: ASoC: simple-card-utils: share asoc_simple_hw_param() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph have almost same functions. This patch shares asoc_simple_hw_param() between in these 2 drivers. One note is that only simple-card supports simple_set_clk_rate() at hw_param from commit e9be4ffd4f40fcb ("ASoC: simple-card: set cpu dai clk in hw_params"). By this patch, audio-graph has same feature. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/audio-graph-card.c | 33 +------------------- sound/soc/generic/simple-card-utils.c | 57 ++++++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 58 +---------------------------------- 4 files changed, 61 insertions(+), 89 deletions(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index edda646c3ebe..89ae0841b711 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -86,6 +86,8 @@ int asoc_simple_card_parse_clk(struct device *dev, struct snd_soc_dai_link_component *dlc); int asoc_simple_startup(struct snd_pcm_substream *substream); void asoc_simple_shutdown(struct snd_pcm_substream *substream); +int asoc_simple_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index f3577b5bf14c..51160adaf5bb 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -56,41 +56,10 @@ static const struct snd_soc_dapm_widget graph_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; -static int graph_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - unsigned int mclk, mclk_fs = 0; - int ret = 0; - - if (dai_props->mclk_fs) - mclk_fs = dai_props->mclk_fs; - - if (mclk_fs) { - mclk = params_rate(params) * mclk_fs; - ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) - goto err; - - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, - SND_SOC_CLOCK_OUT); - if (ret && ret != -ENOTSUPP) - goto err; - } - return 0; -err: - return ret; -} - static const struct snd_soc_ops graph_ops = { .startup = asoc_simple_startup, .shutdown = asoc_simple_shutdown, - .hw_params = graph_hw_params, + .hw_params = asoc_simple_hw_params, }; static int graph_dai_init(struct snd_soc_pcm_runtime *rtd) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 7ede16347ea8..de23cf8bffbd 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -236,6 +236,63 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(asoc_simple_shutdown); +static int asoc_simple_set_clk_rate(struct asoc_simple_dai *simple_dai, + unsigned long rate) +{ + if (!simple_dai) + return 0; + + if (!simple_dai->clk) + return 0; + + if (clk_get_rate(simple_dai->clk) == rate) + return 0; + + return clk_set_rate(simple_dai->clk, rate); +} + +int asoc_simple_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = + simple_priv_to_props(priv, rtd->num); + unsigned int mclk, mclk_fs = 0; + int ret = 0; + + if (dai_props->mclk_fs) + mclk_fs = dai_props->mclk_fs; + + if (mclk_fs) { + mclk = params_rate(params) * mclk_fs; + + ret = asoc_simple_set_clk_rate(dai_props->codec_dai, mclk); + if (ret < 0) + return ret; + + ret = asoc_simple_set_clk_rate(dai_props->cpu_dai, mclk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) + goto err; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret && ret != -ENOTSUPP) + goto err; + } + return 0; +err: + return ret; +} +EXPORT_SYMBOL_GPL(asoc_simple_hw_params); + int asoc_simple_card_parse_dai(struct device_node *node, struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bb57c482eac8..7e836473eb11 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -26,66 +26,10 @@ struct link_info { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -static int simple_set_clk_rate(struct asoc_simple_dai *simple_dai, - unsigned long rate) -{ - if (!simple_dai) - return 0; - - if (!simple_dai->clk) - return 0; - - if (clk_get_rate(simple_dai->clk) == rate) - return 0; - - return clk_set_rate(simple_dai->clk, rate); -} - -static int simple_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = - simple_priv_to_props(priv, rtd->num); - unsigned int mclk, mclk_fs = 0; - int ret = 0; - - if (dai_props->mclk_fs) - mclk_fs = dai_props->mclk_fs; - - if (mclk_fs) { - mclk = params_rate(params) * mclk_fs; - - ret = simple_set_clk_rate(dai_props->codec_dai, mclk); - if (ret < 0) - return ret; - - ret = simple_set_clk_rate(dai_props->cpu_dai, mclk); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) - goto err; - - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, - SND_SOC_CLOCK_OUT); - if (ret && ret != -ENOTSUPP) - goto err; - } - return 0; -err: - return ret; -} - static const struct snd_soc_ops simple_ops = { .startup = asoc_simple_startup, .shutdown = asoc_simple_shutdown, - .hw_params = simple_hw_params, + .hw_params = asoc_simple_hw_params, }; static int simple_dai_init(struct snd_soc_pcm_runtime *rtd) -- cgit v1.2.3-59-g8ed1b From ad934ca8010843482d61fda46786449a9bc99e10 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:55:52 +0900 Subject: ASoC: simple-card-utils: share asoc_simple_dai_init() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph have almost same functions. This patch shares asoc_simple_dai_init() between in these 2 drivers. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 4 +--- sound/soc/generic/audio-graph-card.c | 23 ++--------------------- sound/soc/generic/simple-card-utils.c | 25 ++++++++++++++++++++++--- sound/soc/generic/simple-card.c | 25 +++---------------------- 4 files changed, 28 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 89ae0841b711..204343789ae7 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -88,6 +88,7 @@ int asoc_simple_startup(struct snd_pcm_substream *substream); void asoc_simple_shutdown(struct snd_pcm_substream *substream); int asoc_simple_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); +int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ @@ -130,9 +131,6 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, &(dai)->slots, \ &(dai)->slot_width); -int asoc_simple_card_init_dai(struct snd_soc_dai *dai, - struct asoc_simple_dai *simple_dai); - void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link); void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int is_single_links); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 51160adaf5bb..b1ceefc60879 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -62,25 +62,6 @@ static const struct snd_soc_ops graph_ops = { .hw_params = asoc_simple_hw_params, }; -static int graph_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); - if (ret < 0) - return ret; - - return 0; -} - static int graph_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -255,7 +236,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; dai_link->ops = &graph_ops; - dai_link->init = graph_dai_init; + dai_link->init = asoc_simple_dai_init; return 0; } @@ -327,7 +308,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, return ret; dai_link->ops = &graph_ops; - dai_link->init = graph_dai_init; + dai_link->init = asoc_simple_dai_init; asoc_simple_card_canonicalize_platform(dai_link); asoc_simple_card_canonicalize_cpu(dai_link, diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index de23cf8bffbd..c9b53b340028 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -436,8 +436,8 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_graph_dai); -int asoc_simple_card_init_dai(struct snd_soc_dai *dai, - struct asoc_simple_dai *simple_dai) +static int asoc_simple_card_init_dai(struct snd_soc_dai *dai, + struct asoc_simple_dai *simple_dai) { int ret; @@ -467,7 +467,26 @@ int asoc_simple_card_init_dai(struct snd_soc_dai *dai, return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); + +int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; + + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); + if (ret < 0) + return ret; + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_dai_init); void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link) { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7e836473eb11..bf4482ecfe3e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -32,25 +32,6 @@ static const struct snd_soc_ops simple_ops = { .hw_params = asoc_simple_hw_params, }; -static int simple_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - int ret; - - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); - if (ret < 0) - return ret; - - return 0; -} - static int simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -221,7 +202,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; dai_link->ops = &simple_ops; - dai_link->init = simple_dai_init; + dai_link->init = asoc_simple_dai_init; return 0; } @@ -316,7 +297,7 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, goto dai_link_of_err; dai_link->ops = &simple_ops; - dai_link->init = simple_dai_init; + dai_link->init = asoc_simple_dai_init; asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); asoc_simple_card_canonicalize_platform(dai_link); @@ -694,7 +675,7 @@ static int simple_probe(struct platform_device *pdev) dai_link->stream_name = cinfo->name; dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->dai_fmt = cinfo->daifmt; - dai_link->init = simple_dai_init; + dai_link->init = asoc_simple_dai_init; memcpy(priv->dai_props->cpu_dai, &cinfo->cpu_dai, sizeof(*priv->dai_props->cpu_dai)); memcpy(priv->dai_props->codec_dai, &cinfo->codec_dai, -- cgit v1.2.3-59-g8ed1b From 629f75440a68220a78aef9d8569831824890c47d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:56:06 +0900 Subject: ASoC: simple-card-utils: share asoc_simple_be_hw_params_fixup() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph have almost same functions. This patch shares asoc_simple_be_hw_params_fixup() between in these 2 drivers. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/audio-graph-card.c | 13 +------------ sound/soc/generic/simple-card-utils.c | 12 ++++++++++++ sound/soc/generic/simple-card.c | 13 +------------ 4 files changed, 16 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 204343789ae7..20d41bb14ea8 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -89,6 +89,8 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream); int asoc_simple_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd); +int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index b1ceefc60879..605126fb2810 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -62,17 +62,6 @@ static const struct snd_soc_ops graph_ops = { .hw_params = asoc_simple_hw_params, }; -static int graph_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - - asoc_simple_card_convert_fixup(&dai_props->adata, params); - - return 0; -} - static void graph_parse_convert(struct device *dev, struct device_node *ep, struct asoc_simple_card_data *adata) @@ -186,7 +175,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, /* BE settings */ dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = graph_be_hw_params_fixup; + dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; dai = dai_props->codec_dai = &priv->dais[li->dais++]; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index c9b53b340028..ec4a010400a0 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -293,6 +293,18 @@ err: } EXPORT_SYMBOL_GPL(asoc_simple_hw_params); +int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + + asoc_simple_card_convert_fixup(&dai_props->adata, params); + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_be_hw_params_fixup); + int asoc_simple_card_parse_dai(struct device_node *node, struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bf4482ecfe3e..60a88a55c071 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -32,17 +32,6 @@ static const struct snd_soc_ops simple_ops = { .hw_params = asoc_simple_hw_params, }; -static int simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - - asoc_simple_card_convert_fixup(&dai_props->adata, params); - - return 0; -} - static void simple_parse_convert(struct device *dev, struct device_node *np, struct asoc_simple_card_data *adata) @@ -154,7 +143,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, /* BE settings */ dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = simple_be_hw_params_fixup; + dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; dai = dai_props->codec_dai = &priv->dais[li->dais++]; -- cgit v1.2.3-59-g8ed1b From 65a5056b21202eff7f54243e587183f4bb6ed352 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:56:26 +0900 Subject: ASoC: simple-card-utils: share asoc_simple_card_init_priv() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph are initializing each priv, but it is same operation. This patch adds new asoc_simple_card_init_priv() and initialize priv by same operation. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 10 ++++++- sound/soc/generic/audio-graph-card.c | 45 +++------------------------- sound/soc/generic/simple-card-utils.c | 49 ++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 56 +++++++---------------------------- 4 files changed, 73 insertions(+), 87 deletions(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 20d41bb14ea8..c0be02de6993 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -60,6 +60,13 @@ struct asoc_simple_priv { #define simple_priv_to_dev(priv) (simple_priv_to_card(priv)->dev) #define simple_priv_to_link(priv, i) (simple_priv_to_card(priv)->dai_link + (i)) +struct link_info { + int dais; /* number of dai */ + int link; /* number of link */ + int conf; /* number of codec_conf */ + int cpu; /* turn for CPU / Codec */ +}; + int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, @@ -153,7 +160,8 @@ int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, int asoc_simple_card_init_jack(struct snd_soc_card *card, struct asoc_simple_jack *sjack, int is_hp, char *prefix); - +int asoc_simple_card_init_priv(struct asoc_simple_priv *priv, + struct link_info *li); #ifdef DEBUG inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv, diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 605126fb2810..b9a93379098d 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -20,13 +20,6 @@ #include #include -struct link_info { - int dais; /* number of dai */ - int link; /* number of link */ - int conf; /* number of codec_conf */ - int cpu; /* turn for CPU / Codec */ -}; - #define PREFIX "audio-graph-card," static int graph_outdrv_event(struct snd_soc_dapm_widget *w, @@ -526,14 +519,10 @@ static int graph_card_probe(struct snd_soc_card *card) static int graph_probe(struct platform_device *pdev) { struct asoc_simple_priv *priv; - struct snd_soc_dai_link *dai_link; - struct simple_dai_props *dai_props; - struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; - struct snd_soc_codec_conf *cconf; struct link_info li; - int ret, i; + int ret; /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -552,25 +541,9 @@ static int graph_probe(struct platform_device *pdev) if (!li.link || !li.dais) return -EINVAL; - dai_props = devm_kcalloc(dev, li.link, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, li.link, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, li.dais, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, li.conf, sizeof(*cconf), GFP_KERNEL); - if (!dai_props || !dai_link || !dais) - return -ENOMEM; - - /* - * Use snd_soc_dai_link_component instead of legacy style - * It is codec only. but cpu/platform will be supported in the future. - * see - * soc-core.c :: snd_soc_init_multicodec() - */ - for (i = 0; i < li.link; i++) { - dai_link[i].codecs = &dai_props[i].codecs; - dai_link[i].num_codecs = 1; - dai_link[i].platforms = &dai_props[i].platforms; - dai_link[i].num_platforms = 1; - } + ret = asoc_simple_card_init_priv(priv, &li); + if (ret < 0) + return ret; priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); if (IS_ERR(priv->pa_gpio)) { @@ -579,16 +552,6 @@ static int graph_probe(struct platform_device *pdev) return ret; } - priv->dai_props = dai_props; - priv->dai_link = dai_link; - priv->dais = dais; - priv->codec_conf = cconf; - - card->dai_link = dai_link; - card->num_links = li.link; - card->codec_conf = cconf; - card->num_configs = li.conf; - ret = graph_parse_of(priv); if (ret < 0) { if (ret != -EPROBE_DEFER) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index ec4a010400a0..697e820bee18 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -630,6 +630,55 @@ int asoc_simple_card_init_jack(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_init_jack); +int asoc_simple_card_init_priv(struct asoc_simple_priv *priv, + struct link_info *li) +{ + struct snd_soc_card *card = simple_priv_to_card(priv); + struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link; + struct simple_dai_props *dai_props; + struct asoc_simple_dai *dais; + struct snd_soc_codec_conf *cconf; + int i; + + dai_props = devm_kcalloc(dev, li->link, sizeof(*dai_props), GFP_KERNEL); + dai_link = devm_kcalloc(dev, li->link, sizeof(*dai_link), GFP_KERNEL); + dais = devm_kcalloc(dev, li->dais, sizeof(*dais), GFP_KERNEL); + cconf = devm_kcalloc(dev, li->conf, sizeof(*cconf), GFP_KERNEL); + if (!dai_props || !dai_link || !dais) + return -ENOMEM; + + /* + * Use snd_soc_dai_link_component instead of legacy style + * It is codec only. but cpu/platform will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + * + * "platform" might be removed + * see + * simple-card-utils.c :: asoc_simple_card_canonicalize_platform() + */ + for (i = 0; i < li->link; i++) { + dai_link[i].codecs = &dai_props[i].codecs; + dai_link[i].num_codecs = 1; + dai_link[i].platforms = &dai_props[i].platforms; + dai_link[i].num_platforms = 1; + } + + priv->dai_props = dai_props; + priv->dai_link = dai_link; + priv->dais = dais; + priv->codec_conf = cconf; + + card->dai_link = priv->dai_link; + card->num_links = li->link; + card->codec_conf = cconf; + card->num_configs = li->conf; + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_init_priv); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 60a88a55c071..4e3e6b34593c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -15,13 +15,6 @@ #include #include -struct link_info { - int dais; /* number of dai */ - int link; /* number of link */ - int conf; /* number of codec_conf */ - int cpu; /* turn for CPU / Codec */ -}; - #define DAI "sound-dai" #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," @@ -564,15 +557,11 @@ static int simple_soc_probe(struct snd_soc_card *card) static int simple_probe(struct platform_device *pdev) { struct asoc_simple_priv *priv; - struct snd_soc_dai_link *dai_link; - struct simple_dai_props *dai_props; - struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; struct snd_soc_card *card; - struct snd_soc_codec_conf *cconf; struct link_info li; - int ret, i; + int ret; /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -589,35 +578,9 @@ static int simple_probe(struct platform_device *pdev) if (!li.link || !li.dais) return -EINVAL; - dai_props = devm_kcalloc(dev, li.link, sizeof(*dai_props), GFP_KERNEL); - dai_link = devm_kcalloc(dev, li.link, sizeof(*dai_link), GFP_KERNEL); - dais = devm_kcalloc(dev, li.dais, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, li.conf, sizeof(*cconf), GFP_KERNEL); - if (!dai_props || !dai_link || !dais) - return -ENOMEM; - - /* - * Use snd_soc_dai_link_component instead of legacy style - * It is codec only. but cpu/platform will be supported in the future. - * see - * soc-core.c :: snd_soc_init_multicodec() - */ - for (i = 0; i < li.link; i++) { - dai_link[i].codecs = &dai_props[i].codecs; - dai_link[i].num_codecs = 1; - dai_link[i].platforms = &dai_props[i].platforms; - dai_link[i].num_platforms = 1; - } - - priv->dai_props = dai_props; - priv->dai_link = dai_link; - priv->dais = dais; - priv->codec_conf = cconf; - - card->dai_link = priv->dai_link; - card->num_links = li.link; - card->codec_conf = cconf; - card->num_configs = li.conf; + ret = asoc_simple_card_init_priv(priv, &li); + if (ret < 0) + return ret; if (np && of_device_is_available(np)) { @@ -632,6 +595,9 @@ static int simple_probe(struct platform_device *pdev) struct asoc_simple_card_info *cinfo; struct snd_soc_dai_link_component *codecs; struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dai_link = priv->dai_link; + struct simple_dai_props *dai_props = priv->dai_props; + int dai_idx = 0; cinfo = dev->platform_data; @@ -665,10 +631,10 @@ static int simple_probe(struct platform_device *pdev) dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->dai_fmt = cinfo->daifmt; dai_link->init = asoc_simple_dai_init; - memcpy(priv->dai_props->cpu_dai, &cinfo->cpu_dai, - sizeof(*priv->dai_props->cpu_dai)); - memcpy(priv->dai_props->codec_dai, &cinfo->codec_dai, - sizeof(*priv->dai_props->codec_dai)); + memcpy(dai_props->cpu_dai, &cinfo->cpu_dai, + sizeof(*dai_props->cpu_dai)); + memcpy(dai_props->codec_dai, &cinfo->codec_dai, + sizeof(*dai_props->codec_dai)); } snd_soc_card_set_drvdata(card, priv); -- cgit v1.2.3-59-g8ed1b From 8f7f298a333761a528e103cda3b42f3a416ad1ee Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:56:36 +0900 Subject: ASoC: simple-card-utils: separate asoc_simple_card_parse_dai() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph are using asoc_simple_card_parse_dai() which is different implementation. But, these are implemanted at simple-card-utils. It should be implemanted at each files. This patch separate these into each files. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 39 ++-------- sound/soc/generic/audio-graph-card.c | 114 ++++++++++++++++++++++++--- sound/soc/generic/simple-card-utils.c | 143 ---------------------------------- sound/soc/generic/simple-card.c | 58 ++++++++++++-- 4 files changed, 164 insertions(+), 190 deletions(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index c0be02de6993..0c070c2bc119 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -99,40 +99,17 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd); int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); -#define asoc_simple_card_parse_cpu(node, dai_link, \ - list_name, cells_name, is_single_link) \ - asoc_simple_card_parse_dai(node, NULL, \ - &dai_link->cpu_of_node, \ - &dai_link->cpu_dai_name, list_name, cells_name, is_single_link) -#define asoc_simple_card_parse_codec(node, dai_link, list_name, cells_name) \ +#define asoc_simple_card_parse_cpu(node, dai_link, is_single_link) \ + asoc_simple_card_parse_dai(node, NULL, \ + &dai_link->cpu_of_node, \ + &dai_link->cpu_dai_name, is_single_link) +#define asoc_simple_card_parse_codec(node, dai_link) \ asoc_simple_card_parse_dai(node, dai_link->codecs, \ &dai_link->codec_of_node, \ - &dai_link->codec_dai_name, \ - list_name, cells_name, NULL) -#define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \ + &dai_link->codec_dai_name, NULL) +#define asoc_simple_card_parse_platform(node, dai_link) \ asoc_simple_card_parse_dai(node, dai_link->platforms, \ - &dai_link->platform_of_node, \ - NULL, list_name, cells_name, NULL) -int asoc_simple_card_parse_dai(struct device_node *node, - struct snd_soc_dai_link_component *dlc, - struct device_node **endpoint_np, - const char **dai_name, - const char *list_name, - const char *cells_name, - int *is_single_links); - -#define asoc_simple_card_parse_graph_cpu(ep, dai_link) \ - asoc_simple_card_parse_graph_dai(ep, NULL, \ - &dai_link->cpu_of_node, \ - &dai_link->cpu_dai_name) -#define asoc_simple_card_parse_graph_codec(ep, dai_link) \ - asoc_simple_card_parse_graph_dai(ep, dai_link->codecs, \ - &dai_link->codec_of_node, \ - &dai_link->codec_dai_name) -int asoc_simple_card_parse_graph_dai(struct device_node *ep, - struct snd_soc_dai_link_component *dlc, - struct device_node **endpoint_np, - const char **dai_name); + &dai_link->platform_of_node, NULL, NULL) #define asoc_simple_card_of_parse_tdm(np, dai) \ snd_soc_of_parse_tdm_slot(np, &(dai)->tx_slot_mask, \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index b9a93379098d..3370b6fc6da2 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -55,6 +55,103 @@ static const struct snd_soc_ops graph_ops = { .hw_params = asoc_simple_hw_params, }; +static int graph_get_dai_id(struct device_node *ep) +{ + struct device_node *node; + struct device_node *endpoint; + struct of_endpoint info; + int i, id; + int ret; + + /* use driver specified DAI ID if exist */ + ret = snd_soc_get_dai_id(ep); + if (ret != -ENOTSUPP) + return ret; + + /* use endpoint/port reg if exist */ + ret = of_graph_parse_endpoint(ep, &info); + if (ret == 0) { + /* + * Because it will count port/endpoint if it doesn't have "reg". + * But, we can't judge whether it has "no reg", or "reg = <0>" + * only of_graph_parse_endpoint(). + * We need to check "reg" property + */ + if (of_get_property(ep, "reg", NULL)) + return info.id; + + node = of_get_parent(ep); + of_node_put(node); + if (of_get_property(node, "reg", NULL)) + return info.port; + } + node = of_graph_get_port_parent(ep); + + /* + * Non HDMI sound case, counting port/endpoint on its DT + * is enough. Let's count it. + */ + i = 0; + id = -1; + for_each_endpoint_of_node(node, endpoint) { + if (endpoint == ep) + id = i; + i++; + } + + of_node_put(node); + + if (id < 0) + return -ENODEV; + + return id; +} + +static int asoc_simple_card_parse_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, + struct device_node **dai_of_node, + const char **dai_name, + int *is_single_link) +{ + struct device_node *node; + struct of_phandle_args args; + int ret; + + /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_name = &dlc->dai_name; + dai_of_node = &dlc->of_node; + } + + if (!ep) + return 0; + if (!dai_name) + return 0; + + node = of_graph_get_port_parent(ep); + + /* Get dai->name */ + args.np = node; + args.args[0] = graph_get_dai_id(ep); + args.args_count = (of_graph_get_endpoint_count(node) > 1); + + ret = snd_soc_get_dai_name(&args, dai_name); + if (ret < 0) + return ret; + + *dai_of_node = node; + + if (is_single_link) + *is_single_link = of_graph_get_endpoint_count(node) == 1; + + return 0; +} + static void graph_parse_convert(struct device *dev, struct device_node *ep, struct asoc_simple_card_data *adata) @@ -128,6 +225,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, of_node_put(node); if (li->cpu) { + int is_single_links = 0; /* BE is dummy */ codecs->of_node = NULL; @@ -141,7 +239,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai = dai_props->cpu_dai = &priv->dais[li->dais++]; - ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); + ret = asoc_simple_card_parse_cpu(ep, dai_link, &is_single_links); if (ret) return ret; @@ -156,8 +254,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, return ret; /* card->num_links includes Codec */ - asoc_simple_card_canonicalize_cpu(dai_link, - of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); + asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); } else { struct snd_soc_codec_conf *cconf; @@ -176,7 +273,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, cconf = dai_props->codec_conf = &priv->codec_conf[li->conf++]; - ret = asoc_simple_card_parse_graph_codec(ep, dai_link); + ret = asoc_simple_card_parse_codec(ep, dai_link); if (ret < 0) return ret; @@ -234,7 +331,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct device_node *top = dev->of_node; struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; - int ret; + int ret, single_cpu; /* Do it only CPU turn */ if (!li->cpu) @@ -258,11 +355,11 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, if (ret < 0) return ret; - ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link); + ret = asoc_simple_card_parse_cpu(cpu_ep, dai_link, &single_cpu); if (ret < 0) return ret; - ret = asoc_simple_card_parse_graph_codec(codec_ep, dai_link); + ret = asoc_simple_card_parse_codec(codec_ep, dai_link); if (ret < 0) return ret; @@ -293,8 +390,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, dai_link->init = asoc_simple_dai_init; asoc_simple_card_canonicalize_platform(dai_link); - asoc_simple_card_canonicalize_cpu(dai_link, - of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); + asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); return 0; } diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 697e820bee18..40cefbde6d9b 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -305,149 +305,6 @@ int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, } EXPORT_SYMBOL_GPL(asoc_simple_be_hw_params_fixup); -int asoc_simple_card_parse_dai(struct device_node *node, - struct snd_soc_dai_link_component *dlc, - struct device_node **dai_of_node, - const char **dai_name, - const char *list_name, - const char *cells_name, - int *is_single_link) -{ - struct of_phandle_args args; - int ret; - - if (!node) - return 0; - - /* - * Use snd_soc_dai_link_component instead of legacy style. - * It is only for codec, but cpu will be supported in the future. - * see - * soc-core.c :: snd_soc_init_multicodec() - */ - if (dlc) { - dai_name = &dlc->dai_name; - dai_of_node = &dlc->of_node; - } - - /* - * Get node via "sound-dai = <&phandle port>" - * it will be used as xxx_of_node on soc_bind_dai_link() - */ - ret = of_parse_phandle_with_args(node, list_name, cells_name, 0, &args); - if (ret) - return ret; - - /* Get dai->name */ - if (dai_name) { - ret = snd_soc_of_get_dai_name(node, dai_name); - if (ret < 0) - return ret; - } - - *dai_of_node = args.np; - - if (is_single_link) - *is_single_link = !args.args_count; - - return 0; -} -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_dai); - -static int asoc_simple_card_get_dai_id(struct device_node *ep) -{ - struct device_node *node; - struct device_node *endpoint; - struct of_endpoint info; - int i, id; - int ret; - - /* use driver specified DAI ID if exist */ - ret = snd_soc_get_dai_id(ep); - if (ret != -ENOTSUPP) - return ret; - - /* use endpoint/port reg if exist */ - ret = of_graph_parse_endpoint(ep, &info); - if (ret == 0) { - /* - * Because it will count port/endpoint if it doesn't have "reg". - * But, we can't judge whether it has "no reg", or "reg = <0>" - * only of_graph_parse_endpoint(). - * We need to check "reg" property - */ - if (of_get_property(ep, "reg", NULL)) - return info.id; - - node = of_get_parent(ep); - of_node_put(node); - if (of_get_property(node, "reg", NULL)) - return info.port; - } - node = of_graph_get_port_parent(ep); - - /* - * Non HDMI sound case, counting port/endpoint on its DT - * is enough. Let's count it. - */ - i = 0; - id = -1; - for_each_endpoint_of_node(node, endpoint) { - if (endpoint == ep) - id = i; - i++; - } - - of_node_put(node); - - if (id < 0) - return -ENODEV; - - return id; -} - -int asoc_simple_card_parse_graph_dai(struct device_node *ep, - struct snd_soc_dai_link_component *dlc, - struct device_node **dai_of_node, - const char **dai_name) -{ - struct device_node *node; - struct of_phandle_args args; - int ret; - - /* - * Use snd_soc_dai_link_component instead of legacy style. - * It is only for codec, but cpu will be supported in the future. - * see - * soc-core.c :: snd_soc_init_multicodec() - */ - if (dlc) { - dai_name = &dlc->dai_name; - dai_of_node = &dlc->of_node; - } - - if (!ep) - return 0; - if (!dai_name) - return 0; - - node = of_graph_get_port_parent(ep); - - /* Get dai->name */ - args.np = node; - args.args[0] = asoc_simple_card_get_dai_id(ep); - args.args_count = (of_graph_get_endpoint_count(node) > 1); - - ret = snd_soc_get_dai_name(&args, dai_name); - if (ret < 0) - return ret; - - *dai_of_node = node; - - return 0; -} -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_graph_dai); - static int asoc_simple_card_init_dai(struct snd_soc_dai *dai, struct asoc_simple_dai *simple_dai) { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 4e3e6b34593c..d8560fb1f5de 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -25,6 +25,52 @@ static const struct snd_soc_ops simple_ops = { .hw_params = asoc_simple_hw_params, }; +static int asoc_simple_card_parse_dai(struct device_node *node, + struct snd_soc_dai_link_component *dlc, + struct device_node **dai_of_node, + const char **dai_name, + int *is_single_link) +{ + struct of_phandle_args args; + int ret; + + if (!node) + return 0; + + /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_name = &dlc->dai_name; + dai_of_node = &dlc->of_node; + } + + /* + * Get node via "sound-dai = <&phandle port>" + * it will be used as xxx_of_node on soc_bind_dai_link() + */ + ret = of_parse_phandle_with_args(node, DAI, CELL, 0, &args); + if (ret) + return ret; + + /* Get dai->name */ + if (dai_name) { + ret = snd_soc_of_get_dai_name(node, dai_name); + if (ret < 0) + return ret; + } + + *dai_of_node = args.np; + + if (is_single_link) + *is_single_link = !args.args_count; + + return 0; +} + static void simple_parse_convert(struct device *dev, struct device_node *np, struct asoc_simple_card_data *adata) @@ -110,8 +156,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai = dai_props->cpu_dai = &priv->dais[li->dais++]; - ret = asoc_simple_card_parse_cpu(np, dai_link, DAI, CELL, - &is_single_links); + ret = asoc_simple_card_parse_cpu(np, dai_link, &is_single_links); if (ret) return ret; @@ -144,7 +189,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, cconf = dai_props->codec_conf = &priv->codec_conf[li->conf++]; - ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); + ret = asoc_simple_card_parse_codec(np, dai_link); if (ret < 0) return ret; @@ -242,16 +287,15 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); - ret = asoc_simple_card_parse_cpu(cpu, dai_link, - DAI, CELL, &single_cpu); + ret = asoc_simple_card_parse_cpu(cpu, dai_link, &single_cpu); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_codec(codec, dai_link, DAI, CELL); + ret = asoc_simple_card_parse_codec(codec, dai_link); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_platform(plat, dai_link, DAI, CELL); + ret = asoc_simple_card_parse_platform(plat, dai_link); if (ret < 0) goto dai_link_of_err; -- cgit v1.2.3-59-g8ed1b From ad11e59f52d6fc75037ac3cb66dc711b83c1bbf8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:56:50 +0900 Subject: ASoC: simple-card-utils: rename asoc_simple_card_xxx() to asoc_simple_() Current simple-card-utils is using asoc_simple_card_xxx() for each function naming, but it is very verbose. Thus it is easy to be over 80 char. This patch renames it to asoc_simple_xxx(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 90 ++++++++++++------------- sound/soc/generic/audio-graph-card.c | 98 +++++++++++++-------------- sound/soc/generic/simple-card-utils.c | 122 +++++++++++++++++----------------- sound/soc/generic/simple-card.c | 98 +++++++++++++-------------- 4 files changed, 204 insertions(+), 204 deletions(-) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 0c070c2bc119..67dc3ee6ed01 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -10,10 +10,10 @@ #include -#define asoc_simple_card_init_hp(card, sjack, prefix) \ - asoc_simple_card_init_jack(card, sjack, 1, prefix) -#define asoc_simple_card_init_mic(card, sjack, prefix) \ - asoc_simple_card_init_jack(card, sjack, 0, prefix) +#define asoc_simple_init_hp(card, sjack, prefix) \ + asoc_simple_init_jack(card, sjack, 1, prefix) +#define asoc_simple_init_mic(card, sjack, prefix) \ + asoc_simple_init_jack(card, sjack, 0, prefix) struct asoc_simple_dai { const char *name; @@ -26,7 +26,7 @@ struct asoc_simple_dai { struct clk *clk; }; -struct asoc_simple_card_data { +struct asoc_simple_data { u32 convert_rate; u32 convert_channels; }; @@ -44,7 +44,7 @@ struct asoc_simple_priv { struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ struct snd_soc_dai_link_component platforms; - struct asoc_simple_card_data adata; + struct asoc_simple_data adata; struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; } *dai_props; @@ -67,30 +67,30 @@ struct link_info { int cpu; /* turn for CPU / Codec */ }; -int asoc_simple_card_parse_daifmt(struct device *dev, - struct device_node *node, - struct device_node *codec, - char *prefix, - unsigned int *retfmt); +int asoc_simple_parse_daifmt(struct device *dev, + struct device_node *node, + struct device_node *codec, + char *prefix, + unsigned int *retfmt); __printf(3, 4) -int asoc_simple_card_set_dailink_name(struct device *dev, - struct snd_soc_dai_link *dai_link, - const char *fmt, ...); -int asoc_simple_card_parse_card_name(struct snd_soc_card *card, - char *prefix); - -#define asoc_simple_card_parse_clk_cpu(dev, node, dai_link, simple_dai) \ - asoc_simple_card_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai, \ +int asoc_simple_set_dailink_name(struct device *dev, + struct snd_soc_dai_link *dai_link, + const char *fmt, ...); +int asoc_simple_parse_card_name(struct snd_soc_card *card, + char *prefix); + +#define asoc_simple_parse_clk_cpu(dev, node, dai_link, simple_dai) \ + asoc_simple_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai, \ dai_link->cpu_dai_name, NULL) -#define asoc_simple_card_parse_clk_codec(dev, node, dai_link, simple_dai) \ - asoc_simple_card_parse_clk(dev, node, dai_link->codec_of_node, simple_dai,\ +#define asoc_simple_parse_clk_codec(dev, node, dai_link, simple_dai) \ + asoc_simple_parse_clk(dev, node, dai_link->codec_of_node, simple_dai,\ dai_link->codec_dai_name, dai_link->codecs) -int asoc_simple_card_parse_clk(struct device *dev, - struct device_node *node, - struct device_node *dai_of_node, - struct asoc_simple_dai *simple_dai, - const char *dai_name, - struct snd_soc_dai_link_component *dlc); +int asoc_simple_parse_clk(struct device *dev, + struct device_node *node, + struct device_node *dai_of_node, + struct asoc_simple_dai *simple_dai, + const char *dai_name, + struct snd_soc_dai_link_component *dlc); int asoc_simple_startup(struct snd_pcm_substream *substream); void asoc_simple_shutdown(struct snd_pcm_substream *substream); int asoc_simple_hw_params(struct snd_pcm_substream *substream, @@ -99,45 +99,45 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd); int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); -#define asoc_simple_card_parse_cpu(node, dai_link, is_single_link) \ - asoc_simple_card_parse_dai(node, NULL, \ +#define asoc_simple_parse_cpu(node, dai_link, is_single_link) \ + asoc_simple_parse_dai(node, NULL, \ &dai_link->cpu_of_node, \ &dai_link->cpu_dai_name, is_single_link) -#define asoc_simple_card_parse_codec(node, dai_link) \ - asoc_simple_card_parse_dai(node, dai_link->codecs, \ +#define asoc_simple_parse_codec(node, dai_link) \ + asoc_simple_parse_dai(node, dai_link->codecs, \ &dai_link->codec_of_node, \ &dai_link->codec_dai_name, NULL) -#define asoc_simple_card_parse_platform(node, dai_link) \ - asoc_simple_card_parse_dai(node, dai_link->platforms, \ +#define asoc_simple_parse_platform(node, dai_link) \ + asoc_simple_parse_dai(node, dai_link->platforms, \ &dai_link->platform_of_node, NULL, NULL) -#define asoc_simple_card_of_parse_tdm(np, dai) \ +#define asoc_simple_parse_tdm(np, dai) \ snd_soc_of_parse_tdm_slot(np, &(dai)->tx_slot_mask, \ &(dai)->rx_slot_mask, \ &(dai)->slots, \ &(dai)->slot_width); -void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link); -void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, +void asoc_simple_canonicalize_platform(struct snd_soc_dai_link *dai_link); +void asoc_simple_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int is_single_links); -int asoc_simple_card_clean_reference(struct snd_soc_card *card); +int asoc_simple_clean_reference(struct snd_soc_card *card); -void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, +void asoc_simple_convert_fixup(struct asoc_simple_data *data, struct snd_pcm_hw_params *params); -void asoc_simple_card_parse_convert(struct device *dev, - struct device_node *np, char *prefix, - struct asoc_simple_card_data *data); +void asoc_simple_parse_convert(struct device *dev, + struct device_node *np, char *prefix, + struct asoc_simple_data *data); -int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, +int asoc_simple_parse_routing(struct snd_soc_card *card, char *prefix); -int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, +int asoc_simple_parse_widgets(struct snd_soc_card *card, char *prefix); -int asoc_simple_card_init_jack(struct snd_soc_card *card, +int asoc_simple_init_jack(struct snd_soc_card *card, struct asoc_simple_jack *sjack, int is_hp, char *prefix); -int asoc_simple_card_init_priv(struct asoc_simple_priv *priv, +int asoc_simple_init_priv(struct asoc_simple_priv *priv, struct link_info *li); #ifdef DEBUG diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 3370b6fc6da2..8114b5ea9319 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -107,11 +107,11 @@ static int graph_get_dai_id(struct device_node *ep) return id; } -static int asoc_simple_card_parse_dai(struct device_node *ep, - struct snd_soc_dai_link_component *dlc, - struct device_node **dai_of_node, - const char **dai_name, - int *is_single_link) +static int asoc_simple_parse_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, + struct device_node **dai_of_node, + const char **dai_name, + int *is_single_link) { struct device_node *node; struct of_phandle_args args; @@ -154,18 +154,18 @@ static int asoc_simple_card_parse_dai(struct device_node *ep, static void graph_parse_convert(struct device *dev, struct device_node *ep, - struct asoc_simple_card_data *adata) + struct asoc_simple_data *adata) { struct device_node *top = dev->of_node; struct device_node *port = of_get_parent(ep); struct device_node *ports = of_get_parent(port); struct device_node *node = of_graph_get_port_parent(ep); - asoc_simple_card_parse_convert(dev, top, NULL, adata); - asoc_simple_card_parse_convert(dev, node, PREFIX, adata); - asoc_simple_card_parse_convert(dev, ports, NULL, adata); - asoc_simple_card_parse_convert(dev, port, NULL, adata); - asoc_simple_card_parse_convert(dev, ep, NULL, adata); + asoc_simple_parse_convert(dev, top, NULL, adata); + asoc_simple_parse_convert(dev, node, PREFIX, adata); + asoc_simple_parse_convert(dev, ports, NULL, adata); + asoc_simple_parse_convert(dev, port, NULL, adata); + asoc_simple_parse_convert(dev, ep, NULL, adata); of_node_put(port); of_node_put(ports); @@ -239,22 +239,22 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai = dai_props->cpu_dai = &priv->dais[li->dais++]; - ret = asoc_simple_card_parse_cpu(ep, dai_link, &is_single_links); + ret = asoc_simple_parse_cpu(ep, dai_link, &is_single_links); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); + ret = asoc_simple_parse_clk_cpu(dev, ep, dai_link, dai); if (ret < 0) return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "fe.%s", - dai_link->cpu_dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "fe.%s", + dai_link->cpu_dai_name); if (ret < 0) return ret; /* card->num_links includes Codec */ - asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); + asoc_simple_canonicalize_cpu(dai_link, is_single_links); } else { struct snd_soc_codec_conf *cconf; @@ -273,17 +273,17 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, cconf = dai_props->codec_conf = &priv->codec_conf[li->conf++]; - ret = asoc_simple_card_parse_codec(ep, dai_link); + ret = asoc_simple_parse_codec(ep, dai_link); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); + ret = asoc_simple_parse_clk_codec(dev, ep, dai_link, dai); if (ret < 0) return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "be.%s", - codecs->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "be.%s", + codecs->dai_name); if (ret < 0) return ret; @@ -301,14 +301,14 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, graph_parse_convert(dev, ep, &dai_props->adata); graph_parse_mclk_fs(top, ep, dai_props); - asoc_simple_card_canonicalize_platform(dai_link); + asoc_simple_canonicalize_platform(dai_link); - ret = asoc_simple_card_of_parse_tdm(ep, dai); + ret = asoc_simple_parse_tdm(ep, dai); if (ret) return ret; - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, - NULL, &dai_link->dai_fmt); + ret = asoc_simple_parse_daifmt(dev, cpu_ep, codec_ep, + NULL, &dai_link->dai_fmt); if (ret < 0) return ret; @@ -350,47 +350,47 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, graph_parse_mclk_fs(top, cpu_ep, dai_props); graph_parse_mclk_fs(top, codec_ep, dai_props); - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, - NULL, &dai_link->dai_fmt); + ret = asoc_simple_parse_daifmt(dev, cpu_ep, codec_ep, + NULL, &dai_link->dai_fmt); if (ret < 0) return ret; - ret = asoc_simple_card_parse_cpu(cpu_ep, dai_link, &single_cpu); + ret = asoc_simple_parse_cpu(cpu_ep, dai_link, &single_cpu); if (ret < 0) return ret; - ret = asoc_simple_card_parse_codec(codec_ep, dai_link); + ret = asoc_simple_parse_codec(codec_ep, dai_link); if (ret < 0) return ret; - ret = asoc_simple_card_of_parse_tdm(cpu_ep, cpu_dai); + ret = asoc_simple_parse_tdm(cpu_ep, cpu_dai); if (ret < 0) return ret; - ret = asoc_simple_card_of_parse_tdm(codec_ep, codec_dai); + ret = asoc_simple_parse_tdm(codec_ep, codec_dai); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, cpu_ep, dai_link, cpu_dai); + ret = asoc_simple_parse_clk_cpu(dev, cpu_ep, dai_link, cpu_dai); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, codec_ep, dai_link, codec_dai); + ret = asoc_simple_parse_clk_codec(dev, codec_ep, dai_link, codec_dai); if (ret < 0) return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "%s-%s", - dai_link->cpu_dai_name, - dai_link->codecs->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "%s-%s", + dai_link->cpu_dai_name, + dai_link->codecs->dai_name); if (ret < 0) return ret; dai_link->ops = &graph_ops; dai_link->init = asoc_simple_dai_init; - asoc_simple_card_canonicalize_platform(dai_link); - asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); + asoc_simple_canonicalize_cpu(dai_link, single_cpu); + asoc_simple_canonicalize_platform(dai_link); return 0; } @@ -414,7 +414,7 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old = NULL; - struct asoc_simple_card_data adata; + struct asoc_simple_data adata; int rc, ret; /* loop for all listed CPU port */ @@ -469,11 +469,11 @@ static int graph_parse_of(struct asoc_simple_priv *priv) struct link_info li; int ret; - ret = asoc_simple_card_of_parse_widgets(card, NULL); + ret = asoc_simple_parse_widgets(card, NULL); if (ret < 0) return ret; - ret = asoc_simple_card_of_parse_routing(card, NULL); + ret = asoc_simple_parse_routing(card, NULL); if (ret < 0) return ret; @@ -498,7 +498,7 @@ static int graph_parse_of(struct asoc_simple_priv *priv) return ret; } - return asoc_simple_card_parse_card_name(card, NULL); + return asoc_simple_parse_card_name(card, NULL); } static int graph_count_noml(struct asoc_simple_priv *priv, @@ -601,11 +601,11 @@ static int graph_card_probe(struct snd_soc_card *card) struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(card); int ret; - ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL); + ret = asoc_simple_init_hp(card, &priv->hp_jack, NULL); if (ret < 0) return ret; - ret = asoc_simple_card_init_mic(card, &priv->mic_jack, NULL); + ret = asoc_simple_init_mic(card, &priv->mic_jack, NULL); if (ret < 0) return ret; @@ -637,7 +637,7 @@ static int graph_probe(struct platform_device *pdev) if (!li.link || !li.dais) return -EINVAL; - ret = asoc_simple_card_init_priv(priv, &li); + ret = asoc_simple_init_priv(priv, &li); if (ret < 0) return ret; @@ -665,7 +665,7 @@ static int graph_probe(struct platform_device *pdev) return 0; err: - asoc_simple_card_clean_reference(card); + asoc_simple_clean_reference(card); return ret; } @@ -674,7 +674,7 @@ static int graph_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - return asoc_simple_card_clean_reference(card); + return asoc_simple_clean_reference(card); } static const struct of_device_id graph_of_match[] = { diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 40cefbde6d9b..a794f767ff7f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -14,8 +14,8 @@ #include #include -void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, - struct snd_pcm_hw_params *params) +void asoc_simple_convert_fixup(struct asoc_simple_data *data, + struct snd_pcm_hw_params *params) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); @@ -30,12 +30,12 @@ void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, channels->min = channels->max = data->convert_channels; } -EXPORT_SYMBOL_GPL(asoc_simple_card_convert_fixup); +EXPORT_SYMBOL_GPL(asoc_simple_convert_fixup); -void asoc_simple_card_parse_convert(struct device *dev, - struct device_node *np, - char *prefix, - struct asoc_simple_card_data *data) +void asoc_simple_parse_convert(struct device *dev, + struct device_node *np, + char *prefix, + struct asoc_simple_data *data) { char prop[128]; @@ -50,13 +50,13 @@ void asoc_simple_card_parse_convert(struct device *dev, snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-channels"); of_property_read_u32(np, prop, &data->convert_channels); } -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_convert); +EXPORT_SYMBOL_GPL(asoc_simple_parse_convert); -int asoc_simple_card_parse_daifmt(struct device *dev, - struct device_node *node, - struct device_node *codec, - char *prefix, - unsigned int *retfmt) +int asoc_simple_parse_daifmt(struct device *dev, + struct device_node *node, + struct device_node *codec, + char *prefix, + unsigned int *retfmt) { struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; @@ -92,11 +92,11 @@ int asoc_simple_card_parse_daifmt(struct device *dev, return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_daifmt); +EXPORT_SYMBOL_GPL(asoc_simple_parse_daifmt); -int asoc_simple_card_set_dailink_name(struct device *dev, - struct snd_soc_dai_link *dai_link, - const char *fmt, ...) +int asoc_simple_set_dailink_name(struct device *dev, + struct snd_soc_dai_link *dai_link, + const char *fmt, ...) { va_list ap; char *name = NULL; @@ -115,10 +115,10 @@ int asoc_simple_card_set_dailink_name(struct device *dev, return ret; } -EXPORT_SYMBOL_GPL(asoc_simple_card_set_dailink_name); +EXPORT_SYMBOL_GPL(asoc_simple_set_dailink_name); -int asoc_simple_card_parse_card_name(struct snd_soc_card *card, - char *prefix) +int asoc_simple_parse_card_name(struct snd_soc_card *card, + char *prefix) { int ret; @@ -141,9 +141,9 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); +EXPORT_SYMBOL_GPL(asoc_simple_parse_card_name); -static int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) +static int asoc_simple_clk_enable(struct asoc_simple_dai *dai) { if (dai) return clk_prepare_enable(dai->clk); @@ -151,18 +151,18 @@ static int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) return 0; } -static void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) +static void asoc_simple_clk_disable(struct asoc_simple_dai *dai) { if (dai) clk_disable_unprepare(dai->clk); } -int asoc_simple_card_parse_clk(struct device *dev, - struct device_node *node, - struct device_node *dai_of_node, - struct asoc_simple_dai *simple_dai, - const char *dai_name, - struct snd_soc_dai_link_component *dlc) +int asoc_simple_parse_clk(struct device *dev, + struct device_node *node, + struct device_node *dai_of_node, + struct asoc_simple_dai *simple_dai, + const char *dai_name, + struct snd_soc_dai_link_component *dlc) { struct clk *clk; u32 val; @@ -202,7 +202,7 @@ int asoc_simple_card_parse_clk(struct device *dev, return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); +EXPORT_SYMBOL_GPL(asoc_simple_parse_clk); int asoc_simple_startup(struct snd_pcm_substream *substream) { @@ -211,13 +211,13 @@ int asoc_simple_startup(struct snd_pcm_substream *substream) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + ret = asoc_simple_clk_enable(dai_props->cpu_dai); if (ret) return ret; - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + ret = asoc_simple_clk_enable(dai_props->codec_dai); if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); + asoc_simple_clk_disable(dai_props->cpu_dai); return ret; } @@ -230,9 +230,9 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(dai_props->cpu_dai); + asoc_simple_clk_disable(dai_props->cpu_dai); - asoc_simple_card_clk_disable(dai_props->codec_dai); + asoc_simple_clk_disable(dai_props->codec_dai); } EXPORT_SYMBOL_GPL(asoc_simple_shutdown); @@ -299,13 +299,13 @@ int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_convert_fixup(&dai_props->adata, params); + asoc_simple_convert_fixup(&dai_props->adata, params); return 0; } EXPORT_SYMBOL_GPL(asoc_simple_be_hw_params_fixup); -static int asoc_simple_card_init_dai(struct snd_soc_dai *dai, +static int asoc_simple_init_dai(struct snd_soc_dai *dai, struct asoc_simple_dai *simple_dai) { int ret; @@ -343,13 +343,13 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); + ret = asoc_simple_init_dai(rtd->codec_dai, + dai_props->codec_dai); if (ret < 0) return ret; - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); + ret = asoc_simple_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); if (ret < 0) return ret; @@ -357,16 +357,16 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) } EXPORT_SYMBOL_GPL(asoc_simple_dai_init); -void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link) +void asoc_simple_canonicalize_platform(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ if (!dai_link->platforms->of_node) dai_link->platforms->of_node = dai_link->cpu_of_node; } -EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_platform); +EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_platform); -void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, - int is_single_links) +void asoc_simple_canonicalize_cpu(struct snd_soc_dai_link *dai_link, + int is_single_links) { /* * In soc_bind_dai_link() will check cpu name after @@ -380,9 +380,9 @@ void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, if (is_single_links) dai_link->cpu_dai_name = NULL; } -EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_cpu); +EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_cpu); -int asoc_simple_card_clean_reference(struct snd_soc_card *card) +int asoc_simple_clean_reference(struct snd_soc_card *card) { struct snd_soc_dai_link *dai_link; int i; @@ -393,10 +393,10 @@ int asoc_simple_card_clean_reference(struct snd_soc_card *card) } return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_clean_reference); +EXPORT_SYMBOL_GPL(asoc_simple_clean_reference); -int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, - char *prefix) +int asoc_simple_parse_routing(struct snd_soc_card *card, + char *prefix) { struct device_node *node = card->dev->of_node; char prop[128]; @@ -411,10 +411,10 @@ int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, return snd_soc_of_parse_audio_routing(card, prop); } -EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_routing); +EXPORT_SYMBOL_GPL(asoc_simple_parse_routing); -int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, - char *prefix) +int asoc_simple_parse_widgets(struct snd_soc_card *card, + char *prefix) { struct device_node *node = card->dev->of_node; char prop[128]; @@ -430,11 +430,11 @@ int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, /* no widgets is not error */ return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets); +EXPORT_SYMBOL_GPL(asoc_simple_parse_widgets); -int asoc_simple_card_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, - int is_hp, char *prefix) +int asoc_simple_init_jack(struct snd_soc_card *card, + struct asoc_simple_jack *sjack, + int is_hp, char *prefix) { struct device *dev = card->dev; enum of_gpio_flags flags; @@ -485,10 +485,10 @@ int asoc_simple_card_init_jack(struct snd_soc_card *card, return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_init_jack); +EXPORT_SYMBOL_GPL(asoc_simple_init_jack); -int asoc_simple_card_init_priv(struct asoc_simple_priv *priv, - struct link_info *li) +int asoc_simple_init_priv(struct asoc_simple_priv *priv, + struct link_info *li) { struct snd_soc_card *card = simple_priv_to_card(priv); struct device *dev = simple_priv_to_dev(priv); @@ -513,7 +513,7 @@ int asoc_simple_card_init_priv(struct asoc_simple_priv *priv, * * "platform" might be removed * see - * simple-card-utils.c :: asoc_simple_card_canonicalize_platform() + * simple-card-utils.c :: asoc_simple_canonicalize_platform() */ for (i = 0; i < li->link; i++) { dai_link[i].codecs = &dai_props[i].codecs; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d8560fb1f5de..dd9ac60d61f1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -25,11 +25,11 @@ static const struct snd_soc_ops simple_ops = { .hw_params = asoc_simple_hw_params, }; -static int asoc_simple_card_parse_dai(struct device_node *node, - struct snd_soc_dai_link_component *dlc, - struct device_node **dai_of_node, - const char **dai_name, - int *is_single_link) +static int asoc_simple_parse_dai(struct device_node *node, + struct snd_soc_dai_link_component *dlc, + struct device_node **dai_of_node, + const char **dai_name, + int *is_single_link) { struct of_phandle_args args; int ret; @@ -73,15 +73,15 @@ static int asoc_simple_card_parse_dai(struct device_node *node, static void simple_parse_convert(struct device *dev, struct device_node *np, - struct asoc_simple_card_data *adata) + struct asoc_simple_data *adata) { struct device_node *top = dev->of_node; struct device_node *node = of_get_parent(np); - asoc_simple_card_parse_convert(dev, top, PREFIX, adata); - asoc_simple_card_parse_convert(dev, node, PREFIX, adata); - asoc_simple_card_parse_convert(dev, node, NULL, adata); - asoc_simple_card_parse_convert(dev, np, NULL, adata); + asoc_simple_parse_convert(dev, top, PREFIX, adata); + asoc_simple_parse_convert(dev, node, PREFIX, adata); + asoc_simple_parse_convert(dev, node, NULL, adata); + asoc_simple_parse_convert(dev, np, NULL, adata); of_node_put(node); } @@ -156,21 +156,21 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai = dai_props->cpu_dai = &priv->dais[li->dais++]; - ret = asoc_simple_card_parse_cpu(np, dai_link, &is_single_links); + ret = asoc_simple_parse_cpu(np, dai_link, &is_single_links); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai); + ret = asoc_simple_parse_clk_cpu(dev, np, dai_link, dai); if (ret < 0) return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "fe.%s", - dai_link->cpu_dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "fe.%s", + dai_link->cpu_dai_name); if (ret < 0) return ret; - asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); + asoc_simple_canonicalize_cpu(dai_link, is_single_links); } else { struct snd_soc_codec_conf *cconf; @@ -189,17 +189,17 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, cconf = dai_props->codec_conf = &priv->codec_conf[li->conf++]; - ret = asoc_simple_card_parse_codec(np, dai_link); + ret = asoc_simple_parse_codec(np, dai_link); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai); + ret = asoc_simple_parse_clk_codec(dev, np, dai_link, dai); if (ret < 0) return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "be.%s", - codecs->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "be.%s", + codecs->dai_name); if (ret < 0) return ret; @@ -215,14 +215,14 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, simple_parse_convert(dev, np, &dai_props->adata); simple_parse_mclk_fs(top, np, codec, dai_props, prefix); - asoc_simple_card_canonicalize_platform(dai_link); + asoc_simple_canonicalize_platform(dai_link); - ret = asoc_simple_card_of_parse_tdm(np, dai); + ret = asoc_simple_parse_tdm(np, dai); if (ret) return ret; - ret = asoc_simple_card_parse_daifmt(dev, node, codec, - prefix, &dai_link->dai_fmt); + ret = asoc_simple_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); if (ret < 0) return ret; @@ -280,53 +280,53 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, codec_dai = dai_props->codec_dai = &priv->dais[li->dais++]; - ret = asoc_simple_card_parse_daifmt(dev, node, codec, - prefix, &dai_link->dai_fmt); + ret = asoc_simple_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); if (ret < 0) goto dai_link_of_err; simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); - ret = asoc_simple_card_parse_cpu(cpu, dai_link, &single_cpu); + ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_codec(codec, dai_link); + ret = asoc_simple_parse_codec(codec, dai_link); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_platform(plat, dai_link); + ret = asoc_simple_parse_platform(plat, dai_link); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_of_parse_tdm(cpu, cpu_dai); + ret = asoc_simple_parse_tdm(cpu, cpu_dai); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_of_parse_tdm(codec, codec_dai); + ret = asoc_simple_parse_tdm(codec, codec_dai); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_clk_cpu(dev, cpu, dai_link, cpu_dai); + ret = asoc_simple_parse_clk_cpu(dev, cpu, dai_link, cpu_dai); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_clk_codec(dev, codec, dai_link, codec_dai); + ret = asoc_simple_parse_clk_codec(dev, codec, dai_link, codec_dai); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "%s-%s", - dai_link->cpu_dai_name, - dai_link->codecs->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "%s-%s", + dai_link->cpu_dai_name, + dai_link->codecs->dai_name); if (ret < 0) goto dai_link_of_err; dai_link->ops = &simple_ops; dai_link->init = asoc_simple_dai_init; - asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); - asoc_simple_card_canonicalize_platform(dai_link); + asoc_simple_canonicalize_cpu(dai_link, single_cpu); + asoc_simple_canonicalize_platform(dai_link); dai_link_of_err: of_node_put(plat); @@ -361,7 +361,7 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, /* loop for all dai-link */ do { - struct asoc_simple_card_data adata; + struct asoc_simple_data adata; struct device_node *codec; struct device_node *np; int num = of_get_child_count(node); @@ -451,11 +451,11 @@ static int simple_parse_of(struct asoc_simple_priv *priv) if (!top) return -EINVAL; - ret = asoc_simple_card_of_parse_widgets(card, PREFIX); + ret = asoc_simple_parse_widgets(card, PREFIX); if (ret < 0) return ret; - ret = asoc_simple_card_of_parse_routing(card, PREFIX); + ret = asoc_simple_parse_routing(card, PREFIX); if (ret < 0) return ret; @@ -481,7 +481,7 @@ static int simple_parse_of(struct asoc_simple_priv *priv) return ret; } - ret = asoc_simple_card_parse_card_name(card, PREFIX); + ret = asoc_simple_parse_card_name(card, PREFIX); if (ret < 0) return ret; @@ -587,11 +587,11 @@ static int simple_soc_probe(struct snd_soc_card *card) struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(card); int ret; - ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX); + ret = asoc_simple_init_hp(card, &priv->hp_jack, PREFIX); if (ret < 0) return ret; - ret = asoc_simple_card_init_mic(card, &priv->mic_jack, PREFIX); + ret = asoc_simple_init_mic(card, &priv->mic_jack, PREFIX); if (ret < 0) return ret; @@ -622,7 +622,7 @@ static int simple_probe(struct platform_device *pdev) if (!li.link || !li.dais) return -EINVAL; - ret = asoc_simple_card_init_priv(priv, &li); + ret = asoc_simple_init_priv(priv, &li); if (ret < 0) return ret; @@ -691,7 +691,7 @@ static int simple_probe(struct platform_device *pdev) return 0; err: - asoc_simple_card_clean_reference(card); + asoc_simple_clean_reference(card); return ret; } @@ -700,7 +700,7 @@ static int simple_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - return asoc_simple_card_clean_reference(card); + return asoc_simple_clean_reference(card); } static const struct of_device_id simple_of_match[] = { -- cgit v1.2.3-59-g8ed1b From b87d37d0231fc731a04b599b7c05427ceffdf9f6 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Tue, 5 Mar 2019 11:26:45 +0000 Subject: ASoC: mchp-i2s-mcc: add driver for I2SC Multi-Channel Controller The Inter-IC Sound Controller (I2SMCC) provides a 5-wire, bidirectional, synchronous, digital audio link to external audio devices: I2SMCC_DIN, I2SMCC_DOUT, I2SMCC_WS, I2SMCC_CK, and I2SMCC_MCK pins. The I2SMCC complies with the Inter-IC Sound (I2S) bus specification and supports a Time Division Multiplexed (TDM) interface with external multi-channel audio codecs. The I2SMCC consists of a receiver, a transmitter and a common clock generator that can be enabled separately to provide Master, Slave or Controller modes with receiver and/or transmitter active. DMA Controller channels, separate for the receiver and for the transmitter, allow a continuous high bit rate data transfer without processor intervention to the following: - Audio CODECs in Master, Slave, or Controller mode - Stereo DAC or ADC through a dedicated I2S serial interface - Multi-channel or multiple stereo DACs or ADCs, using the TDM format This IP is embedded in Microchip's new sam9x60 SoC. Signed-off-by: Codrin Ciubotariu Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 14 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mchp-i2s-mcc.c | 974 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 990 insertions(+) create mode 100644 sound/soc/atmel/mchp-i2s-mcc.c (limited to 'sound') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 64f86f0b87e5..c473b9e463ab 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -109,4 +109,18 @@ config SND_SOC_MIKROE_PROTO using I2C over SDA (MPU Data Input) and SCL (MPU Clock Input) pins. Both playback and capture are supported. +config SND_MCHP_SOC_I2S_MCC + tristate "Microchip ASoC driver for boards using I2S MCC" + depends on OF && (ARCH_AT91 || COMPILE_TEST) + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Say Y or M if you want to add support for I2S Multi-Channel ASoC + driver on the following Microchip platforms: + - sam9x60 + + The I2SMCC complies with the Inter-IC Sound (I2S) bus specification + and supports a Time Division Multiplexed (TDM) interface with + external multi-channel audio codecs. + endif diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 9f41bfa0fea3..1f6890ed3738 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -4,11 +4,13 @@ snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o snd-soc-atmel-pcm-dma-objs := atmel-pcm-dma.o snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o snd-soc-atmel-i2s-objs := atmel-i2s.o +snd-soc-mchp-i2s-mcc-objs := mchp-i2s-mcc.o obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o obj-$(CONFIG_SND_ATMEL_SOC_I2S) += snd-soc-atmel-i2s.o +obj-$(CONFIG_SND_MCHP_SOC_I2S_MCC) += snd-soc-mchp-i2s-mcc.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c new file mode 100644 index 000000000000..86495883ca3f --- /dev/null +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -0,0 +1,974 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Driver for Microchip I2S Multi-channel controller +// +// Copyright (C) 2018 Microchip Technology Inc. and its subsidiaries +// +// Author: Codrin Ciubotariu + +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +/* + * ---- I2S Controller Register map ---- + */ +#define MCHP_I2SMCC_CR 0x0000 /* Control Register */ +#define MCHP_I2SMCC_MRA 0x0004 /* Mode Register A */ +#define MCHP_I2SMCC_MRB 0x0008 /* Mode Register B */ +#define MCHP_I2SMCC_SR 0x000C /* Status Register */ +#define MCHP_I2SMCC_IERA 0x0010 /* Interrupt Enable Register A */ +#define MCHP_I2SMCC_IDRA 0x0014 /* Interrupt Disable Register A */ +#define MCHP_I2SMCC_IMRA 0x0018 /* Interrupt Mask Register A */ +#define MCHP_I2SMCC_ISRA 0X001C /* Interrupt Status Register A */ + +#define MCHP_I2SMCC_IERB 0x0020 /* Interrupt Enable Register B */ +#define MCHP_I2SMCC_IDRB 0x0024 /* Interrupt Disable Register B */ +#define MCHP_I2SMCC_IMRB 0x0028 /* Interrupt Mask Register B */ +#define MCHP_I2SMCC_ISRB 0X002C /* Interrupt Status Register B */ + +#define MCHP_I2SMCC_RHR 0x0030 /* Receiver Holding Register */ +#define MCHP_I2SMCC_THR 0x0034 /* Transmitter Holding Register */ + +#define MCHP_I2SMCC_RHL0R 0x0040 /* Receiver Holding Left 0 Register */ +#define MCHP_I2SMCC_RHR0R 0x0044 /* Receiver Holding Right 0 Register */ + +#define MCHP_I2SMCC_RHL1R 0x0048 /* Receiver Holding Left 1 Register */ +#define MCHP_I2SMCC_RHR1R 0x004C /* Receiver Holding Right 1 Register */ + +#define MCHP_I2SMCC_RHL2R 0x0050 /* Receiver Holding Left 2 Register */ +#define MCHP_I2SMCC_RHR2R 0x0054 /* Receiver Holding Right 2 Register */ + +#define MCHP_I2SMCC_RHL3R 0x0058 /* Receiver Holding Left 3 Register */ +#define MCHP_I2SMCC_RHR3R 0x005C /* Receiver Holding Right 3 Register */ + +#define MCHP_I2SMCC_THL0R 0x0060 /* Transmitter Holding Left 0 Register */ +#define MCHP_I2SMCC_THR0R 0x0064 /* Transmitter Holding Right 0 Register */ + +#define MCHP_I2SMCC_THL1R 0x0068 /* Transmitter Holding Left 1 Register */ +#define MCHP_I2SMCC_THR1R 0x006C /* Transmitter Holding Right 1 Register */ + +#define MCHP_I2SMCC_THL2R 0x0070 /* Transmitter Holding Left 2 Register */ +#define MCHP_I2SMCC_THR2R 0x0074 /* Transmitter Holding Right 2 Register */ + +#define MCHP_I2SMCC_THL3R 0x0078 /* Transmitter Holding Left 3 Register */ +#define MCHP_I2SMCC_THR3R 0x007C /* Transmitter Holding Right 3 Register */ + +#define MCHP_I2SMCC_VERSION 0x00FC /* Version Register */ + +/* + * ---- Control Register (Write-only) ---- + */ +#define MCHP_I2SMCC_CR_RXEN BIT(0) /* Receiver Enable */ +#define MCHP_I2SMCC_CR_RXDIS BIT(1) /* Receiver Disable */ +#define MCHP_I2SMCC_CR_CKEN BIT(2) /* Clock Enable */ +#define MCHP_I2SMCC_CR_CKDIS BIT(3) /* Clock Disable */ +#define MCHP_I2SMCC_CR_TXEN BIT(4) /* Transmitter Enable */ +#define MCHP_I2SMCC_CR_TXDIS BIT(5) /* Transmitter Disable */ +#define MCHP_I2SMCC_CR_SWRST BIT(7) /* Software Reset */ + +/* + * ---- Mode Register A (Read/Write) ---- + */ +#define MCHP_I2SMCC_MRA_MODE_MASK GENMASK(0, 0) +#define MCHP_I2SMCC_MRA_MODE_SLAVE (0 << 0) +#define MCHP_I2SMCC_MRA_MODE_MASTER (1 << 0) + +#define MCHP_I2SMCC_MRA_DATALENGTH_MASK GENMASK(3, 1) +#define MCHP_I2SMCC_MRA_DATALENGTH_32_BITS (0 << 1) +#define MCHP_I2SMCC_MRA_DATALENGTH_24_BITS (1 << 1) +#define MCHP_I2SMCC_MRA_DATALENGTH_20_BITS (2 << 1) +#define MCHP_I2SMCC_MRA_DATALENGTH_18_BITS (3 << 1) +#define MCHP_I2SMCC_MRA_DATALENGTH_16_BITS (4 << 1) +#define MCHP_I2SMCC_MRA_DATALENGTH_16_BITS_COMPACT (5 << 1) +#define MCHP_I2SMCC_MRA_DATALENGTH_8_BITS (6 << 1) +#define MCHP_I2SMCC_MRA_DATALENGTH_8_BITS_COMPACT (7 << 1) + +#define MCHP_I2SMCC_MRA_WIRECFG_MASK GENMASK(5, 4) +#define MCHP_I2SMCC_MRA_WIRECFG_I2S_1_TDM_0 (0 << 4) +#define MCHP_I2SMCC_MRA_WIRECFG_I2S_2_TDM_1 (1 << 4) +#define MCHP_I2SMCC_MRA_WIRECFG_I2S_4_TDM_2 (2 << 4) +#define MCHP_I2SMCC_MRA_WIRECFG_TDM_3 (3 << 4) + +#define MCHP_I2SMCC_MRA_FORMAT_MASK GENMASK(7, 6) +#define MCHP_I2SMCC_MRA_FORMAT_I2S (0 << 6) +#define MCHP_I2SMCC_MRA_FORMAT_LJ (1 << 6) /* Left Justified */ +#define MCHP_I2SMCC_MRA_FORMAT_TDM (2 << 6) +#define MCHP_I2SMCC_MRA_FORMAT_TDMLJ (3 << 6) + +/* Transmitter uses one DMA channel ... */ +/* Left audio samples duplicated to right audio channel */ +#define MCHP_I2SMCC_MRA_RXMONO BIT(8) + +/* I2SDO output of I2SC is internally connected to I2SDI input */ +#define MCHP_I2SMCC_MRA_RXLOOP BIT(9) + +/* Receiver uses one DMA channel ... */ +/* Left audio samples duplicated to right audio channel */ +#define MCHP_I2SMCC_MRA_TXMONO BIT(10) + +/* x sample transmitted when underrun */ +#define MCHP_I2SMCC_MRA_TXSAME_ZERO (0 << 11) /* Zero sample */ +#define MCHP_I2SMCC_MRA_TXSAME_PREVIOUS (1 << 11) /* Previous sample */ + +/* select between peripheral clock and generated clock */ +#define MCHP_I2SMCC_MRA_SRCCLK_PCLK (0 << 12) +#define MCHP_I2SMCC_MRA_SRCCLK_GCLK (1 << 12) + +/* Number of TDM Channels - 1 */ +#define MCHP_I2SMCC_MRA_NBCHAN_MASK GENMASK(15, 13) +#define MCHP_I2SMCC_MRA_NBCHAN(ch) \ + ((((ch) - 1) << 13) & MCHP_I2SMCC_MRA_NBCHAN_MASK) + +/* Selected Clock to I2SMCC Master Clock ratio */ +#define MCHP_I2SMCC_MRA_IMCKDIV_MASK GENMASK(21, 16) +#define MCHP_I2SMCC_MRA_IMCKDIV(div) \ + (((div) << 16) & MCHP_I2SMCC_MRA_IMCKDIV_MASK) + +/* TDM Frame Synchronization */ +#define MCHP_I2SMCC_MRA_TDMFS_MASK GENMASK(23, 22) +#define MCHP_I2SMCC_MRA_TDMFS_SLOT (0 << 22) +#define MCHP_I2SMCC_MRA_TDMFS_HALF (1 << 22) +#define MCHP_I2SMCC_MRA_TDMFS_BIT (2 << 22) + +/* Selected Clock to I2SMC Serial Clock ratio */ +#define MCHP_I2SMCC_MRA_ISCKDIV_MASK GENMASK(29, 24) +#define MCHP_I2SMCC_MRA_ISCKDIV(div) \ + (((div) << 24) & MCHP_I2SMCC_MRA_ISCKDIV_MASK) + +/* Master Clock mode */ +#define MCHP_I2SMCC_MRA_IMCKMODE_MASK GENMASK(30, 30) +/* 0: No master clock generated*/ +#define MCHP_I2SMCC_MRA_IMCKMODE_NONE (0 << 30) +/* 1: master clock generated (internally generated clock drives I2SMCK pin) */ +#define MCHP_I2SMCC_MRA_IMCKMODE_GEN (1 << 30) + +/* Slot Width */ +/* 0: slot is 32 bits wide for DATALENGTH = 18/20/24 bits. */ +/* 1: slot is 24 bits wide for DATALENGTH = 18/20/24 bits. */ +#define MCHP_I2SMCC_MRA_IWS BIT(31) + +/* + * ---- Mode Register B (Read/Write) ---- + */ +/* all enabled I2S left channels are filled first, then I2S right channels */ +#define MCHP_I2SMCC_MRB_CRAMODE_LEFT_FIRST (0 << 0) +/* + * an enabled I2S left channel is filled, then the corresponding right + * channel, until all channels are filled + */ +#define MCHP_I2SMCC_MRB_CRAMODE_REGULAR (1 << 0) + +#define MCHP_I2SMCC_MRB_FIFOEN BIT(1) + +#define MCHP_I2SMCC_MRB_DMACHUNK_MASK GENMASK(9, 8) +#define MCHP_I2SMCC_MRB_DMACHUNK(no_words) \ + (((fls(no_words) - 1) << 8) & MCHP_I2SMCC_MRB_DMACHUNK_MASK) + +#define MCHP_I2SMCC_MRB_CLKSEL_MASK GENMASK(16, 16) +#define MCHP_I2SMCC_MRB_CLKSEL_EXT (0 << 16) +#define MCHP_I2SMCC_MRB_CLKSEL_INT (1 << 16) + +/* + * ---- Status Registers (Read-only) ---- + */ +#define MCHP_I2SMCC_SR_RXEN BIT(0) /* Receiver Enabled */ +#define MCHP_I2SMCC_SR_TXEN BIT(4) /* Transmitter Enabled */ + +/* + * ---- Interrupt Enable/Disable/Mask/Status Registers A ---- + */ +#define MCHP_I2SMCC_INT_TXRDY_MASK(ch) GENMASK((ch) - 1, 0) +#define MCHP_I2SMCC_INT_TXRDYCH(ch) BIT(ch) +#define MCHP_I2SMCC_INT_TXUNF_MASK(ch) GENMASK((ch) + 7, 8) +#define MCHP_I2SMCC_INT_TXUNFCH(ch) BIT((ch) + 8) +#define MCHP_I2SMCC_INT_RXRDY_MASK(ch) GENMASK((ch) + 15, 16) +#define MCHP_I2SMCC_INT_RXRDYCH(ch) BIT((ch) + 16) +#define MCHP_I2SMCC_INT_RXOVF_MASK(ch) GENMASK((ch) + 23, 24) +#define MCHP_I2SMCC_INT_RXOVFCH(ch) BIT((ch) + 24) + +/* + * ---- Interrupt Enable/Disable/Mask/Status Registers B ---- + */ +#define MCHP_I2SMCC_INT_WERR BIT(0) +#define MCHP_I2SMCC_INT_TXFFRDY BIT(8) +#define MCHP_I2SMCC_INT_TXFFEMP BIT(9) +#define MCHP_I2SMCC_INT_RXFFRDY BIT(12) +#define MCHP_I2SMCC_INT_RXFFFUL BIT(13) + +/* + * ---- Version Register (Read-only) ---- + */ +#define MCHP_I2SMCC_VERSION_MASK GENMASK(11, 0) + +#define MCHP_I2SMCC_MAX_CHANNELS 8 +#define MCHP_I2MCC_TDM_SLOT_WIDTH 32 + +static const struct regmap_config mchp_i2s_mcc_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = MCHP_I2SMCC_VERSION, +}; + +struct mchp_i2s_mcc_dev { + struct wait_queue_head wq_txrdy; + struct wait_queue_head wq_rxrdy; + struct device *dev; + struct regmap *regmap; + struct clk *pclk; + struct clk *gclk; + struct snd_dmaengine_dai_dma_data playback; + struct snd_dmaengine_dai_dma_data capture; + unsigned int fmt; + unsigned int sysclk; + unsigned int frame_length; + int tdm_slots; + int channels; + int gclk_use:1; + int gclk_running:1; + int tx_rdy:1; + int rx_rdy:1; +}; + +static irqreturn_t mchp_i2s_mcc_interrupt(int irq, void *dev_id) +{ + struct mchp_i2s_mcc_dev *dev = dev_id; + u32 sra, imra, srb, imrb, pendinga, pendingb, idra = 0; + irqreturn_t ret = IRQ_NONE; + + regmap_read(dev->regmap, MCHP_I2SMCC_IMRA, &imra); + regmap_read(dev->regmap, MCHP_I2SMCC_ISRA, &sra); + pendinga = imra & sra; + + regmap_read(dev->regmap, MCHP_I2SMCC_IMRB, &imrb); + regmap_read(dev->regmap, MCHP_I2SMCC_ISRB, &srb); + pendingb = imrb & srb; + + if (!pendinga && !pendingb) + return IRQ_NONE; + + /* + * Tx/Rx ready interrupts are enabled when stopping only, to assure + * availability and to disable clocks if necessary + */ + idra |= pendinga & (MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels) | + MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels)); + if (idra) + ret = IRQ_HANDLED; + + if ((imra & MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels)) && + (imra & MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels)) == + (idra & MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels))) { + dev->tx_rdy = 1; + wake_up_interruptible(&dev->wq_txrdy); + } + if ((imra & MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels)) && + (imra & MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels)) == + (idra & MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels))) { + dev->rx_rdy = 1; + wake_up_interruptible(&dev->wq_rxrdy); + } + regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, idra); + + return ret; +} + +static int mchp_i2s_mcc_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dev->dev, "%s() clk_id=%d freq=%u dir=%d\n", + __func__, clk_id, freq, dir); + + /* We do not need SYSCLK */ + if (dir == SND_SOC_CLOCK_IN) + return 0; + + dev->sysclk = freq; + + return 0; +} + +static int mchp_i2s_mcc_set_bclk_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dev->dev, "%s() ratio=%u\n", __func__, ratio); + + dev->frame_length = ratio; + + return 0; +} + +static int mchp_i2s_mcc_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dev->dev, "%s() fmt=%#x\n", __func__, fmt); + + /* We don't support any kind of clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + + /* We can't generate only FSYNC */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFS) + return -EINVAL; + + /* We can only reconfigure the IP when it's stopped */ + if (fmt & SND_SOC_DAIFMT_CONT) + return -EINVAL; + + dev->fmt = fmt; + + return 0; +} + +static int mchp_i2s_mcc_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dev->dev, + "%s() tx_mask=0x%08x rx_mask=0x%08x slots=%d width=%d\n", + __func__, tx_mask, rx_mask, slots, slot_width); + + if (slots < 0 || slots > MCHP_I2SMCC_MAX_CHANNELS || + slot_width != MCHP_I2MCC_TDM_SLOT_WIDTH) + return -EINVAL; + + if (slots) { + /* We do not support daisy chain */ + if (rx_mask != GENMASK(slots - 1, 0) || + rx_mask != tx_mask) + return -EINVAL; + } + + dev->tdm_slots = slots; + dev->frame_length = slots * MCHP_I2MCC_TDM_SLOT_WIDTH; + + return 0; +} + +static int mchp_i2s_mcc_clk_get_rate_diff(struct clk *clk, + unsigned long rate, + struct clk **best_clk, + unsigned long *best_rate, + unsigned long *best_diff_rate) +{ + long round_rate; + unsigned int diff_rate; + + round_rate = clk_round_rate(clk, rate); + if (round_rate < 0) + return (int)round_rate; + + diff_rate = abs(rate - round_rate); + if (diff_rate < *best_diff_rate) { + *best_clk = clk; + *best_diff_rate = diff_rate; + *best_rate = rate; + } + + return 0; +} + +static int mchp_i2s_mcc_config_divs(struct mchp_i2s_mcc_dev *dev, + unsigned int bclk, unsigned int *mra) +{ + unsigned long clk_rate; + unsigned long lcm_rate; + unsigned long best_rate = 0; + unsigned long best_diff_rate = ~0; + unsigned int sysclk; + struct clk *best_clk = NULL; + int ret; + + /* For code simplification */ + if (!dev->sysclk) + sysclk = bclk; + else + sysclk = dev->sysclk; + + /* + * MCLK is Selected CLK / (2 * IMCKDIV), + * BCLK is Selected CLK / (2 * ISCKDIV); + * if IMCKDIV or ISCKDIV are 0, MCLK or BCLK = Selected CLK + */ + lcm_rate = lcm(sysclk, bclk); + if ((lcm_rate / sysclk % 2 == 1 && lcm_rate / sysclk > 2) || + (lcm_rate / bclk % 2 == 1 && lcm_rate / bclk > 2)) + lcm_rate *= 2; + + for (clk_rate = lcm_rate; + (clk_rate == sysclk || clk_rate / (sysclk * 2) <= GENMASK(5, 0)) && + (clk_rate == bclk || clk_rate / (bclk * 2) <= GENMASK(5, 0)); + clk_rate += lcm_rate) { + ret = mchp_i2s_mcc_clk_get_rate_diff(dev->gclk, clk_rate, + &best_clk, &best_rate, + &best_diff_rate); + if (ret) { + dev_err(dev->dev, "gclk error for rate %lu: %d", + clk_rate, ret); + } else { + if (!best_diff_rate) { + dev_dbg(dev->dev, "found perfect rate on gclk: %lu\n", + clk_rate); + break; + } + } + + ret = mchp_i2s_mcc_clk_get_rate_diff(dev->pclk, clk_rate, + &best_clk, &best_rate, + &best_diff_rate); + if (ret) { + dev_err(dev->dev, "pclk error for rate %lu: %d", + clk_rate, ret); + } else { + if (!best_diff_rate) { + dev_dbg(dev->dev, "found perfect rate on pclk: %lu\n", + clk_rate); + break; + } + } + } + + /* check if clocks returned only errors */ + if (!best_clk) { + dev_err(dev->dev, "unable to change rate to clocks\n"); + return -EINVAL; + } + + dev_dbg(dev->dev, "source CLK is %s with rate %lu, diff %lu\n", + best_clk == dev->pclk ? "pclk" : "gclk", + best_rate, best_diff_rate); + + /* set the rate */ + ret = clk_set_rate(best_clk, best_rate); + if (ret) { + dev_err(dev->dev, "unable to set rate %lu to %s: %d\n", + best_rate, best_clk == dev->pclk ? "PCLK" : "GCLK", + ret); + return ret; + } + + /* Configure divisors */ + if (dev->sysclk) + *mra |= MCHP_I2SMCC_MRA_IMCKDIV(best_rate / (2 * sysclk)); + *mra |= MCHP_I2SMCC_MRA_ISCKDIV(best_rate / (2 * bclk)); + + if (best_clk == dev->gclk) { + *mra |= MCHP_I2SMCC_MRA_SRCCLK_GCLK; + ret = clk_prepare(dev->gclk); + if (ret < 0) + dev_err(dev->dev, "unable to prepare GCLK: %d\n", ret); + else + dev->gclk_use = 1; + } else { + *mra |= MCHP_I2SMCC_MRA_SRCCLK_PCLK; + dev->gclk_use = 0; + } + + return 0; +} + +static int mchp_i2s_mcc_is_running(struct mchp_i2s_mcc_dev *dev) +{ + u32 sr; + + regmap_read(dev->regmap, MCHP_I2SMCC_SR, &sr); + return !!(sr & (MCHP_I2SMCC_SR_TXEN | MCHP_I2SMCC_SR_RXEN)); +} + +static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + u32 mra = 0; + u32 mrb = 0; + unsigned int channels = params_channels(params); + unsigned int frame_length = dev->frame_length; + unsigned int bclk_rate; + int set_divs = 0; + int ret; + bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + dev_dbg(dev->dev, "%s() rate=%u format=%#x width=%u channels=%u\n", + __func__, params_rate(params), params_format(params), + params_width(params), params_channels(params)); + + switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + if (dev->tdm_slots) { + dev_err(dev->dev, "I2S with TDM is not supported\n"); + return -EINVAL; + } + mra |= MCHP_I2SMCC_MRA_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + if (dev->tdm_slots) { + dev_err(dev->dev, "Left-Justified with TDM is not supported\n"); + return -EINVAL; + } + mra |= MCHP_I2SMCC_MRA_FORMAT_LJ; + break; + case SND_SOC_DAIFMT_DSP_A: + mra |= MCHP_I2SMCC_MRA_FORMAT_TDM; + break; + default: + dev_err(dev->dev, "unsupported bus format\n"); + return -EINVAL; + } + + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* cpu is BCLK and LRC master */ + mra |= MCHP_I2SMCC_MRA_MODE_MASTER; + if (dev->sysclk) + mra |= MCHP_I2SMCC_MRA_IMCKMODE_GEN; + set_divs = 1; + break; + case SND_SOC_DAIFMT_CBS_CFM: + /* cpu is BCLK master */ + mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT; + set_divs = 1; + /* fall through */ + case SND_SOC_DAIFMT_CBM_CFM: + /* cpu is slave */ + mra |= MCHP_I2SMCC_MRA_MODE_SLAVE; + if (dev->sysclk) + dev_warn(dev->dev, "Unable to generate MCLK in Slave mode\n"); + break; + default: + dev_err(dev->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + if (dev->fmt & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + switch (channels) { + case 1: + if (is_playback) + mra |= MCHP_I2SMCC_MRA_TXMONO; + else + mra |= MCHP_I2SMCC_MRA_RXMONO; + break; + case 2: + break; + default: + dev_err(dev->dev, "unsupported number of audio channels\n"); + return -EINVAL; + } + + if (!frame_length) + frame_length = 2 * params_physical_width(params); + } else if (dev->fmt & SND_SOC_DAIFMT_DSP_A) { + if (dev->tdm_slots) { + if (channels % 2 && channels * 2 <= dev->tdm_slots) { + /* + * Duplicate data for even-numbered channels + * to odd-numbered channels + */ + if (is_playback) + mra |= MCHP_I2SMCC_MRA_TXMONO; + else + mra |= MCHP_I2SMCC_MRA_RXMONO; + } + channels = dev->tdm_slots; + } + + mra |= MCHP_I2SMCC_MRA_NBCHAN(channels); + if (!frame_length) + frame_length = channels * MCHP_I2MCC_TDM_SLOT_WIDTH; + } + + /* + * We must have the same burst size configured + * in the DMA transfer and in out IP + */ + mrb |= MCHP_I2SMCC_MRB_DMACHUNK(channels); + if (is_playback) + dev->playback.maxburst = 1 << (fls(channels) - 1); + else + dev->capture.maxburst = 1 << (fls(channels) - 1); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + mra |= MCHP_I2SMCC_MRA_DATALENGTH_8_BITS; + break; + case SNDRV_PCM_FORMAT_S16_LE: + mra |= MCHP_I2SMCC_MRA_DATALENGTH_16_BITS; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + mra |= MCHP_I2SMCC_MRA_DATALENGTH_18_BITS | + MCHP_I2SMCC_MRA_IWS; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + mra |= MCHP_I2SMCC_MRA_DATALENGTH_20_BITS | + MCHP_I2SMCC_MRA_IWS; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + mra |= MCHP_I2SMCC_MRA_DATALENGTH_24_BITS | + MCHP_I2SMCC_MRA_IWS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + mra |= MCHP_I2SMCC_MRA_DATALENGTH_24_BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + mra |= MCHP_I2SMCC_MRA_DATALENGTH_32_BITS; + break; + default: + dev_err(dev->dev, "unsupported size/endianness for audio samples\n"); + return -EINVAL; + } + + /* + * If we are already running, the wanted setup must be + * the same with the one that's currently ongoing + */ + if (mchp_i2s_mcc_is_running(dev)) { + u32 mra_cur; + u32 mrb_cur; + + regmap_read(dev->regmap, MCHP_I2SMCC_MRA, &mra_cur); + regmap_read(dev->regmap, MCHP_I2SMCC_MRB, &mrb_cur); + if (mra != mra_cur || mrb != mrb_cur) + return -EINVAL; + + return 0; + } + + /* Save the number of channels to know what interrupts to enable */ + dev->channels = channels; + + if (set_divs) { + bclk_rate = frame_length * params_rate(params); + ret = mchp_i2s_mcc_config_divs(dev, bclk_rate, &mra); + if (ret) { + dev_err(dev->dev, "unable to configure the divisors: %d\n", + ret); + return ret; + } + } + + ret = regmap_write(dev->regmap, MCHP_I2SMCC_MRA, mra); + if (ret < 0) + return ret; + return regmap_write(dev->regmap, MCHP_I2SMCC_MRB, mrb); +} + +static int mchp_i2s_mcc_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + long err; + + if (is_playback) { + err = wait_event_interruptible_timeout(dev->wq_txrdy, + dev->tx_rdy, + msecs_to_jiffies(500)); + } else { + err = wait_event_interruptible_timeout(dev->wq_rxrdy, + dev->rx_rdy, + msecs_to_jiffies(500)); + } + + if (err == 0) { + u32 idra; + + dev_warn_once(dev->dev, "Timeout waiting for %s\n", + is_playback ? "Tx ready" : "Rx ready"); + if (is_playback) + idra = MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels); + else + idra = MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels); + regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, idra); + } + + if (!mchp_i2s_mcc_is_running(dev)) { + regmap_write(dev->regmap, MCHP_I2SMCC_CR, MCHP_I2SMCC_CR_CKDIS); + + if (dev->gclk_running) { + clk_disable_unprepare(dev->gclk); + dev->gclk_running = 0; + } + } + + return 0; +} + +static int mchp_i2s_mcc_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + u32 cr = 0; + u32 iera = 0; + u32 sr; + int err; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (is_playback) + cr = MCHP_I2SMCC_CR_TXEN | MCHP_I2SMCC_CR_CKEN; + else + cr = MCHP_I2SMCC_CR_RXEN | MCHP_I2SMCC_CR_CKEN; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_read(dev->regmap, MCHP_I2SMCC_SR, &sr); + if (is_playback && (sr & MCHP_I2SMCC_SR_TXEN)) { + cr = MCHP_I2SMCC_CR_TXDIS; + dev->tx_rdy = 0; + /* + * Enable Tx Ready interrupts on all channels + * to assure all data is sent + */ + iera = MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels); + } else if (!is_playback && (sr & MCHP_I2SMCC_SR_RXEN)) { + cr = MCHP_I2SMCC_CR_RXDIS; + dev->rx_rdy = 0; + /* + * Enable Rx Ready interrupts on all channels + * to assure all data is received + */ + iera = MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels); + } + break; + default: + return -EINVAL; + } + + if ((cr & MCHP_I2SMCC_CR_CKEN) && dev->gclk_use && + !dev->gclk_running) { + err = clk_enable(dev->gclk); + if (err) { + dev_err_once(dev->dev, "failed to enable GCLK: %d\n", + err); + } else { + dev->gclk_running = 1; + } + } + + regmap_write(dev->regmap, MCHP_I2SMCC_IERA, iera); + regmap_write(dev->regmap, MCHP_I2SMCC_CR, cr); + + return 0; +} + +static int mchp_i2s_mcc_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + + /* Software reset the IP if it's not running */ + if (!mchp_i2s_mcc_is_running(dev)) { + return regmap_write(dev->regmap, MCHP_I2SMCC_CR, + MCHP_I2SMCC_CR_SWRST); + } + + return 0; +} + +static const struct snd_soc_dai_ops mchp_i2s_mcc_dai_ops = { + .set_sysclk = mchp_i2s_mcc_set_sysclk, + .set_bclk_ratio = mchp_i2s_mcc_set_bclk_ratio, + .startup = mchp_i2s_mcc_startup, + .trigger = mchp_i2s_mcc_trigger, + .hw_params = mchp_i2s_mcc_hw_params, + .hw_free = mchp_i2s_mcc_hw_free, + .set_fmt = mchp_i2s_mcc_set_dai_fmt, + .set_tdm_slot = mchp_i2s_mcc_set_dai_tdm_slot, +}; + +static int mchp_i2s_mcc_dai_probe(struct snd_soc_dai *dai) +{ + struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + + init_waitqueue_head(&dev->wq_txrdy); + init_waitqueue_head(&dev->wq_rxrdy); + + snd_soc_dai_init_dma_data(dai, &dev->playback, &dev->capture); + + return 0; +} + +#define MCHP_I2SMCC_RATES SNDRV_PCM_RATE_8000_192000 + +#define MCHP_I2SMCC_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver mchp_i2s_mcc_dai = { + .probe = mchp_i2s_mcc_dai_probe, + .playback = { + .stream_name = "I2SMCC-Playback", + .channels_min = 1, + .channels_max = 8, + .rates = MCHP_I2SMCC_RATES, + .formats = MCHP_I2SMCC_FORMATS, + }, + .capture = { + .stream_name = "I2SMCC-Capture", + .channels_min = 1, + .channels_max = 8, + .rates = MCHP_I2SMCC_RATES, + .formats = MCHP_I2SMCC_FORMATS, + }, + .ops = &mchp_i2s_mcc_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, + .symmetric_channels = 1, +}; + +static const struct snd_soc_component_driver mchp_i2s_mcc_component = { + .name = "mchp-i2s-mcc", +}; + +#ifdef CONFIG_OF +static const struct of_device_id mchp_i2s_mcc_dt_ids[] = { + { + .compatible = "microchip,sam9x60-i2smcc", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, mchp_i2s_mcc_dt_ids); +#endif + +static int mchp_i2s_mcc_probe(struct platform_device *pdev) +{ + struct mchp_i2s_mcc_dev *dev; + struct resource *mem; + struct regmap *regmap; + void __iomem *base; + u32 version; + int irq; + int err; + + dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); + if (!dev) + return -ENOMEM; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(base)) + return PTR_ERR(base); + + regmap = devm_regmap_init_mmio(&pdev->dev, base, + &mchp_i2s_mcc_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) + return irq; + + err = devm_request_irq(&pdev->dev, irq, mchp_i2s_mcc_interrupt, 0, + dev_name(&pdev->dev), dev); + if (err) + return err; + + dev->pclk = devm_clk_get(&pdev->dev, "pclk"); + if (IS_ERR(dev->pclk)) { + err = PTR_ERR(dev->pclk); + dev_err(&pdev->dev, + "failed to get the peripheral clock: %d\n", err); + return err; + } + + /* Get the optional generated clock */ + dev->gclk = devm_clk_get(&pdev->dev, "gclk"); + if (IS_ERR(dev->gclk)) { + if (PTR_ERR(dev->gclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + dev_warn(&pdev->dev, + "generated clock not found: %d\n", err); + dev->gclk = NULL; + } + + dev->dev = &pdev->dev; + dev->regmap = regmap; + platform_set_drvdata(pdev, dev); + + err = clk_prepare_enable(dev->pclk); + if (err) { + dev_err(&pdev->dev, + "failed to enable the peripheral clock: %d\n", err); + return err; + } + + err = devm_snd_soc_register_component(&pdev->dev, + &mchp_i2s_mcc_component, + &mchp_i2s_mcc_dai, 1); + if (err) { + dev_err(&pdev->dev, "failed to register DAI: %d\n", err); + clk_disable_unprepare(dev->pclk); + return err; + } + + dev->playback.addr = (dma_addr_t)mem->start + MCHP_I2SMCC_THR; + dev->capture.addr = (dma_addr_t)mem->start + MCHP_I2SMCC_RHR; + + err = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (err) { + dev_err(&pdev->dev, "failed to register PCM: %d\n", err); + clk_disable_unprepare(dev->pclk); + return err; + } + + /* Get IP version. */ + regmap_read(dev->regmap, MCHP_I2SMCC_VERSION, &version); + dev_info(&pdev->dev, "hw version: %#lx\n", + version & MCHP_I2SMCC_VERSION_MASK); + + return 0; +} + +static int mchp_i2s_mcc_remove(struct platform_device *pdev) +{ + struct mchp_i2s_mcc_dev *dev = platform_get_drvdata(pdev); + + clk_disable_unprepare(dev->pclk); + + return 0; +} + +static struct platform_driver mchp_i2s_mcc_driver = { + .driver = { + .name = "mchp_i2s_mcc", + .of_match_table = of_match_ptr(mchp_i2s_mcc_dt_ids), + }, + .probe = mchp_i2s_mcc_probe, + .remove = mchp_i2s_mcc_remove, +}; +module_platform_driver(mchp_i2s_mcc_driver); + +MODULE_DESCRIPTION("Microchip I2S Multi-Channel Controller driver"); +MODULE_AUTHOR("Codrin Ciubotariu "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-59-g8ed1b From c95e3a4b96293403a427b5185e60fad28af51fdd Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Wed, 20 Mar 2019 19:38:44 -0700 Subject: ASoC: tlv320aic32x4: Properly Set Processing Blocks Different processing blocks are required for different sampling rates and power parameters. Set the processing blocks based on this information. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 56 ++++++++++++++++++++++++++-------------- 1 file changed, 36 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1aa8f5aa4225..71a93fbc5971 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -59,6 +59,8 @@ struct aic32x4_rate_divs { u8 nadc; u8 madc; u8 blck_N; + u8 r_block; + u8 p_block; }; struct aic32x4_priv { @@ -307,34 +309,34 @@ static const struct snd_kcontrol_new aic32x4_snd_controls[] = { static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 8k rate */ - {12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24}, - {24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24}, - {25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24}, + {12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24, 1, 1}, + {24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24, 1, 1}, + {25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24, 1, 1}, /* 11.025k rate */ - {12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16}, - {24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16}, + {12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16, 1, 1}, + {24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16, 1, 1}, /* 16k rate */ - {12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12}, - {24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12}, - {25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12}, + {12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12, 1, 1}, + {24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12, 1, 1}, + {25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12, 1, 1}, /* 22.05k rate */ - {12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8}, - {24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8}, - {25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8}, + {12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8, 1, 1}, + {24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8, 1, 1}, + {25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8, 1, 1}, /* 32k rate */ - {12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6}, - {24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6}, + {12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6, 1, 1}, + {24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6, 1, 1}, /* 44.1k rate */ - {12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4}, - {24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4}, - {25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4}, + {12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4, 1, 1}, + {24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4, 1, 1}, + {25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4, 1, 1}, /* 48k rate */ - {12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, - {24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, - {25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}, + {12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4, 1, 1}, + {24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4, 1, 1}, + {25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4, 1, 1}, /* 96k rate */ - {25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1}, + {25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1, 1, 9}, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -696,6 +698,18 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } +static int aic32x4_set_processing_blocks(struct snd_soc_component *component, + u8 r_block, u8 p_block) +{ + if (r_block > 18 || p_block > 25) + return -EINVAL; + + snd_soc_component_write(component, AIC32X4_ADCSPB, r_block); + snd_soc_component_write(component, AIC32X4_DACSPB, p_block); + + return 0; +} + static int aic32x4_setup_clocks(struct snd_soc_component *component, unsigned int sample_rate, unsigned int parent_rate) @@ -708,6 +722,8 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, return i; } + aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); + /* MCLK as PLL_CLKIN */ snd_soc_component_update_bits(component, AIC32X4_CLKMUX, AIC32X4_PLL_CLKIN_MASK, AIC32X4_PLL_CLKIN_MCLK << AIC32X4_PLL_CLKIN_SHIFT); -- cgit v1.2.3-59-g8ed1b From 16123412a6283bcf956f1a377f2e799a79c2b439 Mon Sep 17 00:00:00 2001 From: Piotr Stankiewicz Date: Wed, 20 Mar 2019 17:37:32 +0000 Subject: ASoC: lochnagar: Add driver to support Lochnagar 2 sound card Lochnagar is an evaluation and development board for Cirrus Logic Smart CODEC and Amp devices. It allows the connection of most Cirrus Logic devices on mini-cards, as well as allowing connection of various application processor systems to provide a full evaluation platform. Lochnagar 2 provides a set of line inputs/outputs, and a USB audio device. This driver adds support for these analog line connections and the Lochnagar side of the USB audio link. Signed-off-by: Piotr Stankiewicz Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- MAINTAINERS | 2 + sound/soc/codecs/Kconfig | 8 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lochnagar-sc.c | 266 ++++++++++++++++++++++++++++++++++++++++ 4 files changed, 278 insertions(+) create mode 100644 sound/soc/codecs/lochnagar-sc.c (limited to 'sound') diff --git a/MAINTAINERS b/MAINTAINERS index e17ebf70b548..1bc9f64ab0e4 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3799,6 +3799,7 @@ F: drivers/clk/clk-lochnagar.c F: drivers/mfd/lochnagar-i2c.c F: drivers/pinctrl/cirrus/pinctrl-lochnagar.c F: drivers/regulator/lochnagar-regulator.c +F: sound/soc/codecs/lochnagar-sc.c F: include/dt-bindings/clk/lochnagar.h F: include/dt-bindings/pinctrl/lochnagar.h F: include/linux/mfd/lochnagar* @@ -3806,6 +3807,7 @@ F: Documentation/devicetree/bindings/mfd/cirrus,lochnagar.txt F: Documentation/devicetree/bindings/clock/cirrus,lochnagar.txt F: Documentation/devicetree/bindings/pinctrl/cirrus,lochnagar.txt F: Documentation/devicetree/bindings/regulator/cirrus,lochnagar.txt +F: Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt CISCO FCOE HBA DRIVER M: Satish Kharat diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 419114edfd57..05f16632296b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -94,6 +94,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_JZ4725B_CODEC select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C + select SND_SOC_LOCHNAGAR_SC select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C @@ -688,6 +689,13 @@ config SND_SOC_ISABELLE config SND_SOC_LM49453 tristate +config SND_SOC_LOCHNAGAR_SC + tristate "Lochnagar Sound Card" + depends on MFD_LOCHNAGAR + help + This driver support the sound card functionality of the Cirrus + Logic Lochnagar audio development board. + config SND_SOC_MAX98088 tristate "Maxim MAX98088/9 Low-Power, Stereo Audio Codec" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index aab2ad95a137..a597de946027 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -91,6 +91,7 @@ snd-soc-jz4725b-codec-objs := jz4725b.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o +snd-soc-lochnagar-sc-objs := lochnagar-sc.o snd-soc-max9759-objs := max9759.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o @@ -364,6 +365,7 @@ obj-$(CONFIG_SND_SOC_JZ4725B_CODEC) += snd-soc-jz4725b-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o +obj-$(CONFIG_SND_SOC_LOCHNAGAR_SC) += snd-soc-lochnagar-sc.o obj-$(CONFIG_SND_SOC_MAX9759) += snd-soc-max9759.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o diff --git a/sound/soc/codecs/lochnagar-sc.c b/sound/soc/codecs/lochnagar-sc.c new file mode 100644 index 000000000000..3209b39e46af --- /dev/null +++ b/sound/soc/codecs/lochnagar-sc.c @@ -0,0 +1,266 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Lochnagar sound card driver +// +// Copyright (c) 2017-2019 Cirrus Logic, Inc. and +// Cirrus Logic International Semiconductor Ltd. +// +// Author: Charles Keepax +// Piotr Stankiewicz + +#include +#include +#include + +#include +#include +#include + +struct lochnagar_sc_priv { + struct clk *mclk; +}; + +static const struct snd_soc_dapm_widget lochnagar_sc_widgets[] = { + SND_SOC_DAPM_LINE("Line Jack", NULL), + SND_SOC_DAPM_LINE("USB Audio", NULL), +}; + +static const struct snd_soc_dapm_route lochnagar_sc_routes[] = { + { "Line Jack", NULL, "AIF1 Playback" }, + { "AIF1 Capture", NULL, "Line Jack" }, + + { "USB Audio", NULL, "USB1 Playback" }, + { "USB Audio", NULL, "USB2 Playback" }, + { "USB1 Capture", NULL, "USB Audio" }, + { "USB2 Capture", NULL, "USB Audio" }, +}; + +static const unsigned int lochnagar_sc_chan_vals[] = { + 4, 8, +}; + +static const struct snd_pcm_hw_constraint_list lochnagar_sc_chan_constraint = { + .count = ARRAY_SIZE(lochnagar_sc_chan_vals), + .list = lochnagar_sc_chan_vals, +}; + +static const unsigned int lochnagar_sc_rate_vals[] = { + 8000, 16000, 24000, 32000, 48000, 96000, 192000, + 22050, 44100, 88200, 176400, +}; + +static const struct snd_pcm_hw_constraint_list lochnagar_sc_rate_constraint = { + .count = ARRAY_SIZE(lochnagar_sc_rate_vals), + .list = lochnagar_sc_rate_vals, +}; + +static int lochnagar_sc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval range = { + .min = 8000, + .max = 24576000 / hw_param_interval(params, rule->deps[0])->max, + }; + + return snd_interval_refine(hw_param_interval(params, rule->var), + &range); +} + +static int lochnagar_sc_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *comp = dai->component; + struct lochnagar_sc_priv *priv = snd_soc_component_get_drvdata(comp); + int ret; + + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &lochnagar_sc_rate_constraint); + if (ret) + return ret; + + return snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + lochnagar_sc_hw_rule_rate, priv, + SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); +} + +static int lochnagar_sc_line_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *comp = dai->component; + struct lochnagar_sc_priv *priv = snd_soc_component_get_drvdata(comp); + int ret; + + ret = clk_prepare_enable(priv->mclk); + if (ret < 0) { + dev_err(dai->dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + + ret = lochnagar_sc_startup(substream, dai); + if (ret) + return ret; + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &lochnagar_sc_chan_constraint); +} + +static void lochnagar_sc_line_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *comp = dai->component; + struct lochnagar_sc_priv *priv = snd_soc_component_get_drvdata(comp); + + clk_disable_unprepare(priv->mclk); +} + +static int lochnagar_sc_check_fmt(struct snd_soc_dai *dai, unsigned int fmt, + unsigned int tar) +{ + tar |= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; + + if ((fmt & ~SND_SOC_DAIFMT_CLOCK_MASK) != tar) + return -EINVAL; + + return 0; +} + +static int lochnagar_sc_set_line_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + return lochnagar_sc_check_fmt(dai, fmt, SND_SOC_DAIFMT_CBS_CFS); +} + +static int lochnagar_sc_set_usb_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + return lochnagar_sc_check_fmt(dai, fmt, SND_SOC_DAIFMT_CBM_CFM); +} + +static const struct snd_soc_dai_ops lochnagar_sc_line_ops = { + .startup = lochnagar_sc_line_startup, + .shutdown = lochnagar_sc_line_shutdown, + .set_fmt = lochnagar_sc_set_line_fmt, +}; + +static const struct snd_soc_dai_ops lochnagar_sc_usb_ops = { + .startup = lochnagar_sc_startup, + .set_fmt = lochnagar_sc_set_usb_fmt, +}; + +static struct snd_soc_dai_driver lochnagar_sc_dai[] = { + { + .name = "lochnagar-line", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 4, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 4, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &lochnagar_sc_line_ops, + .symmetric_rates = true, + .symmetric_samplebits = true, + }, + { + .name = "lochnagar-usb1", + .playback = { + .stream_name = "USB1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "USB1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &lochnagar_sc_usb_ops, + .symmetric_rates = true, + .symmetric_samplebits = true, + }, + { + .name = "lochnagar-usb2", + .playback = { + .stream_name = "USB2 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "USB2 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &lochnagar_sc_usb_ops, + .symmetric_rates = true, + .symmetric_samplebits = true, + }, +}; + +static const struct snd_soc_component_driver lochnagar_sc_driver = { + .non_legacy_dai_naming = 1, + + .dapm_widgets = lochnagar_sc_widgets, + .num_dapm_widgets = ARRAY_SIZE(lochnagar_sc_widgets), + .dapm_routes = lochnagar_sc_routes, + .num_dapm_routes = ARRAY_SIZE(lochnagar_sc_routes), +}; + +static int lochnagar_sc_probe(struct platform_device *pdev) +{ + struct lochnagar_sc_priv *priv; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(priv->mclk)) { + ret = PTR_ERR(priv->mclk); + dev_err(&pdev->dev, "Failed to get MCLK: %d\n", ret); + return ret; + } + + platform_set_drvdata(pdev, priv); + + return devm_snd_soc_register_component(&pdev->dev, + &lochnagar_sc_driver, + lochnagar_sc_dai, + ARRAY_SIZE(lochnagar_sc_dai)); +} + +static const struct of_device_id lochnagar_of_match[] = { + { .compatible = "cirrus,lochnagar2-soundcard" }, + {} +}; +MODULE_DEVICE_TABLE(of, lochnagar_of_match); + +static struct platform_driver lochnagar_sc_codec_driver = { + .driver = { + .name = "lochnagar-soundcard", + .of_match_table = of_match_ptr(lochnagar_of_match), + }, + + .probe = lochnagar_sc_probe, +}; +module_platform_driver(lochnagar_sc_codec_driver); + +MODULE_DESCRIPTION("ASoC Lochnagar Sound Card Driver"); +MODULE_AUTHOR("Piotr Stankiewicz "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:lochnagar-soundcard"); -- cgit v1.2.3-59-g8ed1b From 008fe4e5382293e2da96a4dc75b95782a5f95be8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Mar 2019 13:57:02 +0900 Subject: ASoC: simple-card-utils: test memory allocation li->conf will be 0 if it was not DPCM case. Then, 1) we shouldn't call devm_kcalloc() with size 0, 2) we need NULL pointer check if li->conf was not 0. This patch fixed above issues. Special thanks to Pierre-Louis Bossart Reported-by: Pierre-Louis Bossart Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index a794f767ff7f..4ed68348f939 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -495,16 +495,21 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, struct snd_soc_dai_link *dai_link; struct simple_dai_props *dai_props; struct asoc_simple_dai *dais; - struct snd_soc_codec_conf *cconf; + struct snd_soc_codec_conf *cconf = NULL; int i; dai_props = devm_kcalloc(dev, li->link, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, li->link, sizeof(*dai_link), GFP_KERNEL); dais = devm_kcalloc(dev, li->dais, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, li->conf, sizeof(*cconf), GFP_KERNEL); if (!dai_props || !dai_link || !dais) return -ENOMEM; + if (li->conf) { + cconf = devm_kcalloc(dev, li->conf, sizeof(*cconf), GFP_KERNEL); + if (!cconf) + return -ENOMEM; + } + /* * Use snd_soc_dai_link_component instead of legacy style * It is codec only. but cpu/platform will be supported in the future. -- cgit v1.2.3-59-g8ed1b From 3a9ce0f1b2961f6c3ad2a49dcf85449784c18bb5 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Fri, 22 Mar 2019 12:25:35 +0800 Subject: ASoC: nau8810: fix the issue of 64 bits division Do division with div_u64 for the PLL calculation. These errors are fixed and list as follows: 1."__udivdi3" [sound/soc/codecs/snd-soc-nau8810.ko] undefined! 2."__aeabi_uldivmod" [sound/soc/codecs/snd-soc-nau8810.ko] undefined! 3. nau8810.c:(.text.nau8810_calc_pll+0xd8): undefined reference to `__udivdi3' Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8810.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 125e205e6687..dd82c65cfa7f 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -505,7 +505,8 @@ static int nau8810_calc_pll(unsigned int pll_in, f2_max = 0; scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { - f2 = 256ULL * fs * 4 * nau8810_mclk_scaler[i] / 10; + f2 = 256ULL * fs * 4 * nau8810_mclk_scaler[i]; + f2 = div_u64(f2, 10); if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && f2_max < f2) { f2_max = f2; -- cgit v1.2.3-59-g8ed1b From eddb6084305e5418ce28b8379a6c3d0058a61463 Mon Sep 17 00:00:00 2001 From: Andi Kleen Date: Thu, 21 Mar 2019 15:00:05 -0700 Subject: ASoC: AMD: Fix incorrect extern When using bare externs outside include files that types should at least match. This fixes a type confusion between bool and int. Cc: broonie@kernel.org Signed-off-by: Andi Kleen Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f37a588ba345..8a619a75b3a9 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -47,7 +47,7 @@ static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_clk; -extern int bt_uart_enable; +extern bool bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { -- cgit v1.2.3-59-g8ed1b From cf8817733d10c0147a01c97219dd0551bcf25e08 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Thu, 21 Mar 2019 16:34:56 +0100 Subject: ASoC: stm32: sai: add power management Add support of low power modes to STM32 SAI driver. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai.c | 80 ++++++++++++++++++++++++++++++++++++++----- sound/soc/stm/stm32_sai.h | 2 ++ sound/soc/stm/stm32_sai_sub.c | 45 ++++++++++++++++++------ 3 files changed, 108 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index d68d62f12df5..7550d5f08be3 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include @@ -44,20 +45,41 @@ static const struct of_device_id stm32_sai_ids[] = { {} }; -static int stm32_sai_sync_conf_client(struct stm32_sai_data *sai, int synci) +static int stm32_sai_pclk_disable(struct device *dev) +{ + struct stm32_sai_data *sai = dev_get_drvdata(dev); + + clk_disable_unprepare(sai->pclk); + + return 0; +} + +static int stm32_sai_pclk_enable(struct device *dev) { + struct stm32_sai_data *sai = dev_get_drvdata(dev); int ret; - /* Enable peripheral clock to allow GCR register access */ ret = clk_prepare_enable(sai->pclk); if (ret) { dev_err(&sai->pdev->dev, "failed to enable clock: %d\n", ret); return ret; } + return 0; +} + +static int stm32_sai_sync_conf_client(struct stm32_sai_data *sai, int synci) +{ + int ret; + + /* Enable peripheral clock to allow GCR register access */ + ret = stm32_sai_pclk_enable(&sai->pdev->dev); + if (ret) + return ret; + writel_relaxed(FIELD_PREP(SAI_GCR_SYNCIN_MASK, (synci - 1)), sai->base); - clk_disable_unprepare(sai->pclk); + stm32_sai_pclk_disable(&sai->pdev->dev); return 0; } @@ -68,11 +90,9 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) int ret; /* Enable peripheral clock to allow GCR register access */ - ret = clk_prepare_enable(sai->pclk); - if (ret) { - dev_err(&sai->pdev->dev, "failed to enable clock: %d\n", ret); + ret = stm32_sai_pclk_enable(&sai->pdev->dev); + if (ret) return ret; - } dev_dbg(&sai->pdev->dev, "Set %pOFn%s as synchro provider\n", sai->pdev->dev.of_node, @@ -83,13 +103,13 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); - clk_disable_unprepare(sai->pclk); + stm32_sai_pclk_disable(&sai->pdev->dev); return -EINVAL; } writel_relaxed(FIELD_PREP(SAI_GCR_SYNCOUT_MASK, synco), sai->base); - clk_disable_unprepare(sai->pclk); + stm32_sai_pclk_disable(&sai->pdev->dev); return 0; } @@ -195,12 +215,54 @@ static int stm32_sai_probe(struct platform_device *pdev) return devm_of_platform_populate(&pdev->dev); } +#ifdef CONFIG_PM_SLEEP +/* + * When pins are shared by two sai sub instances, pins have to be defined + * in sai parent node. In this case, pins state is not managed by alsa fw. + * These pins are managed in suspend/resume callbacks. + */ +static int stm32_sai_suspend(struct device *dev) +{ + struct stm32_sai_data *sai = dev_get_drvdata(dev); + int ret; + + ret = stm32_sai_pclk_enable(dev); + if (ret) + return ret; + + sai->gcr = readl_relaxed(sai->base); + stm32_sai_pclk_disable(dev); + + return pinctrl_pm_select_sleep_state(dev); +} + +static int stm32_sai_resume(struct device *dev) +{ + struct stm32_sai_data *sai = dev_get_drvdata(dev); + int ret; + + ret = stm32_sai_pclk_enable(dev); + if (ret) + return ret; + + writel_relaxed(sai->gcr, sai->base); + stm32_sai_pclk_disable(dev); + + return pinctrl_pm_select_default_state(dev); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops stm32_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(stm32_sai_suspend, stm32_sai_resume) +}; + MODULE_DEVICE_TABLE(of, stm32_sai_ids); static struct platform_driver stm32_sai_driver = { .driver = { .name = "st,stm32-sai", .of_match_table = stm32_sai_ids, + .pm = &stm32_sai_pm_ops, }, .probe = stm32_sai_probe, }; diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h index 08de899c766b..9c36a393ab7b 100644 --- a/sound/soc/stm/stm32_sai.h +++ b/sound/soc/stm/stm32_sai.h @@ -268,6 +268,7 @@ struct stm32_sai_conf { * @version: SOC version * @irq: SAI interrupt line * @set_sync: pointer to synchro mode configuration callback + * @gcr: SAI Global Configuration Register */ struct stm32_sai_data { struct platform_device *pdev; @@ -279,4 +280,5 @@ struct stm32_sai_data { int irq; int (*set_sync)(struct stm32_sai_data *sai, struct device_node *np_provider, int synco, int synci); + u32 gcr; }; diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 55d802f51c15..2a90641cf164 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -168,6 +168,7 @@ static bool stm32_sai_sub_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case STM_SAI_DR_REGX: + case STM_SAI_SR_REGX: return true; default: return false; @@ -182,7 +183,6 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg) case STM_SAI_FRCR_REGX: case STM_SAI_SLOTR_REGX: case STM_SAI_IMR_REGX: - case STM_SAI_SR_REGX: case STM_SAI_CLRFR_REGX: case STM_SAI_DR_REGX: case STM_SAI_PDMCR_REGX: @@ -202,6 +202,7 @@ static const struct regmap_config stm32_sai_sub_regmap_config_f4 = { .volatile_reg = stm32_sai_sub_volatile_reg, .writeable_reg = stm32_sai_sub_writeable_reg, .fast_io = true, + .cache_type = REGCACHE_FLAT, }; static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { @@ -213,6 +214,7 @@ static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { .volatile_reg = stm32_sai_sub_volatile_reg, .writeable_reg = stm32_sai_sub_writeable_reg, .fast_io = true, + .cache_type = REGCACHE_FLAT, }; static int snd_pcm_iec958_info(struct snd_kcontrol *kcontrol, @@ -441,8 +443,8 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid) if (!flags) return IRQ_NONE; - regmap_update_bits(sai->regmap, STM_SAI_CLRFR_REGX, SAI_XCLRFR_MASK, - SAI_XCLRFR_MASK); + regmap_write_bits(sai->regmap, STM_SAI_CLRFR_REGX, SAI_XCLRFR_MASK, + SAI_XCLRFR_MASK); if (!sai->substream) { dev_err(&pdev->dev, "Device stopped. Spurious IRQ 0x%x\n", sr); @@ -704,9 +706,8 @@ static int stm32_sai_startup(struct snd_pcm_substream *substream, } /* Enable ITs */ - - regmap_update_bits(sai->regmap, STM_SAI_CLRFR_REGX, - SAI_XCLRFR_MASK, SAI_XCLRFR_MASK); + regmap_write_bits(sai->regmap, STM_SAI_CLRFR_REGX, + SAI_XCLRFR_MASK, SAI_XCLRFR_MASK); imr = SAI_XIMR_OVRUDRIE; if (STM_SAI_IS_CAPTURE(sai)) { @@ -738,10 +739,10 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, * SAI fifo threshold is set to half fifo, to keep enough space * for DMA incoming bursts. */ - regmap_update_bits(sai->regmap, STM_SAI_CR2_REGX, - SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, - SAI_XCR2_FFLUSH | - SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); + regmap_write_bits(sai->regmap, STM_SAI_CR2_REGX, + SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, + SAI_XCR2_FFLUSH | + SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); /* DS bits in CR1 not set for SPDIF (size forced to 24 bits).*/ if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { @@ -1492,10 +1493,34 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int stm32_sai_sub_suspend(struct device *dev) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + return 0; +} + +static int stm32_sai_sub_resume(struct device *dev) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops stm32_sai_sub_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(stm32_sai_sub_suspend, stm32_sai_sub_resume) +}; + static struct platform_driver stm32_sai_sub_driver = { .driver = { .name = "st,stm32-sai-sub", .of_match_table = stm32_sai_sub_ids, + .pm = &stm32_sai_sub_pm_ops, }, .probe = stm32_sai_sub_probe, }; -- cgit v1.2.3-59-g8ed1b From 78540a259b050480d20354efaf1055c233728b3b Mon Sep 17 00:00:00 2001 From: Aditya Pakki Date: Fri, 22 Mar 2019 17:00:29 -0500 Subject: ASoC: sirf-audio: Remove redundant of_match_node call Unlike other drivers probe method, of_match_node return value is not used or checked. This patch removes the redundant code. Signed-off-by: Aditya Pakki Reviewed-by: Steven Price Signed-off-by: Mark Brown --- sound/soc/codecs/sirf-audio-codec.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index e424499a8450..e0af21050078 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -461,9 +461,6 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) struct sirf_audio_codec *sirf_audio_codec; void __iomem *base; struct resource *mem_res; - const struct of_device_id *match; - - match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node); sirf_audio_codec = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_codec), GFP_KERNEL); -- cgit v1.2.3-59-g8ed1b From 50e0908412e6937ebf38b1edae35035a550a6f09 Mon Sep 17 00:00:00 2001 From: Timo Wischer Date: Mon, 25 Mar 2019 16:14:14 +0100 Subject: ALSA: aloop: Support S24 sample formats Currently snd_aloop supports only S16 and S32 audio sample formats. With this patch the S24 formats are also supported. Signed-off-by: Timo Wischer Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 8c3fbe1276be..c14e57b2a135 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -337,7 +337,7 @@ static int loopback_prepare(struct snd_pcm_substream *substream) loopback_timer_stop_sync(dpcm); - salign = (snd_pcm_format_width(runtime->format) * + salign = (snd_pcm_format_physical_width(runtime->format) * runtime->channels) / 8; bps = salign * runtime->rate; if (bps <= 0 || salign <= 0) @@ -562,6 +562,8 @@ static const struct snd_pcm_hardware loopback_pcm_hardware = SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE), .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000, -- cgit v1.2.3-59-g8ed1b From 8633d44002ba5c98f44bacc1397190adba832fd6 Mon Sep 17 00:00:00 2001 From: Pankaj Bharadiya Date: Fri, 22 Mar 2019 21:53:39 +0530 Subject: ASoC: dapm: Initialize private_value in snd_soc_dapm_new_dai In case of single config, private_value is left uninitialized. The private_value does need to be initialized or in snd_soc_dapm_new_control_unlocked() call failure case, it leads to a bogus free in snd_soc_dapm_free_kcontrol() Signed-off-by: Pankaj Bharadiya Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ec06ef6d161..74a6f5705f66 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4038,7 +4038,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, struct snd_soc_dapm_widget template; struct snd_soc_dapm_widget *w; const char **w_param_text; - unsigned long private_value; + unsigned long private_value = 0; char *link_name; int ret; -- cgit v1.2.3-59-g8ed1b From 514b044cba667e4b7c383ec79b42b997e624b91d Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Thu, 21 Mar 2019 17:58:45 -0700 Subject: ASoC: tlv320aic32x4: Model PLL in CCF Model and manage the on-board PLL as a component in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/tlv320aic32x4-clk.c | 323 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 195 +++++++++++---------- sound/soc/codecs/tlv320aic32x4.h | 5 + 5 files changed, 431 insertions(+), 95 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic32x4-clk.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 05f16632296b..6e99320c79b8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1105,6 +1105,7 @@ config SND_SOC_TLV320AIC31XX config SND_SOC_TLV320AIC32X4 tristate + depends on COMMON_CLK config SND_SOC_TLV320AIC32X4_I2C tristate "Texas Instruments TLV320AIC32x4 audio CODECs - I2C" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a597de946027..aa7720a7a0aa 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -193,7 +193,7 @@ snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o -snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o +snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o tlv320aic32x4-clk.o snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c new file mode 100644 index 000000000000..5e495fc8d931 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -0,0 +1,323 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * Clock Tree for the Texas Instruments TLV320AIC32x4 + * + * Copyright 2019 Annaliese McDermond + * + * Author: Annaliese McDermond + */ + +#include +#include +#include +#include + +#include "tlv320aic32x4.h" + +#define to_clk_aic32x4(_hw) container_of(_hw, struct clk_aic32x4, hw) +struct clk_aic32x4 { + struct clk_hw hw; + struct device *dev; + struct regmap *regmap; + unsigned int reg; +}; + +/* + * struct clk_aic32x4_pll_muldiv - Multiplier/divider settings + * @p: Divider + * @r: first multiplier + * @j: integer part of second multiplier + * @d: decimal part of second multiplier + */ +struct clk_aic32x4_pll_muldiv { + u8 p; + u16 r; + u8 j; + u16 d; +}; + +struct aic32x4_clkdesc { + const char *name; + const char * const *parent_names; + unsigned int num_parents; + const struct clk_ops *ops; + unsigned int reg; +}; + +static int clk_aic32x4_pll_prepare(struct clk_hw *hw) +{ + struct clk_aic32x4 *pll = to_clk_aic32x4(hw); + + return regmap_update_bits(pll->regmap, AIC32X4_PLLPR, + AIC32X4_PLLEN, AIC32X4_PLLEN); +} + +static void clk_aic32x4_pll_unprepare(struct clk_hw *hw) +{ + struct clk_aic32x4 *pll = to_clk_aic32x4(hw); + + regmap_update_bits(pll->regmap, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); +} + +static int clk_aic32x4_pll_is_prepared(struct clk_hw *hw) +{ + struct clk_aic32x4 *pll = to_clk_aic32x4(hw); + + unsigned int val; + int ret; + + ret = regmap_read(pll->regmap, AIC32X4_PLLPR, &val); + if (ret < 0) + return ret; + + return !!(val & AIC32X4_PLLEN); +} + +static int clk_aic32x4_pll_get_muldiv(struct clk_aic32x4 *pll, + struct clk_aic32x4_pll_muldiv *settings) +{ + /* Change to use regmap_bulk_read? */ + unsigned int val; + int ret; + + ret = regmap_read(pll->regmap, AIC32X4_PLLPR, &val); + if (ret) + return ret; + settings->r = val & AIC32X4_PLL_R_MASK; + settings->p = (val & AIC32X4_PLL_P_MASK) >> AIC32X4_PLL_P_SHIFT; + + ret = regmap_read(pll->regmap, AIC32X4_PLLJ, &val); + if (ret < 0) + return ret; + settings->j = val; + + ret = regmap_read(pll->regmap, AIC32X4_PLLDMSB, &val); + if (ret < 0) + return ret; + settings->d = val << 8; + + ret = regmap_read(pll->regmap, AIC32X4_PLLDLSB, &val); + if (ret < 0) + return ret; + settings->d |= val; + + return 0; +} + +static int clk_aic32x4_pll_set_muldiv(struct clk_aic32x4 *pll, + struct clk_aic32x4_pll_muldiv *settings) +{ + int ret; + /* Change to use regmap_bulk_write for some if not all? */ + + ret = regmap_update_bits(pll->regmap, AIC32X4_PLLPR, + AIC32X4_PLL_R_MASK, settings->r); + if (ret < 0) + return ret; + + ret = regmap_update_bits(pll->regmap, AIC32X4_PLLPR, + AIC32X4_PLL_P_MASK, + settings->p << AIC32X4_PLL_P_SHIFT); + if (ret < 0) + return ret; + + ret = regmap_write(pll->regmap, AIC32X4_PLLJ, settings->j); + if (ret < 0) + return ret; + + ret = regmap_write(pll->regmap, AIC32X4_PLLDMSB, (settings->d >> 8)); + if (ret < 0) + return ret; + ret = regmap_write(pll->regmap, AIC32X4_PLLDLSB, (settings->d & 0xff)); + if (ret < 0) + return ret; + + return 0; +} + +static unsigned long clk_aic32x4_pll_calc_rate( + struct clk_aic32x4_pll_muldiv *settings, + unsigned long parent_rate) +{ + u64 rate; + /* + * We scale j by 10000 to account for the decimal part of P and divide + * it back out later. + */ + rate = (u64) parent_rate * settings->r * + ((settings->j * 10000) + settings->d); + + return (unsigned long) DIV_ROUND_UP_ULL(rate, settings->p * 10000); +} + +static int clk_aic32x4_pll_calc_muldiv(struct clk_aic32x4_pll_muldiv *settings, + unsigned long rate, unsigned long parent_rate) +{ + u64 multiplier; + + settings->p = parent_rate / AIC32X4_MAX_PLL_CLKIN + 1; + if (settings->p > 8) + return -1; + + /* + * We scale this figure by 10000 so that we can get the decimal part + * of the multiplier. This is because we can't do floating point + * math in the kernel. + */ + multiplier = (u64) rate * settings->p * 10000; + do_div(multiplier, parent_rate); + + /* + * J can't be over 64, so R can scale this. + * R can't be greater than 4. + */ + settings->r = ((u32) multiplier / 640000) + 1; + if (settings->r > 4) + return -1; + do_div(multiplier, settings->r); + + /* + * J can't be < 1. + */ + if (multiplier < 10000) + return -1; + + /* Figure out the integer part, J, and the fractional part, D. */ + settings->j = (u32) multiplier / 10000; + settings->d = (u32) multiplier % 10000; + + return 0; +} + +static unsigned long clk_aic32x4_pll_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct clk_aic32x4 *pll = to_clk_aic32x4(hw); + struct clk_aic32x4_pll_muldiv settings; + int ret; + + ret = clk_aic32x4_pll_get_muldiv(pll, &settings); + if (ret < 0) + return 0; + + return clk_aic32x4_pll_calc_rate(&settings, parent_rate); +} + +static long clk_aic32x4_pll_round_rate(struct clk_hw *hw, + unsigned long rate, + unsigned long *parent_rate) +{ + struct clk_aic32x4_pll_muldiv settings; + int ret; + + ret = clk_aic32x4_pll_calc_muldiv(&settings, rate, *parent_rate); + if (ret < 0) + return 0; + + return clk_aic32x4_pll_calc_rate(&settings, *parent_rate); +} + +static int clk_aic32x4_pll_set_rate(struct clk_hw *hw, + unsigned long rate, + unsigned long parent_rate) +{ + struct clk_aic32x4 *pll = to_clk_aic32x4(hw); + struct clk_aic32x4_pll_muldiv settings; + int ret; + + ret = clk_aic32x4_pll_calc_muldiv(&settings, rate, parent_rate); + if (ret < 0) + return -EINVAL; + + return clk_aic32x4_pll_set_muldiv(pll, &settings); +} + +static int clk_aic32x4_pll_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *pll = to_clk_aic32x4(hw); + + return regmap_update_bits(pll->regmap, + AIC32X4_CLKMUX, + AIC32X4_PLL_CLKIN_MASK, + index << AIC32X4_PLL_CLKIN_SHIFT); +} + +static u8 clk_aic32x4_pll_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *pll = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(pll->regmap, AIC32X4_PLLPR, &val); + + return (val & AIC32X4_PLL_CLKIN_MASK) >> AIC32X4_PLL_CLKIN_SHIFT; +} + + +static const struct clk_ops aic32x4_pll_ops = { + .prepare = clk_aic32x4_pll_prepare, + .unprepare = clk_aic32x4_pll_unprepare, + .is_prepared = clk_aic32x4_pll_is_prepared, + .recalc_rate = clk_aic32x4_pll_recalc_rate, + .round_rate = clk_aic32x4_pll_round_rate, + .set_rate = clk_aic32x4_pll_set_rate, + .set_parent = clk_aic32x4_pll_set_parent, + .get_parent = clk_aic32x4_pll_get_parent, +}; + +static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { + { + .name = "pll", + .parent_names = + (const char* []) { "mclk", "bclk", "gpio", "din" }, + .num_parents = 4, + .ops = &aic32x4_pll_ops, + .reg = 0, + }, +}; + +static struct clk *aic32x4_register_clk(struct device *dev, + struct aic32x4_clkdesc *desc) +{ + struct clk_init_data init; + struct clk_aic32x4 *priv; + const char *devname = dev_name(dev); + + init.ops = desc->ops; + init.name = desc->name; + init.parent_names = desc->parent_names; + init.num_parents = desc->num_parents; + init.flags = 0; + + priv = devm_kzalloc(dev, sizeof(struct clk_aic32x4), GFP_KERNEL); + if (priv == NULL) + return (struct clk *) -ENOMEM; + + priv->dev = dev; + priv->hw.init = &init; + priv->regmap = dev_get_regmap(dev, NULL); + priv->reg = desc->reg; + + clk_hw_register_clkdev(&priv->hw, desc->name, devname); + return devm_clk_register(dev, &priv->hw); +} + +int aic32x4_register_clocks(struct device *dev, const char *mclk_name) +{ + int i; + + /* + * These lines are here to preserve the current functionality of + * the driver with regard to the DT. These should eventually be set + * by DT nodes so that the connections can be set up in configuration + * rather than code. + */ + aic32x4_clkdesc_array[0].parent_names = + (const char* []) { mclk_name, "bclk", "gpio", "din" }; + + for (i = 0; i < ARRAY_SIZE(aic32x4_clkdesc_array); ++i) + aic32x4_register_clk(dev, &aic32x4_clkdesc_array[i]); + + return 0; +} +EXPORT_SYMBOL_GPL(aic32x4_register_clocks); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 71a93fbc5971..7cf8c7cedfe1 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -14,7 +14,7 @@ * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License @@ -33,6 +33,7 @@ #include #include #include +#include #include #include @@ -49,9 +50,7 @@ struct aic32x4_rate_divs { u32 mclk; u32 rate; - u8 p_val; - u8 pll_j; - u16 pll_d; + unsigned long pll_rate; u16 dosr; u8 ndac; u8 mdac; @@ -71,6 +70,7 @@ struct aic32x4_priv { bool swapdacs; int rstn_gpio; struct clk *mclk; + const char *mclk_name; struct regulator *supply_ldo; struct regulator *supply_iov; @@ -309,34 +309,34 @@ static const struct snd_kcontrol_new aic32x4_snd_controls[] = { static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 8k rate */ - {12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24, 1, 1}, - {24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24, 1, 1}, - {25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24, 1, 1}, + { 12000000, 8000, 57120000, 768, 5, 3, 128, 5, 18, 24, 1, 1 }, + { 24000000, 8000, 57120000, 768, 15, 1, 64, 45, 4, 24, 1, 1 }, + { 25000000, 8000, 32620000, 768, 15, 1, 64, 45, 4, 24, 1, 1 }, /* 11.025k rate */ - {12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16, 1, 1}, - {24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16, 1, 1}, + { 12000000, 11025, 44217600, 512, 8, 2, 128, 8, 8, 16, 1, 1 }, + { 24000000, 11025, 44217600, 512, 16, 1, 64, 32, 4, 16, 1, 1 }, /* 16k rate */ - {12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12, 1, 1}, - {24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12, 1, 1}, - {25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12, 1, 1}, + { 12000000, 16000, 57120000, 384, 5, 3, 128, 5, 9, 12, 1, 1 }, + { 24000000, 16000, 57120000, 384, 15, 1, 64, 18, 5, 12, 1, 1 }, + { 25000000, 16000, 32620000, 384, 15, 1, 64, 18, 5, 12, 1, 1 }, /* 22.05k rate */ - {12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8, 1, 1}, - {24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8, 1, 1}, - {25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8, 1, 1}, + { 12000000, 22050, 44217600, 256, 4, 4, 128, 4, 8, 8, 1, 1 }, + { 24000000, 22050, 44217600, 256, 16, 1, 64, 16, 4, 8, 1, 1 }, + { 25000000, 22050, 19713750, 256, 16, 1, 64, 16, 4, 8, 1, 1 }, /* 32k rate */ - {12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6, 1, 1}, - {24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6, 1, 1}, + { 12000000, 32000, 14112000, 192, 2, 7, 64, 2, 21, 6, 1, 1 }, + { 24000000, 32000, 14112000, 192, 7, 2, 64, 7, 6, 6, 1, 1 }, /* 44.1k rate */ - {12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4, 1, 1}, - {24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4, 1, 1}, - {25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4, 1, 1}, + { 12000000, 44100, 44217600, 128, 2, 8, 128, 2, 8, 4, 1, 1 }, + { 24000000, 44100, 44217600, 128, 8, 2, 64, 8, 4, 4, 1, 1 }, + { 25000000, 44100, 19713750, 128, 8, 2, 64, 8, 4, 4, 1, 1 }, /* 48k rate */ - {12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4, 1, 1}, - {24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4, 1, 1}, - {25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4, 1, 1}, + { 12000000, 48000, 18432000, 128, 2, 8, 128, 2, 8, 4, 1, 1 }, + { 24000000, 48000, 18432000, 128, 8, 2, 64, 8, 4, 4, 1, 1 }, + { 25000000, 48000, 75626250, 128, 8, 2, 64, 8, 4, 4, 1, 1 }, /* 96k rate */ - {25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1, 1, 9}, + { 25000000, 96000, 75626250, 64, 4, 4, 64, 4, 4, 1, 1, 9 }, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -393,7 +393,7 @@ static const struct snd_kcontrol_new in3r_to_lmixer_controls[] = { SOC_DAPM_ENUM("IN3_R L- Switch", in3r_lpga_n_enum), }; -/* Right mixer pins */ +/* Right mixer pins */ static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, resistor_text); static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, resistor_text); static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, resistor_text); @@ -595,7 +595,7 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { static const struct regmap_range_cfg aic32x4_regmap_pages[] = { { .selector_reg = 0, - .selector_mask = 0xff, + .selector_mask = 0xff, .window_start = 0, .window_len = 128, .range_min = 0, @@ -616,7 +616,7 @@ static inline int aic32x4_get_divs(int mclk, int rate) for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) { if ((aic32x4_divs[i].rate == rate) - && (aic32x4_divs[i].mclk == mclk)) { + && (aic32x4_divs[i].mclk == mclk)) { return i; } } @@ -688,12 +688,12 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } snd_soc_component_update_bits(component, AIC32X4_IFACE1, - AIC32X4_IFACE1_DATATYPE_MASK | - AIC32X4_IFACE1_MASTER_MASK, iface_reg_1); + AIC32X4_IFACE1_DATATYPE_MASK | + AIC32X4_IFACE1_MASTER_MASK, iface_reg_1); snd_soc_component_update_bits(component, AIC32X4_IFACE2, - AIC32X4_DATA_OFFSET_MASK, iface_reg_2); + AIC32X4_DATA_OFFSET_MASK, iface_reg_2); snd_soc_component_update_bits(component, AIC32X4_IFACE3, - AIC32X4_BCLKINV_MASK, iface_reg_3); + AIC32X4_BCLKINV_MASK, iface_reg_3); return 0; } @@ -715,6 +715,11 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, unsigned int parent_rate) { int i; + int ret; + + struct clk_bulk_data clocks[] = { + { .id = "pll" }, + }; i = aic32x4_get_divs(parent_rate, sample_rate); if (i < 0) { @@ -722,39 +727,29 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, return i; } + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; + + clk_set_rate(clocks[0].clk, sample_rate); + aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* MCLK as PLL_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, AIC32X4_PLL_CLKIN_MASK, - AIC32X4_PLL_CLKIN_MCLK << AIC32X4_PLL_CLKIN_SHIFT); /* PLL as CODEC_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, AIC32X4_CODEC_CLKIN_MASK, - AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); + snd_soc_component_update_bits(component, AIC32X4_CLKMUX, + AIC32X4_CODEC_CLKIN_MASK, + AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); /* DAC_MOD_CLK as BDIV_CLKIN */ snd_soc_component_update_bits(component, AIC32X4_IFACE3, AIC32X4_BDIVCLK_MASK, - AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); - - /* We will fix R value to 1 and will make P & J=K.D as variable */ - snd_soc_component_update_bits(component, AIC32X4_PLLPR, AIC32X4_PLL_R_MASK, 0x01); - - /* PLL P value */ - snd_soc_component_update_bits(component, AIC32X4_PLLPR, AIC32X4_PLL_P_MASK, - aic32x4_divs[i].p_val << AIC32X4_PLL_P_SHIFT); - - /* PLL J value */ - snd_soc_component_write(component, AIC32X4_PLLJ, aic32x4_divs[i].pll_j); - - /* PLL D value */ - snd_soc_component_write(component, AIC32X4_PLLDMSB, (aic32x4_divs[i].pll_d >> 8)); - snd_soc_component_write(component, AIC32X4_PLLDLSB, (aic32x4_divs[i].pll_d & 0xff)); + AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); /* NDAC divider value */ snd_soc_component_update_bits(component, AIC32X4_NDAC, - AIC32X4_NDAC_MASK, aic32x4_divs[i].ndac); + AIC32X4_NDAC_MASK, aic32x4_divs[i].ndac); /* MDAC divider value */ snd_soc_component_update_bits(component, AIC32X4_MDAC, - AIC32X4_MDAC_MASK, aic32x4_divs[i].mdac); + AIC32X4_MDAC_MASK, aic32x4_divs[i].mdac); /* DOSR MSB & LSB values */ snd_soc_component_write(component, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); @@ -762,18 +757,18 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, /* NADC divider value */ snd_soc_component_update_bits(component, AIC32X4_NADC, - AIC32X4_NADC_MASK, aic32x4_divs[i].nadc); + AIC32X4_NADC_MASK, aic32x4_divs[i].nadc); /* MADC divider value */ snd_soc_component_update_bits(component, AIC32X4_MADC, - AIC32X4_MADC_MASK, aic32x4_divs[i].madc); + AIC32X4_MADC_MASK, aic32x4_divs[i].madc); /* AOSR value */ snd_soc_component_write(component, AIC32X4_AOSR, aic32x4_divs[i].aosr); /* BCLK N divider */ snd_soc_component_update_bits(component, AIC32X4_BCLKN, - AIC32X4_BCLK_MASK, aic32x4_divs[i].blck_N); + AIC32X4_BCLK_MASK, aic32x4_divs[i].blck_N); return 0; } @@ -792,23 +787,23 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, switch (params_width(params)) { case 16: iface1_reg |= (AIC32X4_WORD_LEN_16BITS << - AIC32X4_IFACE1_DATALEN_SHIFT); + AIC32X4_IFACE1_DATALEN_SHIFT); break; case 20: iface1_reg |= (AIC32X4_WORD_LEN_20BITS << - AIC32X4_IFACE1_DATALEN_SHIFT); + AIC32X4_IFACE1_DATALEN_SHIFT); break; case 24: iface1_reg |= (AIC32X4_WORD_LEN_24BITS << - AIC32X4_IFACE1_DATALEN_SHIFT); + AIC32X4_IFACE1_DATALEN_SHIFT); break; case 32: iface1_reg |= (AIC32X4_WORD_LEN_32BITS << - AIC32X4_IFACE1_DATALEN_SHIFT); + AIC32X4_IFACE1_DATALEN_SHIFT); break; } snd_soc_component_update_bits(component, AIC32X4_IFACE1, - AIC32X4_IFACE1_DATALEN_MASK, iface1_reg); + AIC32X4_IFACE1_DATALEN_MASK, iface1_reg); if (params_channels(params) == 1) { dacsetup_reg = AIC32X4_RDAC2LCHN | AIC32X4_LDAC2LCHN; @@ -819,7 +814,7 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, dacsetup_reg = AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN; } snd_soc_component_update_bits(component, AIC32X4_DACSETUP, - AIC32X4_DAC_CHAN_MASK, dacsetup_reg); + AIC32X4_DAC_CHAN_MASK, dacsetup_reg); return 0; } @@ -829,7 +824,7 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_component *component = dai->component; snd_soc_component_update_bits(component, AIC32X4_DACMUTE, - AIC32X4_MUTEON, mute ? AIC32X4_MUTEON : 0); + AIC32X4_MUTEON, mute ? AIC32X4_MUTEON : 0); return 0; } @@ -851,27 +846,27 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, /* Switch on PLL */ snd_soc_component_update_bits(component, AIC32X4_PLLPR, - AIC32X4_PLLEN, AIC32X4_PLLEN); + AIC32X4_PLLEN, AIC32X4_PLLEN); /* Switch on NDAC Divider */ snd_soc_component_update_bits(component, AIC32X4_NDAC, - AIC32X4_NDACEN, AIC32X4_NDACEN); + AIC32X4_NDACEN, AIC32X4_NDACEN); /* Switch on MDAC Divider */ snd_soc_component_update_bits(component, AIC32X4_MDAC, - AIC32X4_MDACEN, AIC32X4_MDACEN); + AIC32X4_MDACEN, AIC32X4_MDACEN); /* Switch on NADC Divider */ snd_soc_component_update_bits(component, AIC32X4_NADC, - AIC32X4_NADCEN, AIC32X4_NADCEN); + AIC32X4_NADCEN, AIC32X4_NADCEN); /* Switch on MADC Divider */ snd_soc_component_update_bits(component, AIC32X4_MADC, - AIC32X4_MADCEN, AIC32X4_MADCEN); + AIC32X4_MADCEN, AIC32X4_MADCEN); /* Switch on BCLK_N Divider */ snd_soc_component_update_bits(component, AIC32X4_BCLKN, - AIC32X4_BCLKEN, AIC32X4_BCLKEN); + AIC32X4_BCLKEN, AIC32X4_BCLKEN); break; case SND_SOC_BIAS_PREPARE: break; @@ -882,27 +877,27 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, /* Switch off BCLK_N Divider */ snd_soc_component_update_bits(component, AIC32X4_BCLKN, - AIC32X4_BCLKEN, 0); + AIC32X4_BCLKEN, 0); /* Switch off MADC Divider */ snd_soc_component_update_bits(component, AIC32X4_MADC, - AIC32X4_MADCEN, 0); + AIC32X4_MADCEN, 0); /* Switch off NADC Divider */ snd_soc_component_update_bits(component, AIC32X4_NADC, - AIC32X4_NADCEN, 0); + AIC32X4_NADCEN, 0); /* Switch off MDAC Divider */ snd_soc_component_update_bits(component, AIC32X4_MDAC, - AIC32X4_MDACEN, 0); + AIC32X4_MDACEN, 0); /* Switch off NDAC Divider */ snd_soc_component_update_bits(component, AIC32X4_NDAC, - AIC32X4_NDACEN, 0); + AIC32X4_NDACEN, 0); /* Switch off PLL */ snd_soc_component_update_bits(component, AIC32X4_PLLPR, - AIC32X4_PLLEN, 0); + AIC32X4_PLLEN, 0); /* Switch off master clock */ clk_disable_unprepare(aic32x4->mclk); @@ -914,7 +909,7 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, } #define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000 -#define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ +#define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops aic32x4_ops = { @@ -927,17 +922,17 @@ static const struct snd_soc_dai_ops aic32x4_ops = { static struct snd_soc_dai_driver aic32x4_dai = { .name = "tlv320aic32x4-hifi", .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = AIC32X4_RATES, - .formats = AIC32X4_FORMATS,}, + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = AIC32X4_RATES, - .formats = AIC32X4_FORMATS,}, + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, .ops = &aic32x4_ops, .symmetric_rates = 1, }; @@ -950,7 +945,7 @@ static void aic32x4_setup_gpios(struct snd_soc_component *component) /* MFP1 */ if (aic32x4->setup->gpio_func[0] != AIC32X4_MFPX_DEFAULT_VALUE) { snd_soc_component_write(component, AIC32X4_DINCTL, - aic32x4->setup->gpio_func[0]); + aic32x4->setup->gpio_func[0]); snd_soc_add_component_controls(component, aic32x4_mfp1, ARRAY_SIZE(aic32x4_mfp1)); } @@ -958,7 +953,7 @@ static void aic32x4_setup_gpios(struct snd_soc_component *component) /* MFP2 */ if (aic32x4->setup->gpio_func[1] != AIC32X4_MFPX_DEFAULT_VALUE) { snd_soc_component_write(component, AIC32X4_DOUTCTL, - aic32x4->setup->gpio_func[1]); + aic32x4->setup->gpio_func[1]); snd_soc_add_component_controls(component, aic32x4_mfp2, ARRAY_SIZE(aic32x4_mfp2)); } @@ -966,7 +961,7 @@ static void aic32x4_setup_gpios(struct snd_soc_component *component) /* MFP3 */ if (aic32x4->setup->gpio_func[2] != AIC32X4_MFPX_DEFAULT_VALUE) { snd_soc_component_write(component, AIC32X4_SCLKCTL, - aic32x4->setup->gpio_func[2]); + aic32x4->setup->gpio_func[2]); snd_soc_add_component_controls(component, aic32x4_mfp3, ARRAY_SIZE(aic32x4_mfp3)); } @@ -974,7 +969,7 @@ static void aic32x4_setup_gpios(struct snd_soc_component *component) /* MFP4 */ if (aic32x4->setup->gpio_func[3] != AIC32X4_MFPX_DEFAULT_VALUE) { snd_soc_component_write(component, AIC32X4_MISOCTL, - aic32x4->setup->gpio_func[3]); + aic32x4->setup->gpio_func[3]); snd_soc_add_component_controls(component, aic32x4_mfp4, ARRAY_SIZE(aic32x4_mfp4)); } @@ -982,7 +977,7 @@ static void aic32x4_setup_gpios(struct snd_soc_component *component) /* MFP5 */ if (aic32x4->setup->gpio_func[4] != AIC32X4_MFPX_DEFAULT_VALUE) { snd_soc_component_write(component, AIC32X4_GPIOCTL, - aic32x4->setup->gpio_func[4]); + aic32x4->setup->gpio_func[4]); snd_soc_add_component_controls(component, aic32x4_mfp5, ARRAY_SIZE(aic32x4_mfp5)); } @@ -1006,8 +1001,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { - snd_soc_component_write(component, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | - AIC32X4_MICBIAS_2075V); + snd_soc_component_write(component, AIC32X4_MICBIAS, + AIC32X4_MICBIAS_LDOIN | AIC32X4_MICBIAS_2075V); } if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) snd_soc_component_write(component, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); @@ -1070,12 +1065,18 @@ static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4, struct device_node *np) { struct aic32x4_setup_data *aic32x4_setup; + int ret; aic32x4_setup = devm_kzalloc(aic32x4->dev, sizeof(*aic32x4_setup), GFP_KERNEL); if (!aic32x4_setup) return -ENOMEM; + ret = of_property_match_string(np, "clock-names", "mclk"); + if (ret < 0) + return -EINVAL; + aic32x4->mclk_name = of_clk_get_parent_name(np, ret); + aic32x4->swapdacs = false; aic32x4->micpga_routing = 0; aic32x4->rstn_gpio = of_get_named_gpio(np, "reset-gpios", 0); @@ -1197,7 +1198,7 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) return PTR_ERR(regmap); aic32x4 = devm_kzalloc(dev, sizeof(struct aic32x4_priv), - GFP_KERNEL); + GFP_KERNEL); if (aic32x4 == NULL) return -ENOMEM; @@ -1209,6 +1210,7 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) aic32x4->swapdacs = pdata->swapdacs; aic32x4->micpga_routing = pdata->micpga_routing; aic32x4->rstn_gpio = pdata->rstn_gpio; + aic32x4->mclk_name = "mclk"; } else if (np) { ret = aic32x4_parse_dt(aic32x4, np); if (ret) { @@ -1220,6 +1222,7 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) aic32x4->swapdacs = false; aic32x4->micpga_routing = 0; aic32x4->rstn_gpio = -1; + aic32x4->mclk_name = "mclk"; } aic32x4->mclk = devm_clk_get(dev, "mclk"); @@ -1228,6 +1231,10 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) return PTR_ERR(aic32x4->mclk); } + ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); + if (ret) + return ret; + if (gpio_is_valid(aic32x4->rstn_gpio)) { ret = devm_gpio_request_one(dev, aic32x4->rstn_gpio, GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index c2d74025bf4b..e2b65bbba7c2 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -16,6 +16,7 @@ struct regmap_config; extern const struct regmap_config aic32x4_regmap_config; int aic32x4_probe(struct device *dev, struct regmap *regmap); int aic32x4_remove(struct device *dev); +int aic32x4_register_clocks(struct device *dev, const char *mclk_name); /* tlv320aic32x4 register space (in decimal to match datasheet) */ @@ -205,4 +206,8 @@ int aic32x4_remove(struct device *dev); #define AIC32X4_RMICPGANIN_IN1L_10K 0x10 #define AIC32X4_RMICPGANIN_CM1R_10K 0x40 +/* Clock Limits */ +#define AIC32X4_MAX_PLL_CLKIN 20000000 + + #endif /* _TLV320AIC32X4_H */ -- cgit v1.2.3-59-g8ed1b From fd2df3aeafa4b4cc468d58e147e0822967034b71 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Thu, 21 Mar 2019 17:58:46 -0700 Subject: ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF Model and manage codec clock input as a component in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4-clk.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 18 +++++++++++++----- 2 files changed, 47 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 5e495fc8d931..cded85009c8c 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -265,6 +265,30 @@ static const struct clk_ops aic32x4_pll_ops = { .get_parent = clk_aic32x4_pll_get_parent, }; +static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, + AIC32X4_CLKMUX, + AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); +} + +static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); + + return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; +} + +static const struct clk_ops aic32x4_codec_clkin_ops = { + .set_parent = clk_aic32x4_codec_clkin_set_parent, + .get_parent = clk_aic32x4_codec_clkin_get_parent, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -274,6 +298,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_pll_ops, .reg = 0, }, + { + .name = "codec_clkin", + .parent_names = + (const char *[]) { "mclk", "bclk", "gpio", "pll" }, + .num_parents = 4, + .ops = &aic32x4_codec_clkin_ops, + .reg = 0, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, @@ -314,6 +346,8 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name) */ aic32x4_clkdesc_array[0].parent_names = (const char* []) { mclk_name, "bclk", "gpio", "din" }; + aic32x4_clkdesc_array[1].parent_names = + (const char *[]) { mclk_name, "bclk", "gpio", "pll" }; for (i = 0; i < ARRAY_SIZE(aic32x4_clkdesc_array); ++i) aic32x4_register_clk(dev, &aic32x4_clkdesc_array[i]); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 7cf8c7cedfe1..5496e4e080f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -735,12 +735,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* PLL as CODEC_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, - AIC32X4_CODEC_CLKIN_MASK, - AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, AIC32X4_BDIVCLK_MASK, + snd_soc_component_update_bits(component, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); /* NDAC divider value */ @@ -987,6 +984,15 @@ static int aic32x4_component_probe(struct snd_soc_component *component) { struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u32 tmp_reg; + int ret; + + struct clk_bulk_data clocks[] = { + { .id = "codec_clkin" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); @@ -999,6 +1005,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (aic32x4->setup) aic32x4_setup_gpios(component); + clk_set_parent(clocks[0].clk, clocks[1].clk); + /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { snd_soc_component_write(component, AIC32X4_MICBIAS, -- cgit v1.2.3-59-g8ed1b From a51b50062091619915c5155085bbe13a7aca6903 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Thu, 21 Mar 2019 17:58:47 -0700 Subject: ASoC: tlv320aic32x4: Model DAC/ADC dividers in CCF Model and manage DAC/ADC dividers as components in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4-clk.c | 90 +++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 101 ++++++++++++++++++++--------------- sound/soc/codecs/tlv320aic32x4.h | 4 ++ 3 files changed, 151 insertions(+), 44 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index cded85009c8c..daf14924e324 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -289,6 +289,68 @@ static const struct clk_ops aic32x4_codec_clkin_ops = { .get_parent = clk_aic32x4_codec_clkin_get_parent, }; +static int clk_aic32x4_div_prepare(struct clk_hw *hw) +{ + struct clk_aic32x4 *div = to_clk_aic32x4(hw); + + return regmap_update_bits(div->regmap, div->reg, + AIC32X4_DIVEN, AIC32X4_DIVEN); +} + +static void clk_aic32x4_div_unprepare(struct clk_hw *hw) +{ + struct clk_aic32x4 *div = to_clk_aic32x4(hw); + + regmap_update_bits(div->regmap, div->reg, + AIC32X4_DIVEN, 0); +} + +static int clk_aic32x4_div_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct clk_aic32x4 *div = to_clk_aic32x4(hw); + u8 divisor; + + divisor = DIV_ROUND_UP(parent_rate, rate); + if (divisor > 128) + return -EINVAL; + + return regmap_update_bits(div->regmap, div->reg, + AIC32X4_DIV_MASK, divisor); +} + +static long clk_aic32x4_div_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + unsigned long divisor; + + divisor = DIV_ROUND_UP(*parent_rate, rate); + if (divisor > 128) + return -EINVAL; + + return DIV_ROUND_UP(*parent_rate, divisor); +} + +static unsigned long clk_aic32x4_div_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct clk_aic32x4 *div = to_clk_aic32x4(hw); + + unsigned int val; + + regmap_read(div->regmap, div->reg, &val); + + return DIV_ROUND_UP(parent_rate, val & AIC32X4_DIV_MASK); +} + +static const struct clk_ops aic32x4_div_ops = { + .prepare = clk_aic32x4_div_prepare, + .unprepare = clk_aic32x4_div_unprepare, + .set_rate = clk_aic32x4_div_set_rate, + .round_rate = clk_aic32x4_div_round_rate, + .recalc_rate = clk_aic32x4_div_recalc_rate, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -306,6 +368,34 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_codec_clkin_ops, .reg = 0, }, + { + .name = "ndac", + .parent_names = (const char * []) { "codec_clkin" }, + .num_parents = 1, + .ops = &aic32x4_div_ops, + .reg = AIC32X4_NDAC, + }, + { + .name = "mdac", + .parent_names = (const char * []) { "ndac" }, + .num_parents = 1, + .ops = &aic32x4_div_ops, + .reg = AIC32X4_MDAC, + }, + { + .name = "nadc", + .parent_names = (const char * []) { "codec_clkin" }, + .num_parents = 1, + .ops = &aic32x4_div_ops, + .reg = AIC32X4_NADC, + }, + { + .name = "madc", + .parent_names = (const char * []) { "nadc" }, + .num_parents = 1, + .ops = &aic32x4_div_ops, + .reg = AIC32X4_MADC, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 5496e4e080f4..0cf942938372 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -52,11 +52,11 @@ struct aic32x4_rate_divs { u32 rate; unsigned long pll_rate; u16 dosr; - u8 ndac; - u8 mdac; + unsigned long ndac_rate; + unsigned long mdac_rate; u8 aosr; - u8 nadc; - u8 madc; + unsigned long nadc_rate; + unsigned long madc_rate; u8 blck_N; u8 r_block; u8 p_block; @@ -309,34 +309,54 @@ static const struct snd_kcontrol_new aic32x4_snd_controls[] = { static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 8k rate */ - { 12000000, 8000, 57120000, 768, 5, 3, 128, 5, 18, 24, 1, 1 }, - { 24000000, 8000, 57120000, 768, 15, 1, 64, 45, 4, 24, 1, 1 }, - { 25000000, 8000, 32620000, 768, 15, 1, 64, 45, 4, 24, 1, 1 }, + { 12000000, 8000, 57120000, 768, 18432000, 6144000, 128, 18432000, + 1024000, 24, 1, 1 }, + { 24000000, 8000, 57120000, 768, 6144000, 6144000, 64, 2048000, + 512000, 24, 1, 1 }, + { 25000000, 8000, 32620000, 768, 6144000, 6144000, 64, 2048000, + 512000, 24, 1, 1 }, /* 11.025k rate */ - { 12000000, 11025, 44217600, 512, 8, 2, 128, 8, 8, 16, 1, 1 }, - { 24000000, 11025, 44217600, 512, 16, 1, 64, 32, 4, 16, 1, 1 }, + { 12000000, 11025, 44217600, 512, 11289600, 5644800, 128, 11289600, + 1411200, 16, 1, 1 }, + { 24000000, 11025, 44217600, 512, 5644800, 5644800, 64, 2822400, + 705600, 16, 1, 1 }, /* 16k rate */ - { 12000000, 16000, 57120000, 384, 5, 3, 128, 5, 9, 12, 1, 1 }, - { 24000000, 16000, 57120000, 384, 15, 1, 64, 18, 5, 12, 1, 1 }, - { 25000000, 16000, 32620000, 384, 15, 1, 64, 18, 5, 12, 1, 1 }, + { 12000000, 16000, 57120000, 384, 18432000, 6144000, 128, 18432000, + 2048000, 12, 1, 1 }, + { 24000000, 16000, 57120000, 384, 6144000, 6144000, 64, 5120000, + 1024000, 12, 1, 1 }, + { 25000000, 16000, 32620000, 384, 6144000, 6144000, 64, 5120000, + 1024000, 12, 1, 1 }, /* 22.05k rate */ - { 12000000, 22050, 44217600, 256, 4, 4, 128, 4, 8, 8, 1, 1 }, - { 24000000, 22050, 44217600, 256, 16, 1, 64, 16, 4, 8, 1, 1 }, - { 25000000, 22050, 19713750, 256, 16, 1, 64, 16, 4, 8, 1, 1 }, + { 12000000, 22050, 44217600, 256, 22579200, 5644800, 128, 22579200, + 2822400, 8, 1, 1 }, + { 24000000, 22050, 44217600, 256, 5644800, 5644800, 64, 5644800, + 1411200, 8, 1, 1 }, + { 25000000, 22050, 19713750, 256, 5644800, 5644800, 64, 5644800, + 1411200, 8, 1, 1 }, /* 32k rate */ - { 12000000, 32000, 14112000, 192, 2, 7, 64, 2, 21, 6, 1, 1 }, - { 24000000, 32000, 14112000, 192, 7, 2, 64, 7, 6, 6, 1, 1 }, + { 12000000, 32000, 14112000, 192, 43008000, 6144000, 64, 43008000, + 2048000, 6, 1, 1 }, + { 24000000, 32000, 14112000, 192, 12288000, 6144000, 64, 12288000, + 2048000, 6, 1, 1 }, /* 44.1k rate */ - { 12000000, 44100, 44217600, 128, 2, 8, 128, 2, 8, 4, 1, 1 }, - { 24000000, 44100, 44217600, 128, 8, 2, 64, 8, 4, 4, 1, 1 }, - { 25000000, 44100, 19713750, 128, 8, 2, 64, 8, 4, 4, 1, 1 }, + { 12000000, 44100, 44217600, 128, 45158400, 5644800, 128, 45158400, + 5644800, 4, 1, 1 }, + { 24000000, 44100, 44217600, 128, 11289600, 5644800, 64, 11289600, + 2822400, 4, 1, 1 }, + { 25000000, 44100, 19713750, 128, 11289600, 5644800, 64, 11289600, + 2822400, 4, 1, 1 }, /* 48k rate */ - { 12000000, 48000, 18432000, 128, 2, 8, 128, 2, 8, 4, 1, 1 }, - { 24000000, 48000, 18432000, 128, 8, 2, 64, 8, 4, 4, 1, 1 }, - { 25000000, 48000, 75626250, 128, 8, 2, 64, 8, 4, 4, 1, 1 }, + { 12000000, 48000, 18432000, 128, 49152000, 6144000, 128, 49152000, + 6144000, 4, 1, 1 }, + { 24000000, 48000, 18432000, 128, 12288000, 6144000, 64, 12288000, + 3072000, 4, 1, 1 }, + { 25000000, 48000, 75626250, 128, 12288000, 6144000, 64, 12288000, + 3072000, 4, 1, 1 }, /* 96k rate */ - { 25000000, 96000, 75626250, 64, 4, 4, 64, 4, 4, 1, 1, 9 }, + { 25000000, 96000, 75626250, 64, 24576000, 6144000, 64, 24576000, + 6144000, 1, 1, 9 }, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -719,6 +739,10 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, struct clk_bulk_data clocks[] = { { .id = "pll" }, + { .id = "nadc" }, + { .id = "madc" }, + { .id = "ndac" }, + { .id = "mdac" }, }; i = aic32x4_get_divs(parent_rate, sample_rate); @@ -731,7 +755,11 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, if (ret) return ret; - clk_set_rate(clocks[0].clk, sample_rate); + clk_set_rate(clocks[0].clk, aic32x4_divs[i].pll_rate); + clk_set_rate(clocks[1].clk, aic32x4_divs[i].nadc_rate); + clk_set_rate(clocks[2].clk, aic32x4_divs[i].madc_rate); + clk_set_rate(clocks[3].clk, aic32x4_divs[i].ndac_rate); + clk_set_rate(clocks[4].clk, aic32x4_divs[i].mdac_rate); aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); @@ -740,26 +768,10 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, AIC32X4_BDIVCLK_MASK, AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); - /* NDAC divider value */ - snd_soc_component_update_bits(component, AIC32X4_NDAC, - AIC32X4_NDAC_MASK, aic32x4_divs[i].ndac); - - /* MDAC divider value */ - snd_soc_component_update_bits(component, AIC32X4_MDAC, - AIC32X4_MDAC_MASK, aic32x4_divs[i].mdac); - /* DOSR MSB & LSB values */ snd_soc_component_write(component, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); snd_soc_component_write(component, AIC32X4_DOSRLSB, (aic32x4_divs[i].dosr & 0xff)); - /* NADC divider value */ - snd_soc_component_update_bits(component, AIC32X4_NADC, - AIC32X4_NADC_MASK, aic32x4_divs[i].nadc); - - /* MADC divider value */ - snd_soc_component_update_bits(component, AIC32X4_MADC, - AIC32X4_MADC_MASK, aic32x4_divs[i].madc); - /* AOSR value */ snd_soc_component_write(component, AIC32X4_AOSR, aic32x4_divs[i].aosr); @@ -771,8 +783,8 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, } static int aic32x4_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); @@ -987,7 +999,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) int ret; struct clk_bulk_data clocks[] = { - { .id = "codec_clkin" }, + { .id = "codec_clkin" }, + { .id = "pll" }, }; ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index e2b65bbba7c2..6ede877b00a0 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -206,6 +206,10 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name); #define AIC32X4_RMICPGANIN_IN1L_10K 0x10 #define AIC32X4_RMICPGANIN_CM1R_10K 0x40 +/* Common mask and enable for all of the dividers */ +#define AIC32X4_DIVEN BIT(7) +#define AIC32X4_DIV_MASK GENMASK(6, 0) + /* Clock Limits */ #define AIC32X4_MAX_PLL_CLKIN 20000000 -- cgit v1.2.3-59-g8ed1b From 9b484124ebd906c4d6bc826cc0d417e80cc1105c Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Thu, 21 Mar 2019 17:58:48 -0700 Subject: ASoC: tlv320aic32x4: Model BDIV divider in CCF Model and manage BDIV divider as components in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4-clk.c | 36 +++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 56 +++++++++++++++++------------------- 2 files changed, 62 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index daf14924e324..667ec2c03508 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -351,6 +351,34 @@ static const struct clk_ops aic32x4_div_ops = { .recalc_rate = clk_aic32x4_div_recalc_rate, }; +static int clk_aic32x4_bdiv_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, index); +} + +static u8 clk_aic32x4_bdiv_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_IFACE3, &val); + + return val & AIC32X4_BDIVCLK_MASK; +} + +static const struct clk_ops aic32x4_bdiv_ops = { + .prepare = clk_aic32x4_div_prepare, + .unprepare = clk_aic32x4_div_unprepare, + .set_parent = clk_aic32x4_bdiv_set_parent, + .get_parent = clk_aic32x4_bdiv_get_parent, + .set_rate = clk_aic32x4_div_set_rate, + .round_rate = clk_aic32x4_div_round_rate, + .recalc_rate = clk_aic32x4_div_recalc_rate, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -396,6 +424,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_div_ops, .reg = AIC32X4_MADC, }, + { + .name = "bdiv", + .parent_names = + (const char *[]) { "ndac", "mdac", "nadc", "madc" }, + .num_parents = 4, + .ops = &aic32x4_bdiv_ops, + .reg = AIC32X4_BCLKN, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 0cf942938372..bf9ccda6616d 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -57,7 +57,7 @@ struct aic32x4_rate_divs { u8 aosr; unsigned long nadc_rate; unsigned long madc_rate; - u8 blck_N; + unsigned long bdiv_rate; u8 r_block; u8 p_block; }; @@ -310,53 +310,53 @@ static const struct snd_kcontrol_new aic32x4_snd_controls[] = { static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 8k rate */ { 12000000, 8000, 57120000, 768, 18432000, 6144000, 128, 18432000, - 1024000, 24, 1, 1 }, + 1024000, 256000, 1, 1 }, { 24000000, 8000, 57120000, 768, 6144000, 6144000, 64, 2048000, - 512000, 24, 1, 1 }, + 512000, 256000, 1, 1 }, { 25000000, 8000, 32620000, 768, 6144000, 6144000, 64, 2048000, - 512000, 24, 1, 1 }, + 512000, 256000, 1, 1 }, /* 11.025k rate */ { 12000000, 11025, 44217600, 512, 11289600, 5644800, 128, 11289600, - 1411200, 16, 1, 1 }, + 1411200, 352800, 1, 1 }, { 24000000, 11025, 44217600, 512, 5644800, 5644800, 64, 2822400, - 705600, 16, 1, 1 }, + 705600, 352800, 1, 1 }, /* 16k rate */ { 12000000, 16000, 57120000, 384, 18432000, 6144000, 128, 18432000, - 2048000, 12, 1, 1 }, + 2048000, 512000, 1, 1 }, { 24000000, 16000, 57120000, 384, 6144000, 6144000, 64, 5120000, - 1024000, 12, 1, 1 }, + 1024000, 512000, 1, 1 }, { 25000000, 16000, 32620000, 384, 6144000, 6144000, 64, 5120000, - 1024000, 12, 1, 1 }, + 1024000, 512000, 1, 1 }, /* 22.05k rate */ { 12000000, 22050, 44217600, 256, 22579200, 5644800, 128, 22579200, - 2822400, 8, 1, 1 }, + 2822400, 705600, 1, 1 }, { 24000000, 22050, 44217600, 256, 5644800, 5644800, 64, 5644800, - 1411200, 8, 1, 1 }, + 1411200, 705600, 1, 1 }, { 25000000, 22050, 19713750, 256, 5644800, 5644800, 64, 5644800, - 1411200, 8, 1, 1 }, + 1411200, 705600, 1, 1 }, /* 32k rate */ { 12000000, 32000, 14112000, 192, 43008000, 6144000, 64, 43008000, - 2048000, 6, 1, 1 }, + 2048000, 1024000, 1, 1 }, { 24000000, 32000, 14112000, 192, 12288000, 6144000, 64, 12288000, - 2048000, 6, 1, 1 }, + 2048000, 1024000, 1, 1 }, /* 44.1k rate */ { 12000000, 44100, 44217600, 128, 45158400, 5644800, 128, 45158400, - 5644800, 4, 1, 1 }, + 5644800, 1411200, 1, 1 }, { 24000000, 44100, 44217600, 128, 11289600, 5644800, 64, 11289600, - 2822400, 4, 1, 1 }, + 2822400, 1411200, 1, 1 }, { 25000000, 44100, 19713750, 128, 11289600, 5644800, 64, 11289600, - 2822400, 4, 1, 1 }, + 2822400, 1411200, 1, 1 }, /* 48k rate */ { 12000000, 48000, 18432000, 128, 49152000, 6144000, 128, 49152000, - 6144000, 4, 1, 1 }, + 6144000, 1536000, 1, 1 }, { 24000000, 48000, 18432000, 128, 12288000, 6144000, 64, 12288000, - 3072000, 4, 1, 1 }, + 3072000, 1536000, 1, 1 }, { 25000000, 48000, 75626250, 128, 12288000, 6144000, 64, 12288000, - 3072000, 4, 1, 1 }, + 3072000, 1536000, 1, 1 }, /* 96k rate */ { 25000000, 96000, 75626250, 64, 24576000, 6144000, 64, 24576000, - 6144000, 1, 1, 9 }, + 6144000, 3072000, 1, 9 }, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -743,6 +743,7 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, { .id = "madc" }, { .id = "ndac" }, { .id = "mdac" }, + { .id = "bdiv" }, }; i = aic32x4_get_divs(parent_rate, sample_rate); @@ -760,14 +761,10 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, clk_set_rate(clocks[2].clk, aic32x4_divs[i].madc_rate); clk_set_rate(clocks[3].clk, aic32x4_divs[i].ndac_rate); clk_set_rate(clocks[4].clk, aic32x4_divs[i].mdac_rate); + clk_set_rate(clocks[5].clk, aic32x4_divs[i].bdiv_rate); aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, - AIC32X4_BDIVCLK_MASK, - AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); - /* DOSR MSB & LSB values */ snd_soc_component_write(component, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); snd_soc_component_write(component, AIC32X4_DOSRLSB, (aic32x4_divs[i].dosr & 0xff)); @@ -775,10 +772,6 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, /* AOSR value */ snd_soc_component_write(component, AIC32X4_AOSR, aic32x4_divs[i].aosr); - /* BCLK N divider */ - snd_soc_component_update_bits(component, AIC32X4_BCLKN, - AIC32X4_BCLK_MASK, aic32x4_divs[i].blck_N); - return 0; } @@ -1001,6 +994,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) struct clk_bulk_data clocks[] = { { .id = "codec_clkin" }, { .id = "pll" }, + { .id = "bdiv" }, + { .id = "mdac" }, }; ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); @@ -1019,6 +1014,7 @@ static int aic32x4_component_probe(struct snd_soc_component *component) aic32x4_setup_gpios(component); clk_set_parent(clocks[0].clk, clocks[1].clk); + clk_set_parent(clocks[2].clk, clocks[3].clk); /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { -- cgit v1.2.3-59-g8ed1b From d25970b5fd51e9fcf0afbe190908ea4049454da4 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Thu, 21 Mar 2019 17:58:49 -0700 Subject: ASoC: tlv320aic32x4: Control clock gating with CCF Control the clock gating to the various clock components to use the CCF. This allows us to prepare_enalbe only 3 clocks and the relationships assigned to them will cause upstream clockss to enable automatically. Additionally we can do this in a single call to the CCF. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 67 ++++++++-------------------------------- 1 file changed, 13 insertions(+), 54 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index bf9ccda6616d..c3e9f65b26a7 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -834,41 +834,25 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute) static int aic32x4_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { - struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); int ret; + struct clk_bulk_data clocks[] = { + { .id = "madc" }, + { .id = "mdac" }, + { .id = "bdiv" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; + switch (level) { case SND_SOC_BIAS_ON: - /* Switch on master clock */ - ret = clk_prepare_enable(aic32x4->mclk); + ret = clk_bulk_prepare_enable(ARRAY_SIZE(clocks), clocks); if (ret) { - dev_err(component->dev, "Failed to enable master clock\n"); + dev_err(component->dev, "Failed to enable clocks\n"); return ret; } - - /* Switch on PLL */ - snd_soc_component_update_bits(component, AIC32X4_PLLPR, - AIC32X4_PLLEN, AIC32X4_PLLEN); - - /* Switch on NDAC Divider */ - snd_soc_component_update_bits(component, AIC32X4_NDAC, - AIC32X4_NDACEN, AIC32X4_NDACEN); - - /* Switch on MDAC Divider */ - snd_soc_component_update_bits(component, AIC32X4_MDAC, - AIC32X4_MDACEN, AIC32X4_MDACEN); - - /* Switch on NADC Divider */ - snd_soc_component_update_bits(component, AIC32X4_NADC, - AIC32X4_NADCEN, AIC32X4_NADCEN); - - /* Switch on MADC Divider */ - snd_soc_component_update_bits(component, AIC32X4_MADC, - AIC32X4_MADCEN, AIC32X4_MADCEN); - - /* Switch on BCLK_N Divider */ - snd_soc_component_update_bits(component, AIC32X4_BCLKN, - AIC32X4_BCLKEN, AIC32X4_BCLKEN); break; case SND_SOC_BIAS_PREPARE: break; @@ -877,32 +861,7 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) break; - /* Switch off BCLK_N Divider */ - snd_soc_component_update_bits(component, AIC32X4_BCLKN, - AIC32X4_BCLKEN, 0); - - /* Switch off MADC Divider */ - snd_soc_component_update_bits(component, AIC32X4_MADC, - AIC32X4_MADCEN, 0); - - /* Switch off NADC Divider */ - snd_soc_component_update_bits(component, AIC32X4_NADC, - AIC32X4_NADCEN, 0); - - /* Switch off MDAC Divider */ - snd_soc_component_update_bits(component, AIC32X4_MDAC, - AIC32X4_MDACEN, 0); - - /* Switch off NDAC Divider */ - snd_soc_component_update_bits(component, AIC32X4_NDAC, - AIC32X4_NDACEN, 0); - - /* Switch off PLL */ - snd_soc_component_update_bits(component, AIC32X4_PLLPR, - AIC32X4_PLLEN, 0); - - /* Switch off master clock */ - clk_disable_unprepare(aic32x4->mclk); + clk_bulk_disable_unprepare(ARRAY_SIZE(clocks), clocks); break; case SND_SOC_BIAS_OFF: break; -- cgit v1.2.3-59-g8ed1b From fbafbf6517274a797e6e6508c18dd8dba5920c89 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Thu, 21 Mar 2019 17:58:50 -0700 Subject: ASoC: tlv320aic32x4: Move aosr and dosr setting to separate functions Move these to separate helper functions. This looks cleaner and fits better with the new clock setting in CCF. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c3e9f65b26a7..bf3a80237b03 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -718,6 +718,20 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } +static int aic32x4_set_aosr(struct snd_soc_component *component, u8 aosr) +{ + return snd_soc_component_write(component, AIC32X4_AOSR, aosr); +} + +static int aic32x4_set_dosr(struct snd_soc_component *component, u16 dosr) +{ + snd_soc_component_write(component, AIC32X4_DOSRMSB, dosr >> 8); + snd_soc_component_write(component, AIC32X4_DOSRLSB, + (dosr & 0xff)); + + return 0; +} + static int aic32x4_set_processing_blocks(struct snd_soc_component *component, u8 r_block, u8 p_block) { @@ -763,14 +777,10 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, clk_set_rate(clocks[4].clk, aic32x4_divs[i].mdac_rate); clk_set_rate(clocks[5].clk, aic32x4_divs[i].bdiv_rate); - aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); + aic32x4_set_aosr(component, aic32x4_divs[i].aosr); + aic32x4_set_dosr(component, aic32x4_divs[i].dosr); - /* DOSR MSB & LSB values */ - snd_soc_component_write(component, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); - snd_soc_component_write(component, AIC32X4_DOSRLSB, (aic32x4_divs[i].dosr & 0xff)); - - /* AOSR value */ - snd_soc_component_write(component, AIC32X4_AOSR, aic32x4_divs[i].aosr); + aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); return 0; } -- cgit v1.2.3-59-g8ed1b From 96c3bb00239de4fb5f4ddca42c1f90d6d9b3c697 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Thu, 21 Mar 2019 17:58:51 -0700 Subject: ASoC: tlv320aic32x4: Dynamically Determine Clocking The existing code uses a static lookup table to determine the settings of the various clock devices on board the chip. This is limiting in a couple of ways. First, this doesn't allow for any master clock rates other than the three that have been precalculated. Additionally, new sample rates are difficult to add to the table. Witness that the chip is capable of 192000 Hz sampling, but it is not provided by this driver. Last, if the driver is clocked by something that isn't a crystal, the upstream clock may not be able to achieve exactly the rate requested in the driver. This will mean that clocking will be slightly off for the sampling clock or that it won't work at all. This patch determines the settings for all of the clocks at runtime considering the real conditions of the clocks in the system. The rules for the clocks are in TI's SLAA557 application guide on pages 37, 51 and 77. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 190 ++++++++++++++++++--------------------- sound/soc/codecs/tlv320aic32x4.h | 4 +- 2 files changed, 90 insertions(+), 104 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index bf3a80237b03..625528500f8d 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -47,21 +47,6 @@ #include "tlv320aic32x4.h" -struct aic32x4_rate_divs { - u32 mclk; - u32 rate; - unsigned long pll_rate; - u16 dosr; - unsigned long ndac_rate; - unsigned long mdac_rate; - u8 aosr; - unsigned long nadc_rate; - unsigned long madc_rate; - unsigned long bdiv_rate; - u8 r_block; - u8 p_block; -}; - struct aic32x4_priv { struct regmap *regmap; u32 sysclk; @@ -307,58 +292,6 @@ static const struct snd_kcontrol_new aic32x4_snd_controls[] = { 0, 0x0F, 0), }; -static const struct aic32x4_rate_divs aic32x4_divs[] = { - /* 8k rate */ - { 12000000, 8000, 57120000, 768, 18432000, 6144000, 128, 18432000, - 1024000, 256000, 1, 1 }, - { 24000000, 8000, 57120000, 768, 6144000, 6144000, 64, 2048000, - 512000, 256000, 1, 1 }, - { 25000000, 8000, 32620000, 768, 6144000, 6144000, 64, 2048000, - 512000, 256000, 1, 1 }, - /* 11.025k rate */ - { 12000000, 11025, 44217600, 512, 11289600, 5644800, 128, 11289600, - 1411200, 352800, 1, 1 }, - { 24000000, 11025, 44217600, 512, 5644800, 5644800, 64, 2822400, - 705600, 352800, 1, 1 }, - /* 16k rate */ - { 12000000, 16000, 57120000, 384, 18432000, 6144000, 128, 18432000, - 2048000, 512000, 1, 1 }, - { 24000000, 16000, 57120000, 384, 6144000, 6144000, 64, 5120000, - 1024000, 512000, 1, 1 }, - { 25000000, 16000, 32620000, 384, 6144000, 6144000, 64, 5120000, - 1024000, 512000, 1, 1 }, - /* 22.05k rate */ - { 12000000, 22050, 44217600, 256, 22579200, 5644800, 128, 22579200, - 2822400, 705600, 1, 1 }, - { 24000000, 22050, 44217600, 256, 5644800, 5644800, 64, 5644800, - 1411200, 705600, 1, 1 }, - { 25000000, 22050, 19713750, 256, 5644800, 5644800, 64, 5644800, - 1411200, 705600, 1, 1 }, - /* 32k rate */ - { 12000000, 32000, 14112000, 192, 43008000, 6144000, 64, 43008000, - 2048000, 1024000, 1, 1 }, - { 24000000, 32000, 14112000, 192, 12288000, 6144000, 64, 12288000, - 2048000, 1024000, 1, 1 }, - /* 44.1k rate */ - { 12000000, 44100, 44217600, 128, 45158400, 5644800, 128, 45158400, - 5644800, 1411200, 1, 1 }, - { 24000000, 44100, 44217600, 128, 11289600, 5644800, 64, 11289600, - 2822400, 1411200, 1, 1 }, - { 25000000, 44100, 19713750, 128, 11289600, 5644800, 64, 11289600, - 2822400, 1411200, 1, 1 }, - /* 48k rate */ - { 12000000, 48000, 18432000, 128, 49152000, 6144000, 128, 49152000, - 6144000, 1536000, 1, 1 }, - { 24000000, 48000, 18432000, 128, 12288000, 6144000, 64, 12288000, - 3072000, 1536000, 1, 1 }, - { 25000000, 48000, 75626250, 128, 12288000, 6144000, 64, 12288000, - 3072000, 1536000, 1, 1 }, - - /* 96k rate */ - { 25000000, 96000, 75626250, 64, 24576000, 6144000, 64, 24576000, - 6144000, 3072000, 1, 9 }, -}; - static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_HPLROUTE, 3, 1, 0), SOC_DAPM_SINGLE("IN1_L Switch", AIC32X4_HPLROUTE, 2, 1, 0), @@ -630,20 +563,6 @@ const struct regmap_config aic32x4_regmap_config = { }; EXPORT_SYMBOL(aic32x4_regmap_config); -static inline int aic32x4_get_divs(int mclk, int rate) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) { - if ((aic32x4_divs[i].rate == rate) - && (aic32x4_divs[i].mclk == mclk)) { - return i; - } - } - printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); - return -EINVAL; -} - static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -745,11 +664,17 @@ static int aic32x4_set_processing_blocks(struct snd_soc_component *component, } static int aic32x4_setup_clocks(struct snd_soc_component *component, - unsigned int sample_rate, - unsigned int parent_rate) + unsigned int sample_rate) { - int i; + u8 aosr; + u16 dosr; + u8 adc_resource_class, dac_resource_class; + u8 madc, nadc, mdac, ndac, max_nadc, min_mdac, max_ndac; + u8 dosr_increment; + u16 max_dosr, min_dosr; + unsigned long mclk_rate, adc_clock_rate, dac_clock_rate; int ret; + struct clk *mclk; struct clk_bulk_data clocks[] = { { .id = "pll" }, @@ -759,30 +684,89 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, { .id = "mdac" }, { .id = "bdiv" }, }; - - i = aic32x4_get_divs(parent_rate, sample_rate); - if (i < 0) { - printk(KERN_ERR "aic32x4: sampling rate not supported\n"); - return i; - } - ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); if (ret) return ret; - clk_set_rate(clocks[0].clk, aic32x4_divs[i].pll_rate); - clk_set_rate(clocks[1].clk, aic32x4_divs[i].nadc_rate); - clk_set_rate(clocks[2].clk, aic32x4_divs[i].madc_rate); - clk_set_rate(clocks[3].clk, aic32x4_divs[i].ndac_rate); - clk_set_rate(clocks[4].clk, aic32x4_divs[i].mdac_rate); - clk_set_rate(clocks[5].clk, aic32x4_divs[i].bdiv_rate); - - aic32x4_set_aosr(component, aic32x4_divs[i].aosr); - aic32x4_set_dosr(component, aic32x4_divs[i].dosr); + mclk = clk_get_parent(clocks[1].clk); + mclk_rate = clk_get_rate(mclk); + + if (sample_rate <= 48000) { + aosr = 128; + adc_resource_class = 6; + dac_resource_class = 8; + dosr_increment = 8; + aic32x4_set_processing_blocks(component, 1, 1); + } else if (sample_rate <= 96000) { + aosr = 64; + adc_resource_class = 6; + dac_resource_class = 8; + dosr_increment = 4; + aic32x4_set_processing_blocks(component, 1, 9); + } else if (sample_rate == 192000) { + aosr = 32; + adc_resource_class = 3; + dac_resource_class = 4; + dosr_increment = 2; + aic32x4_set_processing_blocks(component, 13, 19); + } else { + dev_err(component->dev, "Sampling rate not supported\n"); + return -EINVAL; + } - aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); + madc = DIV_ROUND_UP((32 * adc_resource_class), aosr); + max_dosr = (AIC32X4_MAX_DOSR_FREQ / sample_rate / dosr_increment) * + dosr_increment; + min_dosr = (AIC32X4_MIN_DOSR_FREQ / sample_rate / dosr_increment) * + dosr_increment; + max_nadc = AIC32X4_MAX_CODEC_CLKIN_FREQ / (madc * aosr * sample_rate); + + for (nadc = max_nadc; nadc > 0; --nadc) { + adc_clock_rate = nadc * madc * aosr * sample_rate; + for (dosr = max_dosr; dosr >= min_dosr; + dosr -= dosr_increment) { + min_mdac = DIV_ROUND_UP((32 * dac_resource_class), dosr); + max_ndac = AIC32X4_MAX_CODEC_CLKIN_FREQ / + (min_mdac * dosr * sample_rate); + for (mdac = min_mdac; mdac <= 128; ++mdac) { + for (ndac = max_ndac; ndac > 0; --ndac) { + dac_clock_rate = ndac * mdac * dosr * + sample_rate; + if (dac_clock_rate == adc_clock_rate) { + if (clk_round_rate(clocks[0].clk, dac_clock_rate) == 0) + continue; + + clk_set_rate(clocks[0].clk, + dac_clock_rate); + + clk_set_rate(clocks[1].clk, + sample_rate * aosr * + madc); + clk_set_rate(clocks[2].clk, + sample_rate * aosr); + aic32x4_set_aosr(component, + aosr); + + clk_set_rate(clocks[3].clk, + sample_rate * dosr * + mdac); + clk_set_rate(clocks[4].clk, + sample_rate * dosr); + aic32x4_set_dosr(component, + dosr); + + clk_set_rate(clocks[5].clk, + sample_rate * 32); + return 0; + } + } + } + } + } - return 0; + dev_err(component->dev, + "Could not set clocks to support sample rate.\n"); + return -EINVAL; } static int aic32x4_hw_params(struct snd_pcm_substream *substream, @@ -794,7 +778,7 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, u8 iface1_reg = 0; u8 dacsetup_reg = 0; - aic32x4_setup_clocks(component, params_rate(params), aic32x4->sysclk); + aic32x4_setup_clocks(component, params_rate(params)); switch (params_width(params)) { case 16: diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index 6ede877b00a0..88205bc97198 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -211,7 +211,9 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name); #define AIC32X4_DIV_MASK GENMASK(6, 0) /* Clock Limits */ +#define AIC32X4_MAX_DOSR_FREQ 6200000 +#define AIC32X4_MIN_DOSR_FREQ 2800000 +#define AIC32X4_MAX_CODEC_CLKIN_FREQ 110000000 #define AIC32X4_MAX_PLL_CLKIN 20000000 - #endif /* _TLV320AIC32X4_H */ -- cgit v1.2.3-59-g8ed1b From aa6a60f7be925210d5156f0e8025f3afe1f4f54d Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Thu, 21 Mar 2019 17:58:52 -0700 Subject: ASoC: tlv320aic32x4: Restructure set_dai_sysclk The sysclk is now managed by the CCF. Change this function to merely find the system clock and set it using clk_set_rate. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 625528500f8d..289bf411e60e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -49,7 +49,6 @@ struct aic32x4_priv { struct regmap *regmap; - u32 sysclk; u32 power_cfg; u32 micpga_routing; bool swapdacs; @@ -567,17 +566,13 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_component *component = codec_dai->component; - struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); + struct clk *mclk; + struct clk *pll; - switch (freq) { - case 12000000: - case 24000000: - case 25000000: - aic32x4->sysclk = freq; - return 0; - } - printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); - return -EINVAL; + pll = devm_clk_get(component->dev, "pll"); + mclk = clk_get_parent(pll); + + return clk_set_rate(mclk, freq); } static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) -- cgit v1.2.3-59-g8ed1b From 78f2d58a289302e56a7def96a783a7686ebf27e2 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Thu, 21 Mar 2019 17:58:53 -0700 Subject: ASoC: tlv320aic32x4: Remove mclk references mclk is not used by anything anymore. Remove support for it. All that information now comes from the clock tree. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 289bf411e60e..a5457d3de1b1 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -53,7 +53,6 @@ struct aic32x4_priv { u32 micpga_routing; bool swapdacs; int rstn_gpio; - struct clk *mclk; const char *mclk_name; struct regulator *supply_ldo; @@ -1190,12 +1189,6 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) aic32x4->mclk_name = "mclk"; } - aic32x4->mclk = devm_clk_get(dev, "mclk"); - if (IS_ERR(aic32x4->mclk)) { - dev_err(dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); - return PTR_ERR(aic32x4->mclk); - } - ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From 6d56ee1550b8a81bc63c80051ff78d8d704b09ba Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Thu, 21 Mar 2019 17:58:54 -0700 Subject: ASoC: tlv320aic32x4: Allow 192000 Sample Rate The clocking and processing blocks are now properly set up to support 192000 sample rates. Allow drivers to ask for that. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index a5457d3de1b1..75443efeda69 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -857,7 +857,7 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, return 0; } -#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_192000 #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -- cgit v1.2.3-59-g8ed1b From 64f01d2b5ccc621c3aa66b82daf9154f5581f36a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 Mar 2019 13:10:13 +0000 Subject: ASoC: tlv320aic32x4: Only enable with common clock Some architectures do not yet support the common clock API at all but the tlv320aic32x4 driver now requires it. Reported-by: Stephen Rothwell Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6e99320c79b8..419142111b6d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,8 +180,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4_I2C if I2C - select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER + select SND_SOC_TLV320AIC32X4_I2C if I2C && COMMON_CLK + select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER && COMMON_CLK select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -1110,11 +1110,13 @@ config SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC32X4_I2C tristate "Texas Instruments TLV320AIC32x4 audio CODECs - I2C" depends on I2C + depends on COMMON_CLK select SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC32X4_SPI tristate "Texas Instruments TLV320AIC32x4 audio CODECs - SPI" depends on SPI_MASTER + depends on COMMON_CLK select SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC3X -- cgit v1.2.3-59-g8ed1b From be1df61cf06efb355c90702e46b8d46f055acb4e Mon Sep 17 00:00:00 2001 From: Viorel Suman Date: Tue, 22 Jan 2019 11:14:26 +0000 Subject: ASoC: fsl: Add Audio Mixer CPU DAI driver This patch implements Audio Mixer CPU DAI driver for NXP iMX8 SOCs. The Audio Mixer is a on-chip functional module that allows mixing of two audio streams into a single audio stream. Audio Mixer datasheet is available here: https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf Signed-off-by: Viorel Suman Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 3 + sound/soc/fsl/fsl_audmix.c | 576 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_audmix.h | 102 ++++++++ 4 files changed, 688 insertions(+) create mode 100644 sound/soc/fsl/fsl_audmix.c create mode 100644 sound/soc/fsl/fsl_audmix.h (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7b1d9970be8b..0af2e056d211 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,6 +24,13 @@ config SND_SOC_FSL_SAI This option is only useful for out-of-tree drivers since in-tree drivers select it automatically. +config SND_SOC_FSL_AUDMIX + tristate "Audio Mixer (AUDMIX) module support" + select REGMAP_MMIO + help + Say Y if you want to add Audio Mixer (AUDMIX) + support for the NXP iMX CPUs. + config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 3c0ff315b971..4172d5a3e36c 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -12,6 +12,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-audmix-objs := fsl_audmix.o snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o @@ -22,6 +23,8 @@ snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-micfil-objs := fsl_micfil.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o + +obj-$(CONFIG_SND_SOC_FSL_AUDMIX) += snd-soc-fsl-audmix.o obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c new file mode 100644 index 000000000000..3356cb617713 --- /dev/null +++ b/sound/soc/fsl/fsl_audmix.c @@ -0,0 +1,576 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * NXP AUDMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#include +#include +#include +#include +#include +#include + +#include "fsl_audmix.h" + +#define SOC_ENUM_SINGLE_S(xreg, xshift, xtexts) \ + SOC_ENUM_SINGLE(xreg, xshift, ARRAY_SIZE(xtexts), xtexts) + +static const char + *tdm_sel[] = { "TDM1", "TDM2", }, + *mode_sel[] = { "Disabled", "TDM1", "TDM2", "Mixed", }, + *width_sel[] = { "16b", "18b", "20b", "24b", "32b", }, + *endis_sel[] = { "Disabled", "Enabled", }, + *updn_sel[] = { "Downward", "Upward", }, + *mask_sel[] = { "Unmask", "Mask", }; + +static const struct soc_enum fsl_audmix_enum[] = { +/* FSL_AUDMIX_CTR enums */ +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MIXCLK_SHIFT, tdm_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_OUTSRC_SHIFT, mode_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_OUTWIDTH_SHIFT, width_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MASKRTDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_MASKCKDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_SYNCMODE_SHIFT, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_CTR, FSL_AUDMIX_CTR_SYNCSRC_SHIFT, tdm_sel), +/* FSL_AUDMIX_ATCR0 enums */ +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR0, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR0, 1, updn_sel), +/* FSL_AUDMIX_ATCR1 enums */ +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR1, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AUDMIX_ATCR1, 1, updn_sel), +}; + +struct fsl_audmix_state { + u8 tdms; + u8 clk; + char msg[64]; +}; + +static const struct fsl_audmix_state prms[4][4] = {{ + /* DIS->DIS, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* DIS->TDM1*/ + { .tdms = 1, .clk = 1, .msg = "DIS->TDM1: TDM1 not started!\n" }, + /* DIS->TDM2*/ + { .tdms = 2, .clk = 2, .msg = "DIS->TDM2: TDM2 not started!\n" }, + /* DIS->MIX */ + { .tdms = 3, .clk = 0, .msg = "DIS->MIX: Please start both TDMs!\n" } +}, { /* TDM1->DIS */ + { .tdms = 1, .clk = 0, .msg = "TDM1->DIS: TDM1 not started!\n" }, + /* TDM1->TDM1, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM1->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "TDM1->TDM2: Please start both TDMs!\n" }, + /* TDM1->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM1->MIX: Please start both TDMs!\n" } +}, { /* TDM2->DIS */ + { .tdms = 2, .clk = 0, .msg = "TDM2->DIS: TDM2 not started!\n" }, + /* TDM2->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "TDM2->TDM1: Please start both TDMs!\n" }, + /* TDM2->TDM2, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM2->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM2->MIX: Please start both TDMs!\n" } +}, { /* MIX->DIS */ + { .tdms = 3, .clk = 0, .msg = "MIX->DIS: Please start both TDMs!\n" }, + /* MIX->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "MIX->TDM1: Please start both TDMs!\n" }, + /* MIX->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "MIX->TDM2: Please start both TDMs!\n" }, + /* MIX->MIX, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" } +}, }; + +static int fsl_audmix_state_trans(struct snd_soc_component *comp, + unsigned int *mask, unsigned int *ctr, + const struct fsl_audmix_state prm) +{ + struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); + /* Enforce all required TDMs are started */ + if ((priv->tdms & prm.tdms) != prm.tdms) { + dev_dbg(comp->dev, prm.msg); + return -EINVAL; + } + + switch (prm.clk) { + case 1: + case 2: + /* Set mix clock */ + (*mask) |= FSL_AUDMIX_CTR_MIXCLK_MASK; + (*ctr) |= FSL_AUDMIX_CTR_MIXCLK(prm.clk - 1); + break; + default: + break; + } + + return 0; +} + +static int fsl_audmix_put_mix_clk_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int reg_val, val, mix_clk; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, ®_val); + if (ret) + return ret; + + mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK) + >> FSL_AUDMIX_CTR_MIXCLK_SHIFT); + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + if (!(priv->tdms & BIT(val))) { + dev_err(comp->dev, + "The selected clock source has no TDM%d enabled!\n", + val + 1); + return -EINVAL; + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + +static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + u32 out_src, mix_clk; + unsigned int reg_val, val, mask = 0, ctr = 0; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AUDMIX_CTR, ®_val); + if (ret) + return ret; + + /* "From" state */ + out_src = ((reg_val & FSL_AUDMIX_CTR_OUTSRC_MASK) + >> FSL_AUDMIX_CTR_OUTSRC_SHIFT); + mix_clk = ((reg_val & FSL_AUDMIX_CTR_MIXCLK_MASK) + >> FSL_AUDMIX_CTR_MIXCLK_SHIFT); + + /* "To" state */ + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /* Check if state is changing ... */ + if (out_src == val) + return 0; + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + /* Check state transition constraints */ + ret = fsl_audmix_state_trans(comp, &mask, &ctr, prms[out_src][val]); + if (ret) + return ret; + + /* Complete transition to new state */ + mask |= FSL_AUDMIX_CTR_OUTSRC_MASK; + ctr |= FSL_AUDMIX_CTR_OUTSRC(val); + + return snd_soc_component_update_bits(comp, FSL_AUDMIX_CTR, mask, ctr); +} + +static const struct snd_kcontrol_new fsl_audmix_snd_controls[] = { + /* FSL_AUDMIX_CTR controls */ + SOC_ENUM_EXT("Mixing Clock Source", fsl_audmix_enum[0], + snd_soc_get_enum_double, fsl_audmix_put_mix_clk_src), + SOC_ENUM_EXT("Output Source", fsl_audmix_enum[1], + snd_soc_get_enum_double, fsl_audmix_put_out_src), + SOC_ENUM("Output Width", fsl_audmix_enum[2]), + SOC_ENUM("Frame Rate Diff Error", fsl_audmix_enum[3]), + SOC_ENUM("Clock Freq Diff Error", fsl_audmix_enum[4]), + SOC_ENUM("Sync Mode Config", fsl_audmix_enum[5]), + SOC_ENUM("Sync Mode Clk Source", fsl_audmix_enum[6]), + /* TDM1 Attenuation controls */ + SOC_ENUM("TDM1 Attenuation", fsl_audmix_enum[7]), + SOC_ENUM("TDM1 Attenuation Direction", fsl_audmix_enum[8]), + SOC_SINGLE("TDM1 Attenuation Step Divider", FSL_AUDMIX_ATCR0, + 2, 0x00fff, 0), + SOC_SINGLE("TDM1 Attenuation Initial Value", FSL_AUDMIX_ATIVAL0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Up Factor", FSL_AUDMIX_ATSTPUP0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Down Factor", FSL_AUDMIX_ATSTPDN0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Target", FSL_AUDMIX_ATSTPTGT0, + 0, 0x3ffff, 0), + /* TDM2 Attenuation controls */ + SOC_ENUM("TDM2 Attenuation", fsl_audmix_enum[9]), + SOC_ENUM("TDM2 Attenuation Direction", fsl_audmix_enum[10]), + SOC_SINGLE("TDM2 Attenuation Step Divider", FSL_AUDMIX_ATCR1, + 2, 0x00fff, 0), + SOC_SINGLE("TDM2 Attenuation Initial Value", FSL_AUDMIX_ATIVAL1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Up Factor", FSL_AUDMIX_ATSTPUP1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Down Factor", FSL_AUDMIX_ATSTPDN1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Target", FSL_AUDMIX_ATSTPTGT1, + 0, 0x3ffff, 0), +}; + +static int fsl_audmix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *comp = dai->component; + u32 mask = 0, ctr = 0; + + /* AUDMIX is working in DSP_A format only */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + /* For playback the AUDMIX is slave, and for record is master */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + /* Output data will be written on positive edge of the clock */ + ctr |= FSL_AUDMIX_CTR_OUTCKPOL(0); + break; + case SND_SOC_DAIFMT_NB_NF: + /* Output data will be written on negative edge of the clock */ + ctr |= FSL_AUDMIX_CTR_OUTCKPOL(1); + break; + default: + return -EINVAL; + } + + mask |= FSL_AUDMIX_CTR_OUTCKPOL_MASK; + + return snd_soc_component_update_bits(comp, FSL_AUDMIX_CTR, mask, ctr); +} + +static int fsl_audmix_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_audmix *priv = snd_soc_dai_get_drvdata(dai); + + /* Capture stream shall not be handled */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + priv->tdms |= BIT(dai->driver->id); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + priv->tdms &= ~BIT(dai->driver->id); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops fsl_audmix_dai_ops = { + .set_fmt = fsl_audmix_dai_set_fmt, + .trigger = fsl_audmix_dai_trigger, +}; + +static struct snd_soc_dai_driver fsl_audmix_dai[] = { + { + .id = 0, + .name = "audmix-0", + .playback = { + .stream_name = "AUDMIX-Playback-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .capture = { + .stream_name = "AUDMIX-Capture-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .ops = &fsl_audmix_dai_ops, + }, + { + .id = 1, + .name = "audmix-1", + .playback = { + .stream_name = "AUDMIX-Playback-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .capture = { + .stream_name = "AUDMIX-Capture-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AUDMIX_FORMATS, + }, + .ops = &fsl_audmix_dai_ops, + }, +}; + +static const struct snd_soc_component_driver fsl_audmix_component = { + .name = "fsl-audmix-dai", + .controls = fsl_audmix_snd_controls, + .num_controls = ARRAY_SIZE(fsl_audmix_snd_controls), +}; + +static bool fsl_audmix_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AUDMIX_CTR: + case FSL_AUDMIX_STR: + case FSL_AUDMIX_ATCR0: + case FSL_AUDMIX_ATIVAL0: + case FSL_AUDMIX_ATSTPUP0: + case FSL_AUDMIX_ATSTPDN0: + case FSL_AUDMIX_ATSTPTGT0: + case FSL_AUDMIX_ATTNVAL0: + case FSL_AUDMIX_ATSTP0: + case FSL_AUDMIX_ATCR1: + case FSL_AUDMIX_ATIVAL1: + case FSL_AUDMIX_ATSTPUP1: + case FSL_AUDMIX_ATSTPDN1: + case FSL_AUDMIX_ATSTPTGT1: + case FSL_AUDMIX_ATTNVAL1: + case FSL_AUDMIX_ATSTP1: + return true; + default: + return false; + } +} + +static bool fsl_audmix_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AUDMIX_CTR: + case FSL_AUDMIX_ATCR0: + case FSL_AUDMIX_ATIVAL0: + case FSL_AUDMIX_ATSTPUP0: + case FSL_AUDMIX_ATSTPDN0: + case FSL_AUDMIX_ATSTPTGT0: + case FSL_AUDMIX_ATCR1: + case FSL_AUDMIX_ATIVAL1: + case FSL_AUDMIX_ATSTPUP1: + case FSL_AUDMIX_ATSTPDN1: + case FSL_AUDMIX_ATSTPTGT1: + return true; + default: + return false; + } +} + +static const struct reg_default fsl_audmix_reg[] = { + { FSL_AUDMIX_CTR, 0x00060 }, + { FSL_AUDMIX_STR, 0x00003 }, + { FSL_AUDMIX_ATCR0, 0x00000 }, + { FSL_AUDMIX_ATIVAL0, 0x3FFFF }, + { FSL_AUDMIX_ATSTPUP0, 0x2AAAA }, + { FSL_AUDMIX_ATSTPDN0, 0x30000 }, + { FSL_AUDMIX_ATSTPTGT0, 0x00010 }, + { FSL_AUDMIX_ATTNVAL0, 0x00000 }, + { FSL_AUDMIX_ATSTP0, 0x00000 }, + { FSL_AUDMIX_ATCR1, 0x00000 }, + { FSL_AUDMIX_ATIVAL1, 0x3FFFF }, + { FSL_AUDMIX_ATSTPUP1, 0x2AAAA }, + { FSL_AUDMIX_ATSTPDN1, 0x30000 }, + { FSL_AUDMIX_ATSTPTGT1, 0x00010 }, + { FSL_AUDMIX_ATTNVAL1, 0x00000 }, + { FSL_AUDMIX_ATSTP1, 0x00000 }, +}; + +static const struct regmap_config fsl_audmix_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = FSL_AUDMIX_ATSTP1, + .reg_defaults = fsl_audmix_reg, + .num_reg_defaults = ARRAY_SIZE(fsl_audmix_reg), + .readable_reg = fsl_audmix_readable_reg, + .writeable_reg = fsl_audmix_writeable_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int fsl_audmix_probe(struct platform_device *pdev) +{ + struct fsl_audmix *priv; + struct resource *res; + void __iomem *regs; + int ret; + const char *sprop; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "ipg", regs, + &fsl_audmix_regmap_config); + if (IS_ERR(priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap\n"); + return PTR_ERR(priv->regmap); + } + + priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg"); + if (IS_ERR(priv->ipg_clk)) { + dev_err(&pdev->dev, "failed to get ipg clock\n"); + return PTR_ERR(priv->ipg_clk); + } + + platform_set_drvdata(pdev, priv); + pm_runtime_enable(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_audmix_component, + fsl_audmix_dai, + ARRAY_SIZE(fsl_audmix_dai)); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + return ret; + } + + sprop = of_get_property(pdev->dev.of_node, "model", NULL); + if (sprop) { + priv->pdev = platform_device_register_data(&pdev->dev, sprop, 0, + NULL, 0); + if (IS_ERR(priv->pdev)) { + ret = PTR_ERR(priv->pdev); + dev_err(&pdev->dev, + "failed to register platform %s: %d\n", sprop, + ret); + } + } else { + dev_err(&pdev->dev, "[model] attribute missing.\n"); + ret = -EINVAL; + } + + return ret; +} + +static int fsl_audmix_remove(struct platform_device *pdev) +{ + struct fsl_audmix *priv = dev_get_drvdata(&pdev->dev); + + if (priv->pdev) + platform_device_unregister(priv->pdev); + + return 0; +} + +#ifdef CONFIG_PM +static int fsl_audmix_runtime_resume(struct device *dev) +{ + struct fsl_audmix *priv = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(priv->ipg_clk); + if (ret) { + dev_err(dev, "Failed to enable IPG clock: %d\n", ret); + return ret; + } + + regcache_cache_only(priv->regmap, false); + regcache_mark_dirty(priv->regmap); + + return regcache_sync(priv->regmap); +} + +static int fsl_audmix_runtime_suspend(struct device *dev) +{ + struct fsl_audmix *priv = dev_get_drvdata(dev); + + regcache_cache_only(priv->regmap, true); + + clk_disable_unprepare(priv->ipg_clk); + + return 0; +} +#endif /* CONFIG_PM */ + +static const struct dev_pm_ops fsl_audmix_pm = { + SET_RUNTIME_PM_OPS(fsl_audmix_runtime_suspend, + fsl_audmix_runtime_resume, + NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +static const struct of_device_id fsl_audmix_ids[] = { + { .compatible = "fsl,imx8qm-audmix", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, fsl_audmix_ids); + +static struct platform_driver fsl_audmix_driver = { + .probe = fsl_audmix_probe, + .remove = fsl_audmix_remove, + .driver = { + .name = "fsl-audmix", + .of_match_table = fsl_audmix_ids, + .pm = &fsl_audmix_pm, + }, +}; +module_platform_driver(fsl_audmix_driver); + +MODULE_DESCRIPTION("NXP AUDMIX ASoC DAI driver"); +MODULE_AUTHOR("Viorel Suman "); +MODULE_ALIAS("platform:fsl-audmix"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_audmix.h b/sound/soc/fsl/fsl_audmix.h new file mode 100644 index 000000000000..7812ffec45c5 --- /dev/null +++ b/sound/soc/fsl/fsl_audmix.h @@ -0,0 +1,102 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * NXP AUDMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#ifndef __FSL_AUDMIX_H +#define __FSL_AUDMIX_H + +#define FSL_AUDMIX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) +/* AUDMIX Registers */ +#define FSL_AUDMIX_CTR 0x200 /* Control */ +#define FSL_AUDMIX_STR 0x204 /* Status */ + +#define FSL_AUDMIX_ATCR0 0x208 /* Attenuation Control */ +#define FSL_AUDMIX_ATIVAL0 0x20c /* Attenuation Initial Value */ +#define FSL_AUDMIX_ATSTPUP0 0x210 /* Attenuation step up factor */ +#define FSL_AUDMIX_ATSTPDN0 0x214 /* Attenuation step down factor */ +#define FSL_AUDMIX_ATSTPTGT0 0x218 /* Attenuation step target */ +#define FSL_AUDMIX_ATTNVAL0 0x21c /* Attenuation Value */ +#define FSL_AUDMIX_ATSTP0 0x220 /* Attenuation step number */ + +#define FSL_AUDMIX_ATCR1 0x228 /* Attenuation Control */ +#define FSL_AUDMIX_ATIVAL1 0x22c /* Attenuation Initial Value */ +#define FSL_AUDMIX_ATSTPUP1 0x230 /* Attenuation step up factor */ +#define FSL_AUDMIX_ATSTPDN1 0x234 /* Attenuation step down factor */ +#define FSL_AUDMIX_ATSTPTGT1 0x238 /* Attenuation step target */ +#define FSL_AUDMIX_ATTNVAL1 0x23c /* Attenuation Value */ +#define FSL_AUDMIX_ATSTP1 0x240 /* Attenuation step number */ + +/* AUDMIX Control Register */ +#define FSL_AUDMIX_CTR_MIXCLK_SHIFT 0 +#define FSL_AUDMIX_CTR_MIXCLK_MASK BIT(FSL_AUDMIX_CTR_MIXCLK_SHIFT) +#define FSL_AUDMIX_CTR_MIXCLK(i) ((i) << FSL_AUDMIX_CTR_MIXCLK_SHIFT) +#define FSL_AUDMIX_CTR_OUTSRC_SHIFT 1 +#define FSL_AUDMIX_CTR_OUTSRC_MASK (0x3 << FSL_AUDMIX_CTR_OUTSRC_SHIFT) +#define FSL_AUDMIX_CTR_OUTSRC(i) (((i) << FSL_AUDMIX_CTR_OUTSRC_SHIFT)\ + & FSL_AUDMIX_CTR_OUTSRC_MASK) +#define FSL_AUDMIX_CTR_OUTWIDTH_SHIFT 3 +#define FSL_AUDMIX_CTR_OUTWIDTH_MASK (0x7 << FSL_AUDMIX_CTR_OUTWIDTH_SHIFT) +#define FSL_AUDMIX_CTR_OUTWIDTH(i) (((i) << FSL_AUDMIX_CTR_OUTWIDTH_SHIFT)\ + & FSL_AUDMIX_CTR_OUTWIDTH_MASK) +#define FSL_AUDMIX_CTR_OUTCKPOL_SHIFT 6 +#define FSL_AUDMIX_CTR_OUTCKPOL_MASK BIT(FSL_AUDMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AUDMIX_CTR_OUTCKPOL(i) ((i) << FSL_AUDMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AUDMIX_CTR_MASKRTDF_SHIFT 7 +#define FSL_AUDMIX_CTR_MASKRTDF_MASK BIT(FSL_AUDMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AUDMIX_CTR_MASKRTDF(i) ((i) << FSL_AUDMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AUDMIX_CTR_MASKCKDF_SHIFT 8 +#define FSL_AUDMIX_CTR_MASKCKDF_MASK BIT(FSL_AUDMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AUDMIX_CTR_MASKCKDF(i) ((i) << FSL_AUDMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AUDMIX_CTR_SYNCMODE_SHIFT 9 +#define FSL_AUDMIX_CTR_SYNCMODE_MASK BIT(FSL_AUDMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AUDMIX_CTR_SYNCMODE(i) ((i) << FSL_AUDMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AUDMIX_CTR_SYNCSRC_SHIFT 10 +#define FSL_AUDMIX_CTR_SYNCSRC_MASK BIT(FSL_AUDMIX_CTR_SYNCSRC_SHIFT) +#define FSL_AUDMIX_CTR_SYNCSRC(i) ((i) << FSL_AUDMIX_CTR_SYNCSRC_SHIFT) + +/* AUDMIX Status Register */ +#define FSL_AUDMIX_STR_RATEDIFF BIT(0) +#define FSL_AUDMIX_STR_CLKDIFF BIT(1) +#define FSL_AUDMIX_STR_MIXSTAT_SHIFT 2 +#define FSL_AUDMIX_STR_MIXSTAT_MASK (0x3 << FSL_AUDMIX_STR_MIXSTAT_SHIFT) +#define FSL_AUDMIX_STR_MIXSTAT(i) (((i) & FSL_AUDMIX_STR_MIXSTAT_MASK) \ + >> FSL_AUDMIX_STR_MIXSTAT_SHIFT) +/* AUDMIX Attenuation Control Register */ +#define FSL_AUDMIX_ATCR_AT_EN BIT(0) +#define FSL_AUDMIX_ATCR_AT_UPDN BIT(1) +#define FSL_AUDMIX_ATCR_ATSTPDIF_SHIFT 2 +#define FSL_AUDMIX_ATCR_ATSTPDFI_MASK \ + (0xfff << FSL_AUDMIX_ATCR_ATSTPDIF_SHIFT) + +/* AUDMIX Attenuation Initial Value Register */ +#define FSL_AUDMIX_ATIVAL_ATINVAL_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Up Factor Register */ +#define FSL_AUDMIX_ATSTPUP_ATSTEPUP_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Down Factor Register */ +#define FSL_AUDMIX_ATSTPDN_ATSTEPDN_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Target Register */ +#define FSL_AUDMIX_ATSTPTGT_ATSTPTG_MASK 0x3FFFF + +/* AUDMIX Attenuation Value Register */ +#define FSL_AUDMIX_ATTNVAL_ATCURVAL_MASK 0x3FFFF + +/* AUDMIX Attenuation Step Number Register */ +#define FSL_AUDMIX_ATSTP_STPCTR_MASK 0x3FFFF + +#define FSL_AUDMIX_MAX_DAIS 2 +struct fsl_audmix { + struct platform_device *pdev; + struct regmap *regmap; + struct clk *ipg_clk; + u8 tdms; +}; + +#endif /* __FSL_AUDMIX_H */ -- cgit v1.2.3-59-g8ed1b From b86ef5367761603df5f66ce08fb014b991f6b51d Mon Sep 17 00:00:00 2001 From: Viorel Suman Date: Tue, 22 Jan 2019 11:14:30 +0000 Subject: ASoC: fsl: Add Audio Mixer machine driver This patch implements Audio Mixer machine driver for NXP iMX8 SOCs. It connects together Audio Mixer and related SAI instances. Signed-off-by: Viorel Suman Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 9 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-audmix.c | 327 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 338 insertions(+) create mode 100644 sound/soc/fsl/imx-audmix.c (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 0af2e056d211..d87c842806bd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -303,6 +303,15 @@ config SND_SOC_FSL_ASOC_CARD CS4271, CS4272 and SGTL5000. Say Y if you want to add support for Freescale Generic ASoC Sound Card. +config SND_SOC_IMX_AUDMIX + tristate "SoC Audio support for i.MX boards with AUDMIX" + select SND_SOC_FSL_AUDMIX + select SND_SOC_FSL_SAI + help + SoC Audio support for i.MX boards with Audio Mixer + Say Y if you want to add support for SoC audio on an i.MX board with + an Audio Mixer. + endif # SND_IMX_SOC endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 4172d5a3e36c..c0dd04422fe9 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -62,6 +62,7 @@ snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o +snd-soc-imx-audmix-objs := imx-audmix.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o @@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o +obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c new file mode 100644 index 000000000000..72e37ca6cfbb --- /dev/null +++ b/sound/soc/fsl/imx-audmix.c @@ -0,0 +1,327 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright 2017 NXP + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include +#include +#include +#include +#include +#include +#include "fsl_sai.h" +#include "fsl_audmix.h" + +struct imx_audmix { + struct platform_device *pdev; + struct snd_soc_card card; + struct platform_device *audmix_pdev; + struct platform_device *out_pdev; + struct clk *cpu_mclk; + int num_dai; + struct snd_soc_dai_link *dai; + int num_dai_conf; + struct snd_soc_codec_conf *dai_conf; + int num_dapm_routes; + struct snd_soc_dapm_route *dapm_routes; +}; + +static const u32 imx_audmix_rates[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 64000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list imx_audmix_rate_constraints = { + .count = ARRAY_SIZE(imx_audmix_rates), + .list = imx_audmix_rates, +}; + +static int imx_audmix_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_audmix *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = rtd->card->dev; + unsigned long clk_rate = clk_get_rate(priv->cpu_mclk); + int ret; + + if (clk_rate % 24576000 == 0) { + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &imx_audmix_rate_constraints); + if (ret < 0) + return ret; + } else { + dev_warn(dev, "mclk may be not supported %lu\n", clk_rate); + } + + ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, + 1, 8); + if (ret < 0) + return ret; + + return snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT, + FSL_AUDMIX_FORMATS); +} + +static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; + u32 channels = params_channels(params); + int ret, dir; + + /* For playback the AUDMIX is slave, and for record is master */ + fmt |= tx ? SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBM_CFM; + dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; + + /* set DAI configuration */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret) { + dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir); + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + /* + * Per datasheet, AUDMIX expects 8 slots and 32 bits + * for every slot in TDM mode. + */ + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1, + BIT(channels) - 1, 8, 32); + if (ret) + dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret); + + return ret; +} + +static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; + int ret; + + if (!tx) + return 0; + + /* For playback the AUDMIX is slave */ + fmt |= SND_SOC_DAIFMT_CBM_CFM; + + /* set AUDMIX DAI configuration */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret) + dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops imx_audmix_fe_ops = { + .startup = imx_audmix_fe_startup, + .hw_params = imx_audmix_fe_hw_params, +}; + +static struct snd_soc_ops imx_audmix_be_ops = { + .hw_params = imx_audmix_be_hw_params, +}; + +static int imx_audmix_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *audmix_np = NULL, *out_cpu_np = NULL; + struct platform_device *audmix_pdev = NULL; + struct platform_device *cpu_pdev; + struct of_phandle_args args; + struct imx_audmix *priv; + int i, num_dai, ret; + const char *fe_name_pref = "HiFi-AUDMIX-FE-"; + char *be_name, *be_pb, *be_cp, *dai_name, *capture_dai_name; + + if (pdev->dev.parent) { + audmix_np = pdev->dev.parent->of_node; + } else { + dev_err(&pdev->dev, "Missing parent device.\n"); + return -EINVAL; + } + + if (!audmix_np) { + dev_err(&pdev->dev, "Missign DT node for parent device.\n"); + return -EINVAL; + } + + audmix_pdev = of_find_device_by_node(audmix_np); + if (!audmix_pdev) { + dev_err(&pdev->dev, "Missing AUDMIX platform device for %s\n", + np->full_name); + return -EINVAL; + } + + num_dai = of_count_phandle_with_args(audmix_np, "dais", NULL); + if (num_dai != FSL_AUDMIX_MAX_DAIS) { + dev_err(&pdev->dev, "Need 2 dais to be provided for %s\n", + audmix_np->full_name); + return -EINVAL; + } + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->num_dai = 2 * num_dai; + priv->dai = devm_kzalloc(&pdev->dev, priv->num_dai * + sizeof(struct snd_soc_dai_link), GFP_KERNEL); + if (!priv->dai) + return -ENOMEM; + + priv->num_dai_conf = num_dai; + priv->dai_conf = devm_kzalloc(&pdev->dev, priv->num_dai_conf * + sizeof(struct snd_soc_codec_conf), + GFP_KERNEL); + if (!priv->dai_conf) + return -ENOMEM; + + priv->num_dapm_routes = 3 * num_dai; + priv->dapm_routes = devm_kzalloc(&pdev->dev, priv->num_dapm_routes * + sizeof(struct snd_soc_dapm_route), + GFP_KERNEL); + if (!priv->dapm_routes) + return -ENOMEM; + + for (i = 0; i < num_dai; i++) { + ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i, + &args); + if (ret < 0) { + dev_err(&pdev->dev, "of_parse_phandle_with_args failed\n"); + return ret; + } + + cpu_pdev = of_find_device_by_node(args.np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + return -EINVAL; + } + + dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s", + fe_name_pref, args.np->full_name + 1); + + dev_info(pdev->dev.parent, "DAI FE name:%s\n", dai_name); + + if (i == 0) { + out_cpu_np = args.np; + capture_dai_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + dai_name, "CPU-Capture"); + } + + priv->dai[i].name = dai_name; + priv->dai[i].stream_name = "HiFi-AUDMIX-FE"; + priv->dai[i].codec_dai_name = "snd-soc-dummy-dai"; + priv->dai[i].codec_name = "snd-soc-dummy"; + priv->dai[i].cpu_of_node = args.np; + priv->dai[i].cpu_dai_name = dev_name(&cpu_pdev->dev); + priv->dai[i].platform_of_node = args.np; + priv->dai[i].dynamic = 1; + priv->dai[i].dpcm_playback = 1; + priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0); + priv->dai[i].ignore_pmdown_time = 1; + priv->dai[i].ops = &imx_audmix_fe_ops; + + /* Add AUDMIX Backend */ + be_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "audmix-%d", i); + be_pb = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "AUDMIX-Playback-%d", i); + be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "AUDMIX-Capture-%d", i); + + priv->dai[num_dai + i].name = be_name; + priv->dai[num_dai + i].codec_dai_name = "snd-soc-dummy-dai"; + priv->dai[num_dai + i].codec_name = "snd-soc-dummy"; + priv->dai[num_dai + i].cpu_of_node = audmix_np; + priv->dai[num_dai + i].cpu_dai_name = be_name; + priv->dai[num_dai + i].platform_name = "snd-soc-dummy"; + priv->dai[num_dai + i].no_pcm = 1; + priv->dai[num_dai + i].dpcm_playback = 1; + priv->dai[num_dai + i].dpcm_capture = 1; + priv->dai[num_dai + i].ignore_pmdown_time = 1; + priv->dai[num_dai + i].ops = &imx_audmix_be_ops; + + priv->dai_conf[i].of_node = args.np; + priv->dai_conf[i].name_prefix = dai_name; + + priv->dapm_routes[i].source = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + dai_name, "CPU-Playback"); + priv->dapm_routes[i].sink = be_pb; + priv->dapm_routes[num_dai + i].source = be_pb; + priv->dapm_routes[num_dai + i].sink = be_cp; + priv->dapm_routes[2 * num_dai + i].source = be_cp; + priv->dapm_routes[2 * num_dai + i].sink = capture_dai_name; + } + + cpu_pdev = of_find_device_by_node(out_cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + return -EINVAL; + } + priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1"); + if (IS_ERR(priv->cpu_mclk)) { + ret = PTR_ERR(priv->cpu_mclk); + dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret); + return -EINVAL; + } + + priv->audmix_pdev = audmix_pdev; + priv->out_pdev = cpu_pdev; + + priv->card.dai_link = priv->dai; + priv->card.num_links = priv->num_dai; + priv->card.codec_conf = priv->dai_conf; + priv->card.num_configs = priv->num_dai_conf; + priv->card.dapm_routes = priv->dapm_routes; + priv->card.num_dapm_routes = priv->num_dapm_routes; + priv->card.dev = pdev->dev.parent; + priv->card.owner = THIS_MODULE; + priv->card.name = "imx-audmix"; + + platform_set_drvdata(pdev, &priv->card); + snd_soc_card_set_drvdata(&priv->card, priv); + + ret = devm_snd_soc_register_card(pdev->dev.parent, &priv->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + return ret; + } + + return ret; +} + +static struct platform_driver imx_audmix_driver = { + .probe = imx_audmix_probe, + .driver = { + .name = "imx-audmix", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(imx_audmix_driver); + +MODULE_DESCRIPTION("NXP AUDMIX ASoC machine driver"); +MODULE_AUTHOR("Viorel Suman "); +MODULE_ALIAS("platform:imx-audmix"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-59-g8ed1b From e42dd3ee3f9c6007c569386a8477a19d3e7503f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Mar 2019 17:32:02 +0100 Subject: ALSA: emux: Add support of loading GUS-patch It's a feature request for the ancient sutff, but it's still valid; the loading of a GUS-patch isn't available via hwdep device although it's supported over OSS sequencer. The only missing piece is the call of snd_soundfont_load_guspatch() in synth emux hwdep code. Signed-off-by: Takashi Iwai --- sound/synth/emux/emux_hwdep.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index d9fcae071b47..fae48d108b97 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -39,6 +39,11 @@ snd_emux_hwdep_load_patch(struct snd_emux *emu, void __user *arg) if (copy_from_user(&patch, arg, sizeof(patch))) return -EFAULT; + if (patch.key == GUS_PATCH) + return snd_soundfont_load_guspatch(emu->sflist, arg, + patch.len + sizeof(patch), + TMP_CLIENT_ID); + if (patch.type >= SNDRV_SFNT_LOAD_INFO && patch.type <= SNDRV_SFNT_PROBE_DATA) { err = snd_soundfont_load(emu->sflist, arg, patch.len + sizeof(patch), TMP_CLIENT_ID); -- cgit v1.2.3-59-g8ed1b From 5fb94d46c3179cb7afd805a5a9252111316a926f Mon Sep 17 00:00:00 2001 From: Viorel Suman Date: Wed, 27 Mar 2019 09:29:38 +0000 Subject: ASoC: fsl_audmix: Fix kbuild failure The format in dev_dbg function must be a constant. Signed-off-by: Viorel Suman Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_audmix.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 3356cb617713..dabde0342c95 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -90,7 +90,7 @@ static int fsl_audmix_state_trans(struct snd_soc_component *comp, struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); /* Enforce all required TDMs are started */ if ((priv->tdms & prm.tdms) != prm.tdms) { - dev_dbg(comp->dev, prm.msg); + dev_dbg(comp->dev, "%s", prm.msg); return -EINVAL; } -- cgit v1.2.3-59-g8ed1b From 8748b850beccdbc87aa8776d63abd6b5628720c8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Mar 2019 16:42:51 +0100 Subject: ALSA: timer: Unify timer callback process code The timer core has two almost identical code for processing callbacks: once in snd_timer_interrupt() for fast callbacks and another in snd_timer_tasklet() for delayed callbacks. Let's unify them. In the new version, the resolution is read from ti->resolution at each call, and this must be fine; ti->resolution is set in the preparation step in snd_timer_interrupt(). Signed-off-by: Takashi Iwai --- sound/core/timer.c | 62 +++++++++++++++++++++++------------------------------- 1 file changed, 26 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index 61a0cec6e1f6..fdcddfb756b4 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -720,29 +720,19 @@ static void snd_timer_reschedule(struct snd_timer * timer, unsigned long ticks_l timer->sticks = ticks; } -/* - * timer tasklet - * - */ -static void snd_timer_tasklet(unsigned long arg) +/* call callbacks in timer ack list */ +static void snd_timer_process_callbacks(struct snd_timer *timer, + struct list_head *head) { - struct snd_timer *timer = (struct snd_timer *) arg; struct snd_timer_instance *ti; - struct list_head *p; unsigned long resolution, ticks; - unsigned long flags; - if (timer->card && timer->card->shutdown) - return; - - spin_lock_irqsave(&timer->lock, flags); - /* now process all callbacks */ - while (!list_empty(&timer->sack_list_head)) { - p = timer->sack_list_head.next; /* get first item */ - ti = list_entry(p, struct snd_timer_instance, ack_list); + while (!list_empty(head)) { + ti = list_first_entry(head, struct snd_timer_instance, + ack_list); /* remove from ack_list and make empty */ - list_del_init(p); + list_del_init(&ti->ack_list); ticks = ti->pticks; ti->pticks = 0; @@ -755,6 +745,22 @@ static void snd_timer_tasklet(unsigned long arg) spin_lock(&timer->lock); ti->flags &= ~SNDRV_TIMER_IFLG_CALLBACK; } +} + +/* + * timer tasklet + * + */ +static void snd_timer_tasklet(unsigned long arg) +{ + struct snd_timer *timer = (struct snd_timer *) arg; + unsigned long flags; + + if (timer->card && timer->card->shutdown) + return; + + spin_lock_irqsave(&timer->lock, flags); + snd_timer_process_callbacks(timer, &timer->sack_list_head); spin_unlock_irqrestore(&timer->lock, flags); } @@ -767,8 +773,8 @@ static void snd_timer_tasklet(unsigned long arg) void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) { struct snd_timer_instance *ti, *ts, *tmp; - unsigned long resolution, ticks; - struct list_head *p, *ack_list_head; + unsigned long resolution; + struct list_head *ack_list_head; unsigned long flags; int use_tasklet = 0; @@ -839,23 +845,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) } /* now process all fast callbacks */ - while (!list_empty(&timer->ack_list_head)) { - p = timer->ack_list_head.next; /* get first item */ - ti = list_entry(p, struct snd_timer_instance, ack_list); - - /* remove from ack_list and make empty */ - list_del_init(p); - - ticks = ti->pticks; - ti->pticks = 0; - - ti->flags |= SNDRV_TIMER_IFLG_CALLBACK; - spin_unlock(&timer->lock); - if (ti->callback) - ti->callback(ti, resolution, ticks); - spin_lock(&timer->lock); - ti->flags &= ~SNDRV_TIMER_IFLG_CALLBACK; - } + snd_timer_process_callbacks(timer, &timer->ack_list_head); /* do we have any slow callbacks? */ use_tasklet = !list_empty(&timer->sack_list_head); -- cgit v1.2.3-59-g8ed1b From 7bb4a8a2cc9382da720b46988bc976ebccaa49fd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Mar 2019 16:51:58 +0100 Subject: ALSA: timer: Make sure to clear pending ack list When a card is under disconnection, we bail out immediately at each timer interrupt or tasklet. This might leave some items left in ack list. For a better integration of the upcoming change to check ack_list emptiness, clear out the whole list upon the emergency exit route. Signed-off-by: Takashi Iwai --- sound/core/timer.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index fdcddfb756b4..107d8ebeeb2e 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -747,6 +747,18 @@ static void snd_timer_process_callbacks(struct snd_timer *timer, } } +/* clear pending instances from ack list */ +static void snd_timer_clear_callbacks(struct snd_timer *timer, + struct list_head *head) +{ + unsigned long flags; + + spin_lock_irqsave(&timer->lock, flags); + while (!list_empty(head)) + list_del_init(head->next); + spin_unlock_irqrestore(&timer->lock, flags); +} + /* * timer tasklet * @@ -756,8 +768,10 @@ static void snd_timer_tasklet(unsigned long arg) struct snd_timer *timer = (struct snd_timer *) arg; unsigned long flags; - if (timer->card && timer->card->shutdown) + if (timer->card && timer->card->shutdown) { + snd_timer_clear_callbacks(timer, &timer->sack_list_head); return; + } spin_lock_irqsave(&timer->lock, flags); snd_timer_process_callbacks(timer, &timer->sack_list_head); @@ -781,8 +795,10 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) if (timer == NULL) return; - if (timer->card && timer->card->shutdown) + if (timer->card && timer->card->shutdown) { + snd_timer_clear_callbacks(timer, &timer->ack_list_head); return; + } spin_lock_irqsave(&timer->lock, flags); -- cgit v1.2.3-59-g8ed1b From a7588c896b05444929ecb3d0115481988720abf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Mar 2019 16:56:08 +0100 Subject: ALSA: timer: Check ack_list emptiness instead of bit flag For checking the pending timer instance that is still left on the timer object that is being closed, we set/clear a bit flag SNDRV_TIMER_IFLG_CALLBACK around the call of callbacks. This can be simplified by replace with the list_empty() call for ti->ack_list. This covers the existence more comprehensively and safely. A gratis bonus is that we can get rid of SNDRV_TIMER_IFLG_CALLBACK bit flag definition as well. Signed-off-by: Takashi Iwai --- include/sound/timer.h | 1 - sound/core/timer.c | 10 ++++------ 2 files changed, 4 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/include/sound/timer.h b/include/sound/timer.h index 7ae226ab6990..bcfee20ea226 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -43,7 +43,6 @@ #define SNDRV_TIMER_IFLG_START 0x00000004 #define SNDRV_TIMER_IFLG_AUTO 0x00000008 /* auto restart */ #define SNDRV_TIMER_IFLG_FAST 0x00000010 /* fast callback (do not use tasklet) */ -#define SNDRV_TIMER_IFLG_CALLBACK 0x00000020 /* timer callback is active */ #define SNDRV_TIMER_IFLG_EXCLUSIVE 0x00000040 /* exclusive owner - no more instances */ #define SNDRV_TIMER_IFLG_EARLY_EVENT 0x00000080 /* write early event to the poll queue */ diff --git a/sound/core/timer.c b/sound/core/timer.c index 107d8ebeeb2e..e343de0e4f9e 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -366,7 +366,7 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri) timer->num_instances--; /* wait, until the active callback is finished */ spin_lock_irq(&timer->lock); - while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) { + while (!list_empty(&timeri->ack_list)) { spin_unlock_irq(&timer->lock); udelay(10); spin_lock_irq(&timer->lock); @@ -731,19 +731,17 @@ static void snd_timer_process_callbacks(struct snd_timer *timer, ti = list_first_entry(head, struct snd_timer_instance, ack_list); - /* remove from ack_list and make empty */ - list_del_init(&ti->ack_list); - ticks = ti->pticks; ti->pticks = 0; resolution = ti->resolution; - ti->flags |= SNDRV_TIMER_IFLG_CALLBACK; spin_unlock(&timer->lock); if (ti->callback) ti->callback(ti, resolution, ticks); spin_lock(&timer->lock); - ti->flags &= ~SNDRV_TIMER_IFLG_CALLBACK; + + /* remove from ack_list and make empty */ + list_del_init(&ti->ack_list); } } -- cgit v1.2.3-59-g8ed1b From fe1b26c93d430400ac37d820425e2468218ae8b2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Mar 2019 17:02:40 +0100 Subject: ALSA: timer: Make snd_timer_close() really kill pending actions snd_timer_close() is supposed to close the timer instance and sync with the deactivation of pending actions. However, there are still some overlooked cases: - It calls snd_timer_stop() at the beginning, but some other might re-trigger the timer right after that. - snd_timer_stop() calls del_timer_sync() only when all belonging instances are closed. If multiple instances were assigned to a timer object and one is closed, the timer is still running. Then the pending action assigned to this timer might be left. Actually either of the above is the likely cause of the reported syzkaller UAF. This patch plug these holes by introducing SNDRV_TIMER_IFLG_DEAD flag. This is set at the beginning of snd_timer_close(), and the flag is checked at snd_timer_start*() and else, so that no longer new action is left after snd_timer_close(). Reported-by: syzbot+d5136d4d3240cbe45a2a@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai --- sound/core/timer.c | 45 +++++++++++++++++++++++++++++++++------------ 1 file changed, 33 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index e343de0e4f9e..bb7e90ab90f8 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -38,6 +38,7 @@ /* internal flags */ #define SNDRV_TIMER_IFLG_PAUSED 0x00010000 +#define SNDRV_TIMER_IFLG_DEAD 0x00020000 #if IS_ENABLED(CONFIG_SND_HRTIMER) #define DEFAULT_TIMER_LIMIT 4 @@ -353,15 +354,20 @@ EXPORT_SYMBOL(snd_timer_open); */ static int snd_timer_close_locked(struct snd_timer_instance *timeri) { - struct snd_timer *timer = NULL; + struct snd_timer *timer = timeri->timer; struct snd_timer_instance *slave, *tmp; + if (timer) { + spin_lock_irq(&timer->lock); + timeri->flags |= SNDRV_TIMER_IFLG_DEAD; + spin_unlock_irq(&timer->lock); + } + list_del(&timeri->open_list); /* force to stop the timer */ snd_timer_stop(timeri); - timer = timeri->timer; if (timer) { timer->num_instances--; /* wait, until the active callback is finished */ @@ -497,6 +503,10 @@ static int snd_timer_start1(struct snd_timer_instance *timeri, return -EINVAL; spin_lock_irqsave(&timer->lock, flags); + if (timeri->flags & SNDRV_TIMER_IFLG_DEAD) { + result = -EINVAL; + goto unlock; + } if (timer->card && timer->card->shutdown) { result = -ENODEV; goto unlock; @@ -541,11 +551,16 @@ static int snd_timer_start_slave(struct snd_timer_instance *timeri, bool start) { unsigned long flags; + int err; spin_lock_irqsave(&slave_active_lock, flags); + if (timeri->flags & SNDRV_TIMER_IFLG_DEAD) { + err = -EINVAL; + goto unlock; + } if (timeri->flags & SNDRV_TIMER_IFLG_RUNNING) { - spin_unlock_irqrestore(&slave_active_lock, flags); - return -EBUSY; + err = -EBUSY; + goto unlock; } timeri->flags |= SNDRV_TIMER_IFLG_RUNNING; if (timeri->master && timeri->timer) { @@ -556,8 +571,10 @@ static int snd_timer_start_slave(struct snd_timer_instance *timeri, SNDRV_TIMER_EVENT_CONTINUE); spin_unlock(&timeri->timer->lock); } + err = 1; /* delayed start */ + unlock: spin_unlock_irqrestore(&slave_active_lock, flags); - return 1; /* delayed start */ + return err; } /* stop/pause a master timer */ @@ -731,14 +748,16 @@ static void snd_timer_process_callbacks(struct snd_timer *timer, ti = list_first_entry(head, struct snd_timer_instance, ack_list); - ticks = ti->pticks; - ti->pticks = 0; - resolution = ti->resolution; + if (!(ti->flags & SNDRV_TIMER_IFLG_DEAD)) { + ticks = ti->pticks; + ti->pticks = 0; + resolution = ti->resolution; - spin_unlock(&timer->lock); - if (ti->callback) - ti->callback(ti, resolution, ticks); - spin_lock(&timer->lock); + spin_unlock(&timer->lock); + if (ti->callback) + ti->callback(ti, resolution, ticks); + spin_lock(&timer->lock); + } /* remove from ack_list and make empty */ list_del_init(&ti->ack_list); @@ -810,6 +829,8 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) */ list_for_each_entry_safe(ti, tmp, &timer->active_list_head, active_list) { + if (ti->flags & SNDRV_TIMER_IFLG_DEAD) + continue; if (!(ti->flags & SNDRV_TIMER_IFLG_RUNNING)) continue; ti->pticks += ticks_left; -- cgit v1.2.3-59-g8ed1b From 734b5a0bbdf43518e6739c8156a985e385e557fe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Nov 2018 19:38:13 +0100 Subject: ALSA: Replace snd_malloc_pages() and snd_free_pages() with standard helpers, take#2 snd_malloc_pages() and snd_free_pages() are merely thin wrappers of the standard page allocator / free functions. Even the arguments are compatible with some standard helpers, so there is little merit of keeping these wrappers. This patch replaces the all existing callers of snd_malloc_pages() and snd_free_pages() with the direct calls of the standard helper functions. In this version, we use a recently introduced one, alloc_pages_exact(), which suits better than the old snd_malloc_pages() implementation for our purposes. Then we can avoid the waste of pages by alignment to power-of-two. Since alloc_pages_exact() does split pages, we need no longer __GFP_COMP flag; or better to say, we must not pass __GFP_COMP to alloc_pages_exact(). So the former unconditional addition of __GFP_COMP flag in snd_malloc_pages() is dropped, as well as in most other places. Reviewed-by: Takashi Sakamoto Acked-by: Michal Hocko Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 4 ---- sound/core/memalloc.c | 53 +++-------------------------------------- sound/core/pcm.c | 14 +++++------ sound/usb/usx2y/usX2Yhwdep.c | 3 ++- sound/usb/usx2y/usbusx2y.c | 3 ++- sound/usb/usx2y/usx2yhwdeppcm.c | 6 +++-- 6 files changed, 18 insertions(+), 65 deletions(-) (limited to 'sound') diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 1ac0dd82a916..4c6f3b5a7cff 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -151,9 +151,5 @@ int snd_dma_alloc_pages_fallback(int type, struct device *dev, size_t size, struct snd_dma_buffer *dmab); void snd_dma_free_pages(struct snd_dma_buffer *dmab); -/* basic memory allocation functions */ -void *snd_malloc_pages(size_t size, gfp_t gfp_flags); -void snd_free_pages(void *ptr, size_t size); - #endif /* __SOUND_MEMALLOC_H */ diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index eb974235c92b..9f48e1d3a257 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,53 +30,6 @@ #endif #include -/* - * - * Generic memory allocators - * - */ - -/** - * snd_malloc_pages - allocate pages with the given size - * @size: the size to allocate in bytes - * @gfp_flags: the allocation conditions, GFP_XXX - * - * Allocates the physically contiguous pages with the given size. - * - * Return: The pointer of the buffer, or %NULL if no enough memory. - */ -void *snd_malloc_pages(size_t size, gfp_t gfp_flags) -{ - int pg; - - if (WARN_ON(!size)) - return NULL; - if (WARN_ON(!gfp_flags)) - return NULL; - gfp_flags |= __GFP_COMP; /* compound page lets parts be mapped */ - pg = get_order(size); - return (void *) __get_free_pages(gfp_flags, pg); -} -EXPORT_SYMBOL(snd_malloc_pages); - -/** - * snd_free_pages - release the pages - * @ptr: the buffer pointer to release - * @size: the allocated buffer size - * - * Releases the buffer allocated via snd_malloc_pages(). - */ -void snd_free_pages(void *ptr, size_t size) -{ - int pg; - - if (ptr == NULL) - return; - pg = get_order(size); - free_pages((unsigned long) ptr, pg); -} -EXPORT_SYMBOL(snd_free_pages); - /* * * Bus-specific memory allocators @@ -190,8 +143,8 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->bytes = 0; switch (type) { case SNDRV_DMA_TYPE_CONTINUOUS: - dmab->area = snd_malloc_pages(size, - (__force gfp_t)(unsigned long)device); + dmab->area = alloc_pages_exact(size, + (__force gfp_t)(unsigned long)device); dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA @@ -275,7 +228,7 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) { switch (dmab->dev.type) { case SNDRV_DMA_TYPE_CONTINUOUS: - snd_free_pages(dmab->area, dmab->bytes); + free_pages_exact(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA #ifdef CONFIG_GENERIC_ALLOCATOR diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 7b63aee124af..998e477522fd 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -959,22 +959,22 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, return -ENOMEM; size = PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status)); - runtime->status = snd_malloc_pages(size, GFP_KERNEL); + runtime->status = alloc_pages_exact(size, GFP_KERNEL); if (runtime->status == NULL) { kfree(runtime); return -ENOMEM; } - memset((void*)runtime->status, 0, size); + memset(runtime->status, 0, size); size = PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)); - runtime->control = snd_malloc_pages(size, GFP_KERNEL); + runtime->control = alloc_pages_exact(size, GFP_KERNEL); if (runtime->control == NULL) { - snd_free_pages((void*)runtime->status, + free_pages_exact(runtime->status, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status))); kfree(runtime); return -ENOMEM; } - memset((void*)runtime->control, 0, size); + memset(runtime->control, 0, size); init_waitqueue_head(&runtime->sleep); init_waitqueue_head(&runtime->tsleep); @@ -1000,9 +1000,9 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) runtime = substream->runtime; if (runtime->private_free != NULL) runtime->private_free(runtime); - snd_free_pages((void*)runtime->status, + free_pages_exact(runtime->status, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_status))); - snd_free_pages((void*)runtime->control, + free_pages_exact(runtime->control, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); /* Avoid concurrent access to runtime via PCM timer interface */ diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index c1dd9a7b48df..bfe1108416cf 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -75,7 +75,8 @@ static int snd_us428ctls_mmap(struct snd_hwdep * hw, struct file *filp, struct v if (!us428->us428ctls_sharedmem) { init_waitqueue_head(&us428->us428ctls_wait_queue_head); - if(!(us428->us428ctls_sharedmem = snd_malloc_pages(sizeof(struct us428ctls_sharedmem), GFP_KERNEL))) + us428->us428ctls_sharedmem = alloc_pages_exact(sizeof(struct us428ctls_sharedmem), GFP_KERNEL); + if (!us428->us428ctls_sharedmem) return -ENOMEM; memset(us428->us428ctls_sharedmem, -1, sizeof(struct us428ctls_sharedmem)); us428->us428ctls_sharedmem->CtlSnapShotLast = -2; diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index da4a5a541512..9f7bbed2c0f0 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -437,7 +437,8 @@ static void snd_usX2Y_card_private_free(struct snd_card *card) kfree(usX2Y(card)->In04Buf); usb_free_urb(usX2Y(card)->In04urb); if (usX2Y(card)->us428ctls_sharedmem) - snd_free_pages(usX2Y(card)->us428ctls_sharedmem, sizeof(*usX2Y(card)->us428ctls_sharedmem)); + free_pages_exact(usX2Y(card)->us428ctls_sharedmem, + sizeof(*usX2Y(card)->us428ctls_sharedmem)); if (usX2Y(card)->card_index >= 0 && usX2Y(card)->card_index < SNDRV_CARDS) snd_usX2Y_card_used[usX2Y(card)->card_index] = 0; } diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 714cf50d4a4c..ace8185c3f6d 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -488,7 +488,9 @@ static int snd_usX2Y_usbpcm_prepare(struct snd_pcm_substream *substream) snd_printdd("snd_usX2Y_pcm_prepare(%p)\n", substream); if (NULL == usX2Y->hwdep_pcm_shm) { - if (NULL == (usX2Y->hwdep_pcm_shm = snd_malloc_pages(sizeof(struct snd_usX2Y_hwdep_pcm_shm), GFP_KERNEL))) + usX2Y->hwdep_pcm_shm = alloc_pages_exact(sizeof(struct snd_usX2Y_hwdep_pcm_shm), + GFP_KERNEL); + if (!usX2Y->hwdep_pcm_shm) return -ENOMEM; memset(usX2Y->hwdep_pcm_shm, 0, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); } @@ -700,7 +702,7 @@ static void snd_usX2Y_hwdep_pcm_private_free(struct snd_hwdep *hwdep) { struct usX2Ydev *usX2Y = hwdep->private_data; if (NULL != usX2Y->hwdep_pcm_shm) - snd_free_pages(usX2Y->hwdep_pcm_shm, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); + free_pages_exact(usX2Y->hwdep_pcm_shm, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); } -- cgit v1.2.3-59-g8ed1b From 36b8defc447627c3e91058c43941ec15d827556e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 16 Mar 2019 08:48:52 +0100 Subject: ALSA: us122l: Use alloc_pages_exact() alloc_pages_exact() is more suitable choice for allocating the sound buffers, as it doesn't need to align with power-of-two. Along with the conversion, we can drop __GFP_COMP as well. The patch also replace the error messages to be more explicit. Acked-by: Michal Hocko Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usb_stream.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 221adf68bd0c..51d73111263a 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -155,9 +155,9 @@ void usb_stream_free(struct usb_stream_kernel *sk) if (!s) return; - free_pages((unsigned long)sk->write_page, get_order(s->write_size)); + free_pages_exact(sk->write_page, s->write_size); sk->write_page = NULL; - free_pages((unsigned long)s, get_order(s->read_size)); + free_pages_exact(s, s->read_size); sk->s = NULL; } @@ -172,7 +172,6 @@ struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk, int read_size = sizeof(struct usb_stream); int write_size; int usb_frames = dev->speed == USB_SPEED_HIGH ? 8000 : 1000; - int pg; in_pipe = usb_rcvisocpipe(dev, in_endpoint); out_pipe = usb_sndisocpipe(dev, out_endpoint); @@ -202,11 +201,10 @@ struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk, goto out; } - pg = get_order(read_size); - sk->s = (void *) __get_free_pages(GFP_KERNEL|__GFP_COMP|__GFP_ZERO| - __GFP_NOWARN, pg); + sk->s = alloc_pages_exact(read_size, + GFP_KERNEL | __GFP_ZERO | __GFP_NOWARN); if (!sk->s) { - snd_printk(KERN_WARNING "couldn't __get_free_pages()\n"); + pr_warn("us122l: couldn't allocate read buffer\n"); goto out; } sk->s->cfg.version = USB_STREAM_INTERFACE_VERSION; @@ -221,13 +219,11 @@ struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk, sk->s->period_size = frame_size * period_frames; sk->s->write_size = write_size; - pg = get_order(write_size); - sk->write_page = - (void *)__get_free_pages(GFP_KERNEL|__GFP_COMP|__GFP_ZERO| - __GFP_NOWARN, pg); + sk->write_page = alloc_pages_exact(write_size, + GFP_KERNEL | __GFP_ZERO | __GFP_NOWARN); if (!sk->write_page) { - snd_printk(KERN_WARNING "couldn't __get_free_pages()\n"); + pr_warn("us122l: couldn't allocate write buffer\n"); usb_stream_free(sk); return NULL; } -- cgit v1.2.3-59-g8ed1b From ad6bb3067c4d7c684488eb9f8f8a32f623b12af3 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Fri, 29 Mar 2019 16:37:37 +0100 Subject: ASoC: cs42l51: add multi endpoint support Support multiple endpoints on cs42L51 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: STM32MP157C-DK2 board uses CS42L51 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (CS42L51) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: cs42l51_port: port { cs42l51_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; cs42l51_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index fd2bd74024c1..80da3cd73e04 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -464,6 +464,13 @@ static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute) return snd_soc_component_write(component, CS42L51_DAC_OUT_CTL, reg); } +static int cs42l51_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_dai_ops cs42l51_dai_ops = { .hw_params = cs42l51_hw_params, .set_sysclk = cs42l51_set_dai_sysclk, @@ -526,6 +533,7 @@ static const struct snd_soc_component_driver soc_component_device_cs42l51 = { .num_dapm_widgets = ARRAY_SIZE(cs42l51_dapm_widgets), .dapm_routes = cs42l51_routes, .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), + .of_xlate_dai_id = cs42l51_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, -- cgit v1.2.3-59-g8ed1b From 2b070f6739025ecbf2ccb55daf9e19d3fb343c7e Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Thu, 28 Mar 2019 10:27:00 +0800 Subject: ASoC: rt5677-spi: Add ACPI ID Add the ACPI ID for the product "chromebook pixel 2015" to match the coreboot settings. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677-spi.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 84501c2020c7..167a02773a0b 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -25,6 +25,7 @@ #include #include #include +#include #include "rt5677-spi.h" @@ -226,9 +227,16 @@ static int rt5677_spi_probe(struct spi_device *spi) return 0; } +static const struct acpi_device_id rt5677_spi_acpi_id[] = { + { "RT5677AA", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, rt5677_spi_acpi_id); + static struct spi_driver rt5677_spi_driver = { .driver = { .name = "rt5677", + .acpi_match_table = ACPI_PTR(rt5677_spi_acpi_id), }, .probe = rt5677_spi_probe, }; -- cgit v1.2.3-59-g8ed1b From 11c0269017b212fd47c593307d2dc3eb9713b2d0 Mon Sep 17 00:00:00 2001 From: Shunli Wang Date: Fri, 29 Mar 2019 16:34:46 +0800 Subject: ASoC: Mediatek: MT8183: Add machine driver with TS3A227 This patch adds support for the machine board with TS3A227. Signed-off-by: Shunli Wang Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 13 + sound/soc/mediatek/mt8183/Makefile | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 382 +++++++++++++++++++++ 3 files changed, 396 insertions(+) create mode 100644 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c (limited to 'sound') diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index b35410e4020e..617fb8f4f694 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -116,6 +116,19 @@ config SND_SOC_MT8183 Select Y if you have such device. If unsure select "N". +config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A + tristate "ASoC Audio driver for MT8183 with MT6358 TS3A227E MAX98357A codec" + depends on SND_SOC_MT8183 + select SND_SOC_MT6358 + select SND_SOC_MAX98357A + select SND_SOC_BT_SCO + select SND_SOC_TS3A227E + help + This adds ASoC driver for Mediatek MT8183 boards + with the MT6358 TS3A227E MAX98357A audio codec. + Select Y if you have such device. + If unsure select "N". + config SND_SOC_MTK_BTCVSD tristate "ALSA BT SCO CVSD/MSBC Driver" help diff --git a/sound/soc/mediatek/mt8183/Makefile b/sound/soc/mediatek/mt8183/Makefile index f3ee6ac98fe8..6177bbd9caaa 100644 --- a/sound/soc/mediatek/mt8183/Makefile +++ b/sound/soc/mediatek/mt8183/Makefile @@ -11,3 +11,4 @@ snd-soc-mt8183-afe-objs := \ mt8183-dai-adda.o obj-$(CONFIG_SND_SOC_MT8183) += snd-soc-mt8183-afe.o +obj-$(CONFIG_SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A) += mt8183-mt6358-ts3a227-max98357.o diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c new file mode 100644 index 000000000000..2da56232a9e1 --- /dev/null +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -0,0 +1,382 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// mt8183-mt6358.c -- +// MT8183-MT6358-TS3A227-MAX98357 ALSA SoC machine driver +// +// Copyright (c) 2018 MediaTek Inc. +// Author: Shunli Wang + +#include +#include +#include +#include +#include + +#include "mt8183-afe-common.h" +#include "../../codecs/ts3a227e.h" + +static struct snd_soc_jack headset_jack; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin headset_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + +}; + +static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int rate = params_rate(params); + unsigned int mclk_fs_ratio = 128; + unsigned int mclk_fs = rate * mclk_fs_ratio; + + return snd_soc_dai_set_sysclk(rtd->cpu_dai, + 0, mclk_fs, SND_SOC_CLOCK_OUT); +} + +static const struct snd_soc_ops mt8183_mt6358_i2s_ops = { + .hw_params = mt8183_mt6358_i2s_hw_params, +}; + +static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + dev_dbg(rtd->dev, "%s(), fix format to 32bit\n", __func__); + + /* fix BE i2s format to 32bit, clean param mask first */ + snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + 0, SNDRV_PCM_FORMAT_LAST); + + params_set_format(params, SNDRV_PCM_FORMAT_S32_LE); + return 0; +} + +static const struct snd_soc_dapm_widget +mt8183_mt6358_ts3a227_max98357_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("IT6505_8CH"), +}; + +static const struct snd_soc_dapm_route +mt8183_mt6358_ts3a227_max98357_dapm_routes[] = { + {"IT6505_8CH", NULL, "TDM"}, +}; + +static struct snd_soc_dai_link +mt8183_mt6358_ts3a227_max98357_dai_links[] = { + /* FE */ + { + .name = "Playback_1", + .stream_name = "Playback_1", + .cpu_dai_name = "DL1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_playback = 1, + }, + { + .name = "Playback_2", + .stream_name = "Playback_2", + .cpu_dai_name = "DL2", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_playback = 1, + }, + { + .name = "Playback_3", + .stream_name = "Playback_3", + .cpu_dai_name = "DL3", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_playback = 1, + }, + { + .name = "Capture_1", + .stream_name = "Capture_1", + .cpu_dai_name = "UL1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_capture = 1, + }, + { + .name = "Capture_2", + .stream_name = "Capture_2", + .cpu_dai_name = "UL2", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_capture = 1, + }, + { + .name = "Capture_3", + .stream_name = "Capture_3", + .cpu_dai_name = "UL3", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_capture = 1, + }, + { + .name = "Capture_Mono_1", + .stream_name = "Capture_Mono_1", + .cpu_dai_name = "UL_MONO_1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_capture = 1, + }, + { + .name = "Playback_HDMI", + .stream_name = "Playback_HDMI", + .cpu_dai_name = "HDMI", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_playback = 1, + }, + /* BE */ + { + .name = "Primary Codec", + .cpu_dai_name = "ADDA", + .codec_dai_name = "mt6358-snd-codec-aif1", + .codec_name = "mt6358-sound", + .no_pcm = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + }, + { + .name = "PCM 1", + .cpu_dai_name = "PCM 1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .no_pcm = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + }, + { + .name = "PCM 2", + .cpu_dai_name = "PCM 2", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .no_pcm = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + }, + { + .name = "I2S0", + .cpu_dai_name = "I2S0", + .codec_dai_name = "bt-sco-pcm", + .codec_name = "bt-sco", + .no_pcm = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_mt6358_i2s_ops, + }, + { + .name = "I2S1", + .cpu_dai_name = "I2S1", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_mt6358_i2s_ops, + }, + { + .name = "I2S2", + .cpu_dai_name = "I2S2", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .no_pcm = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_mt6358_i2s_ops, + }, + { + .name = "I2S3", + .cpu_dai_name = "I2S3", + .codec_dai_name = "HiFi", + .codec_name = "max98357a", + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_mt6358_i2s_ops, + }, + { + .name = "I2S5", + .cpu_dai_name = "I2S5", + .codec_dai_name = "bt-sco-pcm", + .codec_name = "bt-sco", + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_mt6358_i2s_ops, + }, + { + .name = "TDM", + .cpu_dai_name = "TDM", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_suspend = 1, + }, +}; + +static int +mt8183_mt6358_ts3a227_max98357_headset_init(struct snd_soc_component *cpnt); + +static struct snd_soc_aux_dev mt8183_mt6358_ts3a227_max98357_headset_dev = { + .name = "Headset Chip", + .init = mt8183_mt6358_ts3a227_max98357_headset_init, +}; + +static struct snd_soc_card mt8183_mt6358_ts3a227_max98357_card = { + .name = "mt8183_mt6358_ts3a227_max98357", + .owner = THIS_MODULE, + .dai_link = mt8183_mt6358_ts3a227_max98357_dai_links, + .num_links = ARRAY_SIZE(mt8183_mt6358_ts3a227_max98357_dai_links), + .aux_dev = &mt8183_mt6358_ts3a227_max98357_headset_dev, + .num_aux_devs = 1, +}; + +static int +mt8183_mt6358_ts3a227_max98357_headset_init(struct snd_soc_component *component) +{ + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(&mt8183_mt6358_ts3a227_max98357_card, + "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &headset_jack, + headset_jack_pins, + ARRAY_SIZE(headset_jack_pins)); + if (ret) + return ret; + + ret = ts3a227e_enable_jack_detect(component, &headset_jack); + + return ret; +} + +static int +mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &mt8183_mt6358_ts3a227_max98357_card; + struct device_node *platform_node; + struct snd_soc_dai_link *dai_link; + struct pinctrl *default_pins; + int ret, i; + + card->dev = &pdev->dev; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for_each_card_prelinks(card, i, dai_link) { + /* In the alsa soc-core, the "platform" will be + * allocated by devm_kzalloc if null. + * There is a special case that registerring + * sound card is failed at the first time, but + * the "platform" will not null when probe is trying + * again. It's not expected normally. + */ + dai_link->platform = NULL; + + if (dai_link->platform_name) + continue; + dai_link->platform_of_node = platform_node; + } + + mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node = + of_parse_phandle(pdev->dev.of_node, + "mediatek,headset-codec", 0); + if (!mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node) { + dev_err(&pdev->dev, + "Property 'mediatek,headset-codec' missing/invalid\n"); + return -EINVAL; + } + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + default_pins = + devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); + if (IS_ERR(default_pins)) { + dev_err(&pdev->dev, "%s set pins failed\n", + __func__); + return PTR_ERR(default_pins); + } + + return ret; +} + +#ifdef CONFIG_OF +static const struct of_device_id mt8183_mt6358_ts3a227_max98357_dt_match[] = { + {.compatible = "mediatek,mt8183_mt6358_ts3a227_max98357",}, + {} +}; +#endif + +static struct platform_driver mt8183_mt6358_ts3a227_max98357_driver = { + .driver = { + .name = "mt8183_mt6358_ts3a227_max98357", + .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = mt8183_mt6358_ts3a227_max98357_dt_match, +#endif + }, + .probe = mt8183_mt6358_ts3a227_max98357_dev_probe, +}; + +module_platform_driver(mt8183_mt6358_ts3a227_max98357_driver); + +/* Module information */ +MODULE_DESCRIPTION("MT8183-MT6358-TS3A227-MAX98357 ALSA SoC machine driver"); +MODULE_AUTHOR("Shunli Wang "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("mt8183_mt6358_ts3a227_max98357 soc card"); + -- cgit v1.2.3-59-g8ed1b From ebbddc75bbe8d8eab64d3285b0480f5ae9232b2e Mon Sep 17 00:00:00 2001 From: Shunli Wang Date: Fri, 29 Mar 2019 16:34:48 +0800 Subject: ASoC: Mediatek: MT8183: Add machine driver with DA7219 This patch adds support for the machine board with mt6358, da7219 and max98357 codecs. Signed-off-by: Shunli Wang Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 13 + sound/soc/mediatek/mt8183/Makefile | 1 + sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 472 +++++++++++++++++++++ 3 files changed, 486 insertions(+) create mode 100644 sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c (limited to 'sound') diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 617fb8f4f694..874404bcccfd 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -129,6 +129,19 @@ config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A Select Y if you have such device. If unsure select "N". +config SND_SOC_MT8183_DA7219_MAX98357A + tristate "ASoC Audio driver for MT8183 with DA7219 MAX98357A codec" + depends on SND_SOC_MT8183 + select SND_SOC_MT6358 + select SND_SOC_MAX98357A + select SND_SOC_DA7219 + select SND_SOC_BT_SCO + help + This adds ASoC driver for Mediatek MT8183 boards + with the DA7219 MAX98357A audio codec. + Select Y if you have such device. + If unsure select "N". + config SND_SOC_MTK_BTCVSD tristate "ALSA BT SCO CVSD/MSBC Driver" help diff --git a/sound/soc/mediatek/mt8183/Makefile b/sound/soc/mediatek/mt8183/Makefile index 6177bbd9caaa..c0a3bbc2c1f6 100644 --- a/sound/soc/mediatek/mt8183/Makefile +++ b/sound/soc/mediatek/mt8183/Makefile @@ -12,3 +12,4 @@ snd-soc-mt8183-afe-objs := \ obj-$(CONFIG_SND_SOC_MT8183) += snd-soc-mt8183-afe.o obj-$(CONFIG_SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A) += mt8183-mt6358-ts3a227-max98357.o +obj-$(CONFIG_SND_SOC_MT8183_DA7219_MAX98357A) += mt8183-da7219-max98357.o diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c new file mode 100644 index 000000000000..93536659a859 --- /dev/null +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -0,0 +1,472 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// mt8183-da7219-max98357.c +// -- MT8183-DA7219-MAX98357 ALSA SoC machine driver +// +// Copyright (c) 2018 MediaTek Inc. +// Author: Shunli Wang + +#include +#include +#include +#include +#include + +#include "mt8183-afe-common.h" +#include "../../codecs/da7219-aad.h" +#include "../../codecs/da7219.h" + +static struct snd_soc_jack headset_jack; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin headset_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static struct snd_soc_dai_link_component +mt8183_da7219_max98357_external_codecs[] = { + { + .name = "max98357a", + .dai_name = "HiFi", + }, + { + .name = "da7219.5-001a", + .dai_name = "da7219-hifi", + }, +}; + +static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int rate = params_rate(params); + unsigned int mclk_fs_ratio = 128; + unsigned int mclk_fs = rate * mclk_fs_ratio; + + return snd_soc_dai_set_sysclk(rtd->cpu_dai, + 0, mclk_fs, SND_SOC_CLOCK_OUT); +} + +static const struct snd_soc_ops mt8183_mt6358_i2s_ops = { + .hw_params = mt8183_mt6358_i2s_hw_params, +}; + +static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int rate = params_rate(params); + unsigned int mclk_fs_ratio = 256; + unsigned int mclk_fs = rate * mclk_fs_ratio; + unsigned int freq; + int ret = 0, j; + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, + mclk_fs, SND_SOC_CLOCK_OUT); + if (ret < 0) + dev_err(rtd->dev, "failed to set cpu dai sysclk\n"); + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + + if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { + ret = snd_soc_dai_set_sysclk(codec_dai, + DA7219_CLKSRC_MCLK, + mclk_fs, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "failed to set sysclk\n"); + + if ((rate % 8000) == 0) + freq = DA7219_PLL_FREQ_OUT_98304; + else + freq = DA7219_PLL_FREQ_OUT_90316; + + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_PLL_SRM, + 0, freq); + if (ret) + dev_err(rtd->dev, "failed to start PLL: %d\n", + ret); + } + } + + return ret; +} + +static int mt8183_da7219_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0, j; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + + if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { + ret = snd_soc_dai_set_pll(codec_dai, + 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(rtd->dev, "failed to stop PLL: %d\n", + ret); + break; + } + } + } + + return ret; +} + +static const struct snd_soc_ops mt8183_da7219_i2s_ops = { + .hw_params = mt8183_da7219_i2s_hw_params, + .hw_free = mt8183_da7219_hw_free, +}; + +static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + /* fix BE i2s format to 32bit, clean param mask first */ + snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + 0, SNDRV_PCM_FORMAT_LAST); + + params_set_format(params, SNDRV_PCM_FORMAT_S32_LE); + + return 0; +} + +static const struct snd_soc_dapm_widget +mt8183_da7219_max98357_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("IT6505_8CH"), +}; + +static const struct snd_soc_dapm_route mt8183_da7219_max98357_dapm_routes[] = { + {"IT6505_8CH", NULL, "TDM"}, +}; + +static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { + /* FE */ + { + .name = "Playback_1", + .stream_name = "Playback_1", + .cpu_dai_name = "DL1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_playback = 1, + }, + { + .name = "Playback_2", + .stream_name = "Playback_2", + .cpu_dai_name = "DL2", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_playback = 1, + }, + { + .name = "Playback_3", + .stream_name = "Playback_3", + .cpu_dai_name = "DL3", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_playback = 1, + }, + { + .name = "Capture_1", + .stream_name = "Capture_1", + .cpu_dai_name = "UL1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_capture = 1, + }, + { + .name = "Capture_2", + .stream_name = "Capture_2", + .cpu_dai_name = "UL2", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_capture = 1, + }, + { + .name = "Capture_3", + .stream_name = "Capture_3", + .cpu_dai_name = "UL3", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_capture = 1, + }, + { + .name = "Capture_Mono_1", + .stream_name = "Capture_Mono_1", + .cpu_dai_name = "UL_MONO_1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_capture = 1, + }, + { + .name = "Playback_HDMI", + .stream_name = "Playback_HDMI", + .cpu_dai_name = "HDMI", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_PRE, + SND_SOC_DPCM_TRIGGER_PRE}, + .dynamic = 1, + .dpcm_playback = 1, + }, + /* BE */ + { + .name = "Primary Codec", + .cpu_dai_name = "ADDA", + .codec_dai_name = "mt6358-snd-codec-aif1", + .codec_name = "mt6358-sound", + .no_pcm = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + }, + { + .name = "PCM 1", + .cpu_dai_name = "PCM 1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .no_pcm = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + }, + { + .name = "PCM 2", + .cpu_dai_name = "PCM 2", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .no_pcm = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + }, + { + .name = "I2S0", + .cpu_dai_name = "I2S0", + .codec_dai_name = "bt-sco-pcm", + .codec_name = "bt-sco", + .no_pcm = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_mt6358_i2s_ops, + }, + { + .name = "I2S1", + .cpu_dai_name = "I2S1", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_mt6358_i2s_ops, + }, + { + .name = "I2S2", + .cpu_dai_name = "I2S2", + .codec_dai_name = "da7219-hifi", + .codec_name = "da7219.5-001a", + .no_pcm = 1, + .dpcm_capture = 1, + .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_da7219_i2s_ops, + }, + { + .name = "I2S3", + .cpu_dai_name = "I2S3", + .codecs = mt8183_da7219_max98357_external_codecs, + .num_codecs = + ARRAY_SIZE(mt8183_da7219_max98357_external_codecs), + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_da7219_i2s_ops, + }, + { + .name = "I2S5", + .cpu_dai_name = "I2S5", + .codec_dai_name = "bt-sco-pcm", + .codec_name = "bt-sco", + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_mt6358_i2s_ops, + }, + { + .name = "TDM", + .cpu_dai_name = "TDM", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_suspend = 1, + }, +}; + +static int +mt8183_da7219_max98357_headset_init(struct snd_soc_component *component); + +static struct snd_soc_aux_dev mt8183_da7219_max98357_headset_dev = { + .name = "Headset Chip", + .init = mt8183_da7219_max98357_headset_init, +}; + +static struct snd_soc_codec_conf mt6358_codec_conf[] = { + { + .dev_name = "mt6358-sound", + .name_prefix = "Mt6358", + }, +}; + +static struct snd_soc_card mt8183_da7219_max98357_card = { + .name = "mt8183_da7219_max98357", + .owner = THIS_MODULE, + .dai_link = mt8183_da7219_max98357_dai_links, + .num_links = ARRAY_SIZE(mt8183_da7219_max98357_dai_links), + .aux_dev = &mt8183_da7219_max98357_headset_dev, + .num_aux_devs = 1, + .codec_conf = mt6358_codec_conf, + .num_configs = ARRAY_SIZE(mt6358_codec_conf), +}; + +static int +mt8183_da7219_max98357_headset_init(struct snd_soc_component *component) +{ + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(&mt8183_da7219_max98357_card, + "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &headset_jack, + headset_jack_pins, + ARRAY_SIZE(headset_jack_pins)); + if (ret) + return ret; + + da7219_aad_jack_det(component, &headset_jack); + + return ret; +} + +static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &mt8183_da7219_max98357_card; + struct device_node *platform_node; + struct snd_soc_dai_link *dai_link; + struct pinctrl *default_pins; + int ret, i; + + card->dev = &pdev->dev; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for_each_card_prelinks(card, i, dai_link) { + /* In the alsa soc-core, the "platform" will be + * allocated by devm_kzalloc if null. + * There is a special case that registerring + * sound card is failed at the first time, but + * the "platform" will not null when probe is trying + * again. It's not expected normally. + */ + dai_link->platform = NULL; + + if (dai_link->platform_name) + continue; + dai_link->platform_of_node = platform_node; + } + + mt8183_da7219_max98357_headset_dev.codec_of_node = + of_parse_phandle(pdev->dev.of_node, + "mediatek,headset-codec", 0); + if (!mt8183_da7219_max98357_headset_dev.codec_of_node) { + dev_err(&pdev->dev, + "Property 'mediatek,headset-codec' missing/invalid\n"); + return -EINVAL; + } + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) { + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + return ret; + } + + default_pins = + devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); + if (IS_ERR(default_pins)) { + dev_err(&pdev->dev, "%s set pins failed\n", + __func__); + return PTR_ERR(default_pins); + } + + return ret; +} + +#ifdef CONFIG_OF +static const struct of_device_id mt8183_da7219_max98357_dt_match[] = { + {.compatible = "mediatek,mt8183_da7219_max98357",}, + {} +}; +#endif + +static struct platform_driver mt8183_da7219_max98357_driver = { + .driver = { + .name = "mt8183_da7219_max98357", + .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = mt8183_da7219_max98357_dt_match, +#endif + }, + .probe = mt8183_da7219_max98357_dev_probe, +}; + +module_platform_driver(mt8183_da7219_max98357_driver); + +/* Module information */ +MODULE_DESCRIPTION("MT8183-DA7219-MAX98357 ALSA SoC machine driver"); +MODULE_AUTHOR("Shunli Wang "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("mt8183_da7219_max98357 soc card"); + -- cgit v1.2.3-59-g8ed1b From de70b2a581125a8a627db6fdc1d7f9c05536fe84 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 28 Mar 2019 09:28:16 +0000 Subject: ASoC: fsl: fix spelling mistake: "missign" -> "missing" There is a spelling mistake in a dev_err message. Fix this. Signed-off-by: Colin Ian King Reviewed-by: Mukesh Ojha Acked-by: Viorel Suman Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmix.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 72e37ca6cfbb..7983bd339c01 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -161,7 +161,7 @@ static int imx_audmix_probe(struct platform_device *pdev) } if (!audmix_np) { - dev_err(&pdev->dev, "Missign DT node for parent device.\n"); + dev_err(&pdev->dev, "Missing DT node for parent device.\n"); return -EINVAL; } -- cgit v1.2.3-59-g8ed1b From 809589a94dcce27cb5f2081280d1155c71ae45f7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 27 Mar 2019 15:24:51 +0000 Subject: ASoC: wm_adsp: Fix typo in comment Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5e615fbddf9a..0aa62b26f61c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3263,7 +3263,7 @@ static void wm_halo_stop_core(struct wm_adsp *dsp) regmap_update_bits(dsp->regmap, dsp->base + HALO_CCM_CORE_CONTROL, HALO_CORE_EN, 0); - /* reset halo core with CORE_SOFT_REEST */ + /* reset halo core with CORE_SOFT_RESET */ regmap_update_bits(dsp->regmap, dsp->base + HALO_CORE_SOFT_RESET, HALO_CORE_SOFT_RESET_MASK, 1); } -- cgit v1.2.3-59-g8ed1b From 0bbcedd6bb41b82b77115fc001441ff34e7c4ea7 Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 29 Mar 2019 22:50:43 +0100 Subject: ASoC: es8316: Add support for inverted jack detect On some devices (Teclast X98+ II tablet, maybe others), the jack detection has been wired backwards, so when the ES8316 reports headphones being present it means they are actually not plugged. Use a quirk around this incorrect behaviour, which can be enabled through the 'everest,jack-detect-inverted' boolean device property. Signed-off-by: Paul Cercueil Signed-off-by: Mark Brown --- sound/soc/codecs/es8316.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 6d4a323f786b..ec2770b3f77d 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -43,6 +43,7 @@ struct es8316_priv { unsigned int sysclk; unsigned int allowed_rates[NR_SUPPORTED_MCLK_LRCK_RATIOS]; struct snd_pcm_hw_constraint_list sysclk_constraints; + bool jd_inverted; }; /* @@ -577,6 +578,9 @@ static irqreturn_t es8316_irq(int irq, void *data) if (!es8316->jack) goto out; + if (es8316->jd_inverted) + flags ^= ES8316_GPIO_FLAG_HP_NOT_INSERTED; + dev_dbg(comp->dev, "gpio flags %#04x\n", flags); if (flags & ES8316_GPIO_FLAG_HP_NOT_INSERTED) { /* Jack removed, or spurious IRQ? */ @@ -592,6 +596,8 @@ static irqreturn_t es8316_irq(int irq, void *data) /* Jack inserted, determine type */ es8316_enable_micbias_for_mic_gnd_short_detect(comp); regmap_read(es8316->regmap, ES8316_GPIO_FLAG, &flags); + if (es8316->jd_inverted) + flags ^= ES8316_GPIO_FLAG_HP_NOT_INSERTED; dev_dbg(comp->dev, "gpio flags %#04x\n", flags); if (flags & ES8316_GPIO_FLAG_HP_NOT_INSERTED) { /* Jack unplugged underneath us */ @@ -633,6 +639,14 @@ static void es8316_enable_jack_detect(struct snd_soc_component *component, { struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + /* + * Init es8316->jd_inverted here and not in the probe, as we cannot + * guarantee that the bytchr-es8316 driver, which might set this + * property, will probe before us. + */ + es8316->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + mutex_lock(&es8316->lock); es8316->jack = jack; -- cgit v1.2.3-59-g8ed1b From 35146467bd43704c52d1d5760af9839a10ad5b12 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Mon, 25 Mar 2019 03:17:40 +0000 Subject: ASoC: simple-card-utils: remove set but not used variable 'dai_name' Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_parse_clk': sound/soc/generic/simple-card-utils.c:164:18: warning: parameter 'dai_name' set but not used [-Wunused-but-set-parameter] It's not used since commit 0580dde59438 ("ASoC: simple-card-utils: add asoc_simple_debug_info()"), so can be removed. Signed-off-by: YueHaibing Acked-by: Kuninori Morimoto Reviewed-by: Mukesh Ojha Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 4ed68348f939..db1458a19985 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -173,10 +173,8 @@ int asoc_simple_parse_clk(struct device *dev, * see * soc-core.c :: snd_soc_init_multicodec() */ - if (dlc) { + if (dlc) dai_of_node = dlc->of_node; - dai_name = dlc->dai_name; - } /* * Parse dai->sysclk come from "clocks = <&xxx>" -- cgit v1.2.3-59-g8ed1b From ebf3326cd96989a498892a1b344faee3110f7877 Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" Date: Mon, 1 Apr 2019 10:13:28 -0500 Subject: ASoC: tlv320aic31xx: Add headphone/headset detection This device can detect the insertion/removal of headphones and headsets. Enable reporting this status by enabling this interrupt and forwarding this to upper-layers if a jack has been defined. This jack definition and the resulting operation from a jack detection event must currently be defined by sound card platform code until CODEC outputs to jack mappings can be defined generically. Signed-off-by: Andrew F. Davis Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 48 +++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/tlv320aic31xx.h | 11 +++++++++ 2 files changed, 58 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index c544a1e35f5e..a60cd8c08a14 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -89,6 +90,7 @@ static bool aic31xx_volatile(struct device *dev, unsigned int reg) case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ case AIC31XX_INTRDACFLAG2: case AIC31XX_INTRADCFLAG2: + case AIC31XX_HSDETECT: return true; } return false; @@ -163,6 +165,7 @@ struct aic31xx_priv { struct aic31xx_pdata pdata; struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; + struct snd_soc_jack *jack; unsigned int sysclk; u8 p_div; int rate_div_line; @@ -1261,6 +1264,20 @@ static int aic31xx_set_bias_level(struct snd_soc_component *component, return 0; } +int aic31xx_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); + + aic31xx->jack = jack; + + /* Enable/Disable jack detection */ + regmap_write(aic31xx->regmap, AIC31XX_HSDETECT, + jack ? AIC31XX_HSD_ENABLE : 0); + + return 0; +} + static int aic31xx_codec_probe(struct snd_soc_component *component) { struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); @@ -1301,6 +1318,7 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, + .set_jack = aic31xx_set_jack, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), @@ -1405,8 +1423,35 @@ static irqreturn_t aic31xx_irq(int irq, void *data) dev_err(dev, "Short circuit on Left output is detected\n"); if (value & AIC31XX_HPRSCDETECT) dev_err(dev, "Short circuit on Right output is detected\n"); + if (value & AIC31XX_HSPLUG) { + unsigned int val; + int status = 0; + + ret = regmap_read(aic31xx->regmap, AIC31XX_HSDETECT, &val); + if (ret) { + dev_err(dev, "Failed to read headset type: %d\n", ret); + goto exit; + } + + switch ((val & AIC31XX_HSD_TYPE_MASK) >> + AIC31XX_HSD_TYPE_SHIFT) { + case AIC31XX_HSD_HP: + status |= SND_JACK_HEADPHONE; + break; + case AIC31XX_HSD_HS: + status |= SND_JACK_HEADSET; + break; + default: + break; + } + + if (aic31xx->jack) + snd_soc_jack_report(aic31xx->jack, status, + AIC31XX_JACK_MASK); + } if (value & ~(AIC31XX_HPLSCDETECT | - AIC31XX_HPRSCDETECT)) + AIC31XX_HPRSCDETECT | + AIC31XX_HSPLUG)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); read_overflow: @@ -1518,6 +1563,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, AIC31XX_GPIO1_FUNC_SHIFT); regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_HSPLUGDET | AIC31XX_SC | AIC31XX_ENGINE); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 2636f2c6bc79..9985ceed97ae 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -20,6 +20,9 @@ #define AIC31XX_MINIDSP_BIT BIT(2) #define DAC31XX_BIT BIT(3) +#define AIC31XX_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_HEADSET) + enum aic31xx_type { AIC3100 = 0, AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, @@ -220,6 +223,14 @@ struct aic31xx_pdata { /* AIC31XX_DACMUTE */ #define AIC31XX_DACMUTE_MASK GENMASK(3, 2) +/* AIC31XX_HSDETECT */ +#define AIC31XX_HSD_ENABLE BIT(7) +#define AIC31XX_HSD_TYPE_MASK GENMASK(6, 5) +#define AIC31XX_HSD_TYPE_SHIFT 5 +#define AIC31XX_HSD_NONE 0x00 +#define AIC31XX_HSD_HP 0x01 +#define AIC31XX_HSD_HS 0x03 + /* AIC31XX_MICBIAS */ #define AIC31XX_MICBIAS_MASK GENMASK(1, 0) #define AIC31XX_MICBIAS_SHIFT 0 -- cgit v1.2.3-59-g8ed1b From 90f0202bf2e15e8933a550b7dfd439b500a76e1c Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" Date: Mon, 1 Apr 2019 10:13:29 -0500 Subject: ASoC: tlv320aic31xx: Add button press detection This device can optionally detect headset or microphone button presses. Add support for this by passing this event to the jack layer. Signed-off-by: Andrew F. Davis Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 17 +++++++++++++++-- sound/soc/codecs/tlv320aic31xx.h | 3 ++- 2 files changed, 17 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index a60cd8c08a14..fa47bfcb93e9 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1423,10 +1423,21 @@ static irqreturn_t aic31xx_irq(int irq, void *data) dev_err(dev, "Short circuit on Left output is detected\n"); if (value & AIC31XX_HPRSCDETECT) dev_err(dev, "Short circuit on Right output is detected\n"); - if (value & AIC31XX_HSPLUG) { + if (value & (AIC31XX_HSPLUG | AIC31XX_BUTTONPRESS)) { unsigned int val; int status = 0; + ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG2, + &val); + if (ret) { + dev_err(dev, "Failed to read interrupt mask: %d\n", + ret); + goto exit; + } + + if (val & AIC31XX_BUTTONPRESS) + status |= SND_JACK_BTN_0; + ret = regmap_read(aic31xx->regmap, AIC31XX_HSDETECT, &val); if (ret) { dev_err(dev, "Failed to read headset type: %d\n", ret); @@ -1451,7 +1462,8 @@ static irqreturn_t aic31xx_irq(int irq, void *data) } if (value & ~(AIC31XX_HPLSCDETECT | AIC31XX_HPRSCDETECT | - AIC31XX_HSPLUG)) + AIC31XX_HSPLUG | + AIC31XX_BUTTONPRESS)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); read_overflow: @@ -1564,6 +1576,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, AIC31XX_HSPLUGDET | + AIC31XX_BUTTONPRESSDET | AIC31XX_SC | AIC31XX_ENGINE); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 9985ceed97ae..cb024955c978 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -21,7 +21,8 @@ #define DAC31XX_BIT BIT(3) #define AIC31XX_JACK_MASK (SND_JACK_HEADPHONE | \ - SND_JACK_HEADSET) + SND_JACK_HEADSET | \ + SND_JACK_BTN_0) enum aic31xx_type { AIC3100 = 0, -- cgit v1.2.3-59-g8ed1b From f661fa2804af6950d1b7f5ffa22f6a80dcf52958 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 27 Mar 2019 08:43:19 -0700 Subject: ASoC: fix SND_SOC_LOCHNAGAR_SC kconfig warning Fix kconfig warning for SND_SOC_LOCHNAGAR_SC: WARNING: unmet direct dependencies detected for SND_SOC_LOCHNAGAR_SC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=m] && MFD_LOCHNAGAR [=n] Selected by [m]: - SND_SOC_ALL_CODECS [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=m] && COMPILE_TEST [=y] Signed-off-by: Randy Dunlap Cc: Piotr Stankiewicz Cc: Charles Keepax Cc: Richard Fitzgerald Cc: patches@opensource.cirrus.com Cc: Liam Girdwood Cc: Mark Brown Cc: alsa-devel@alsa-project.org Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 419142111b6d..9981d40ef45b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -94,7 +94,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_JZ4725B_CODEC select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C - select SND_SOC_LOCHNAGAR_SC + select SND_SOC_LOCHNAGAR_SC if MFD_LOCHNAGAR select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C -- cgit v1.2.3-59-g8ed1b From cce1396936ef2b347d622b4d49718818eb32029d Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Wed, 27 Mar 2019 19:47:03 +0800 Subject: ASoC: sprd: Add Spreadtrum audio compress offload support We use 2-stage DMA mode to support Spreadtrum audio compress offload, which means we use one DMA source channel to transfer data from IRAM buffer to the DSP fifo to do decoding/encoding, once IRAM buffer is empty by transferring done, another DMA destination channel will be triggered automatically to start to transfer data from DDR buffer to the IRAM buffer. This can reduce the AP subsystem wakeup times to save power. Co-developed-by: Yintang Ren Signed-off-by: Baolin Wang Signed-off-by: Mark Brown --- sound/soc/sprd/Kconfig | 1 + sound/soc/sprd/Makefile | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 675 +++++++++++++++++++++++++++++++++++++ sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/sprd/sprd-pcm-dma.h | 43 +++ 5 files changed, 721 insertions(+), 2 deletions(-) create mode 100644 sound/soc/sprd/sprd-pcm-compress.c (limited to 'sound') diff --git a/sound/soc/sprd/Kconfig b/sound/soc/sprd/Kconfig index 43ece7daf0e9..3b1eb320dab4 100644 --- a/sound/soc/sprd/Kconfig +++ b/sound/soc/sprd/Kconfig @@ -1,6 +1,7 @@ config SND_SOC_SPRD tristate "SoC Audio for the Spreadtrum SoC chips" depends on ARCH_SPRD || COMPILE_TEST + select SND_SOC_COMPRESS help Say Y or M if you want to add support for codecs attached to the Spreadtrum SoCs' Audio interfaces. diff --git a/sound/soc/sprd/Makefile b/sound/soc/sprd/Makefile index 47620e57a9f2..fa1d5476b8da 100644 --- a/sound/soc/sprd/Makefile +++ b/sound/soc/sprd/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0 # Spreadtrum Audio Support -obj-$(CONFIG_SND_SOC_SPRD) += sprd-pcm-dma.o +obj-$(CONFIG_SND_SOC_SPRD) += sprd-pcm-dma.o sprd-pcm-compress.o diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c new file mode 100644 index 000000000000..dc84257cffc2 --- /dev/null +++ b/sound/soc/sprd/sprd-pcm-compress.c @@ -0,0 +1,675 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (C) 2019 Spreadtrum Communications Inc. + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sprd-pcm-dma.h" + +#define SPRD_COMPR_DMA_CHANS 2 + +/* Default values if userspace does not set */ +#define SPRD_COMPR_MIN_FRAGMENT_SIZE SZ_8K +#define SPRD_COMPR_MAX_FRAGMENT_SIZE SZ_128K +#define SPRD_COMPR_MIN_NUM_FRAGMENTS 4 +#define SPRD_COMPR_MAX_NUM_FRAGMENTS 64 + +/* DSP FIFO size */ +#define SPRD_COMPR_MCDT_EMPTY_WMK 0 +#define SPRD_COMPR_MCDT_FIFO_SIZE 512 + +/* Stage 0 IRAM buffer size definition */ +#define SPRD_COMPR_IRAM_BUF_SIZE SZ_32K +#define SPRD_COMPR_IRAM_INFO_SIZE (sizeof(struct sprd_compr_playinfo)) +#define SPRD_COMPR_IRAM_LINKLIST_SIZE (1024 - SPRD_COMPR_IRAM_INFO_SIZE) +#define SPRD_COMPR_IRAM_SIZE (SPRD_COMPR_IRAM_BUF_SIZE + \ + SPRD_COMPR_IRAM_INFO_SIZE + \ + SPRD_COMPR_IRAM_LINKLIST_SIZE) + +/* Stage 1 DDR buffer size definition */ +#define SPRD_COMPR_AREA_BUF_SIZE SZ_2M +#define SPRD_COMPR_AREA_LINKLIST_SIZE 1024 +#define SPRD_COMPR_AREA_SIZE (SPRD_COMPR_AREA_BUF_SIZE + \ + SPRD_COMPR_AREA_LINKLIST_SIZE) + +struct sprd_compr_dma { + struct dma_chan *chan; + struct dma_async_tx_descriptor *desc; + dma_cookie_t cookie; + dma_addr_t phys; + void *virt; + int trans_len; +}; + +/* + * The Spreadtrum Audio compress offload mode will use 2-stage DMA transfer to + * save power. That means we can request 2 dma channels, one for source channel, + * and another one for destination channel. Once the source channel's transaction + * is done, it will trigger the destination channel's transaction automatically + * by hardware signal. + * + * For 2-stage DMA transfer, we can allocate 2 buffers: IRAM buffer (always + * power-on) and DDR buffer. The source channel will transfer data from IRAM + * buffer to the DSP fifo to decoding/encoding, once IRAM buffer is empty by + * transferring done, the destination channel will start to transfer data from + * DDR buffer to IRAM buffer. + * + * Since the DSP fifo is only 512B, IRAM buffer is allocated by 32K, and DDR + * buffer is larger to 2M. That means only the IRAM 32k data is transferred + * done, we can wake up the AP system to transfer data from DDR to IRAM, and + * other time the AP system can be suspended to save power. + */ +struct sprd_compr_stream { + struct snd_compr_stream *cstream; + struct sprd_compr_ops *compr_ops; + struct sprd_compr_dma dma[SPRD_COMPR_DMA_CHANS]; + + /* DMA engine channel number */ + int num_channels; + + /* Stage 0 IRAM buffer */ + struct snd_dma_buffer iram_buffer; + /* Stage 1 DDR buffer */ + struct snd_dma_buffer compr_buffer; + + /* DSP play information IRAM buffer */ + dma_addr_t info_phys; + void *info_area; + int info_size; + + /* Data size copied to IRAM buffer */ + int copied_total; + /* Total received data size from userspace */ + int received_total; + /* Stage 0 IRAM buffer received data size */ + int received_stage0; + /* Stage 1 DDR buffer received data size */ + int received_stage1; + /* Stage 1 DDR buffer pointer */ + int stage1_pointer; +}; + +static int sprd_platform_compr_trigger(struct snd_compr_stream *cstream, + int cmd); + +static void sprd_platform_compr_drain_notify(void *arg) +{ + struct snd_compr_stream *cstream = arg; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + + memset(stream->info_area, 0, sizeof(struct sprd_compr_playinfo)); + + if (cstream) + snd_compr_drain_notify(cstream); +} + +static void sprd_platform_compr_dma_complete(void *data) +{ + struct snd_compr_stream *cstream = data; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct sprd_compr_dma *dma = &stream->dma[1]; + + /* Update data size copied to IRAM buffer */ + stream->copied_total += dma->trans_len; + if (stream->copied_total > stream->received_total) + stream->copied_total = stream->received_total; + + snd_compr_fragment_elapsed(cstream); +} + +static int sprd_platform_compr_dma_config(struct snd_compr_stream *cstream, + struct snd_compr_params *params, + int channel) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = component->dev; + struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct sprd_pcm_dma_params *dma_params = data->dma_params; + struct sprd_compr_dma *dma = &stream->dma[channel]; + struct dma_slave_config config = { }; + struct sprd_dma_linklist link = { }; + enum dma_transfer_direction dir; + struct scatterlist *sg, *sgt; + enum dma_slave_buswidth bus_width; + int period, period_cnt, sg_num = 2; + dma_addr_t src_addr, dst_addr; + unsigned long flags; + int ret, j; + + if (!dma_params) { + dev_err(dev, "no dma parameters setting\n"); + return -EINVAL; + } + + dma->chan = dma_request_slave_channel(dev, + dma_params->chan_name[channel]); + if (!dma->chan) { + dev_err(dev, "failed to request dma channel\n"); + return -ENODEV; + } + + sgt = sg = devm_kcalloc(dev, sg_num, sizeof(*sg), GFP_KERNEL); + if (!sg) { + ret = -ENOMEM; + goto sg_err; + } + + switch (channel) { + case 0: + bus_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + period = (SPRD_COMPR_MCDT_FIFO_SIZE - SPRD_COMPR_MCDT_EMPTY_WMK) * 4; + period_cnt = params->buffer.fragment_size / period; + src_addr = stream->iram_buffer.addr; + dst_addr = dma_params->dev_phys[channel]; + flags = SPRD_DMA_FLAGS(SPRD_DMA_SRC_CHN1, + SPRD_DMA_TRANS_DONE_TRG, + SPRD_DMA_FRAG_REQ, + SPRD_DMA_TRANS_INT); + break; + + case 1: + bus_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + period = params->buffer.fragment_size; + period_cnt = params->buffer.fragments; + src_addr = stream->compr_buffer.addr; + dst_addr = stream->iram_buffer.addr; + flags = SPRD_DMA_FLAGS(SPRD_DMA_DST_CHN1, + SPRD_DMA_TRANS_DONE_TRG, + SPRD_DMA_FRAG_REQ, + SPRD_DMA_TRANS_INT); + break; + + default: + ret = -EINVAL; + goto config_err; + } + + dma->trans_len = period * period_cnt; + + config.src_maxburst = period; + config.src_addr_width = bus_width; + config.dst_addr_width = bus_width; + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + config.src_addr = src_addr; + config.dst_addr = dst_addr; + dir = DMA_MEM_TO_DEV; + } else { + config.src_addr = dst_addr; + config.dst_addr = src_addr; + dir = DMA_DEV_TO_MEM; + } + + sg_init_table(sgt, sg_num); + for (j = 0; j < sg_num; j++, sgt++) { + sg_dma_len(sgt) = dma->trans_len; + sg_dma_address(sgt) = dst_addr; + } + + /* + * Configure the link-list address for the DMA engine link-list + * mode. + */ + link.virt_addr = (unsigned long)dma->virt; + link.phy_addr = dma->phys; + + ret = dmaengine_slave_config(dma->chan, &config); + if (ret) { + dev_err(dev, + "failed to set slave configuration: %d\n", ret); + goto config_err; + } + + /* + * We configure the DMA request mode, interrupt mode, channel + * mode and channel trigger mode by the flags. + */ + dma->desc = dma->chan->device->device_prep_slave_sg(dma->chan, sg, + sg_num, dir, + flags, &link); + if (!dma->desc) { + dev_err(dev, "failed to prepare slave sg\n"); + ret = -ENOMEM; + goto config_err; + } + + /* Only channel 1 transfer can wake up the AP system. */ + if (!params->no_wake_mode && channel == 1) { + dma->desc->callback = sprd_platform_compr_dma_complete; + dma->desc->callback_param = cstream; + } + + devm_kfree(dev, sg); + + return 0; + +config_err: + devm_kfree(dev, sg); +sg_err: + dma_release_channel(dma->chan); + return ret; +} + +static int sprd_platform_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = component->dev; + struct sprd_compr_params compr_params = { }; + int ret; + + /* + * Configure the DMA engine 2-stage transfer mode. Channel 1 set as the + * destination channel, and channel 0 set as the source channel, that + * means once the source channel's transaction is done, it will trigger + * the destination channel's transaction automatically. + */ + ret = sprd_platform_compr_dma_config(cstream, params, 1); + if (ret) { + dev_err(dev, "faied to config stage 1 DMA: %d\n", ret); + return ret; + } + + ret = sprd_platform_compr_dma_config(cstream, params, 0); + if (ret) { + dev_err(dev, "faied to config stage 0 DMA: %d\n", ret); + goto config_err; + } + + compr_params.direction = cstream->direction; + compr_params.sample_rate = params->codec.sample_rate; + compr_params.channels = stream->num_channels; + compr_params.info_phys = stream->info_phys; + compr_params.info_size = stream->info_size; + compr_params.rate = params->codec.bit_rate; + compr_params.format = params->codec.id; + + ret = stream->compr_ops->set_params(cstream->direction, &compr_params); + if (ret) { + dev_err(dev, "failed to set parameters: %d\n", ret); + goto params_err; + } + + return 0; + +params_err: + dma_release_channel(stream->dma[0].chan); +config_err: + dma_release_channel(stream->dma[1].chan); + return ret; +} + +static int sprd_platform_compr_open(struct snd_compr_stream *cstream) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = component->dev; + struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct sprd_compr_stream *stream; + struct sprd_compr_callback cb; + int stream_id = cstream->direction, ret; + + ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + stream = devm_kzalloc(dev, sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->cstream = cstream; + stream->num_channels = 2; + stream->compr_ops = data->ops; + + /* + * Allocate the stage 0 IRAM buffer size, including the DMA 0 + * link-list size and play information of DSP address size. + */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_IRAM, dev, + SPRD_COMPR_IRAM_SIZE, &stream->iram_buffer); + if (ret < 0) + goto err_iram; + + /* Use to save link-list configuration for DMA 0. */ + stream->dma[0].virt = stream->iram_buffer.area + SPRD_COMPR_IRAM_SIZE; + stream->dma[0].phys = stream->iram_buffer.addr + SPRD_COMPR_IRAM_SIZE; + + /* Use to update the current data offset of DSP. */ + stream->info_phys = stream->iram_buffer.addr + SPRD_COMPR_IRAM_SIZE + + SPRD_COMPR_IRAM_LINKLIST_SIZE; + stream->info_area = stream->iram_buffer.area + SPRD_COMPR_IRAM_SIZE + + SPRD_COMPR_IRAM_LINKLIST_SIZE; + stream->info_size = SPRD_COMPR_IRAM_INFO_SIZE; + + /* + * Allocate the stage 1 DDR buffer size, including the DMA 1 link-list + * size. + */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, + SPRD_COMPR_AREA_SIZE, &stream->compr_buffer); + if (ret < 0) + goto err_compr; + + /* Use to save link-list configuration for DMA 1. */ + stream->dma[1].virt = stream->compr_buffer.area + SPRD_COMPR_AREA_SIZE; + stream->dma[1].phys = stream->compr_buffer.addr + SPRD_COMPR_AREA_SIZE; + + cb.drain_notify = sprd_platform_compr_drain_notify; + cb.drain_data = cstream; + ret = stream->compr_ops->open(stream_id, &cb); + if (ret) { + dev_err(dev, "failed to open compress platform: %d\n", ret); + goto err_open; + } + + runtime->private_data = stream; + return 0; + +err_open: + snd_dma_free_pages(&stream->compr_buffer); +err_compr: + snd_dma_free_pages(&stream->iram_buffer); +err_iram: + devm_kfree(dev, stream); + + return ret; +} + +static int sprd_platform_compr_free(struct snd_compr_stream *cstream) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = component->dev; + int stream_id = cstream->direction, i; + + for (i = 0; i < stream->num_channels; i++) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (dma->chan) { + dma_release_channel(dma->chan); + dma->chan = NULL; + } + } + + snd_dma_free_pages(&stream->compr_buffer); + snd_dma_free_pages(&stream->iram_buffer); + + stream->compr_ops->close(stream_id); + + devm_kfree(dev, stream); + return 0; +} + +static int sprd_platform_compr_trigger(struct snd_compr_stream *cstream, + int cmd) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = component->dev; + int channels = stream->num_channels, ret = 0, i; + int stream_id = cstream->direction; + + if (cstream->direction != SND_COMPRESS_PLAYBACK) { + dev_err(dev, "unsupported compress direction\n"); + return -EINVAL; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + for (i = channels - 1; i >= 0; i--) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (!dma->desc) + continue; + + dma->cookie = dmaengine_submit(dma->desc); + ret = dma_submit_error(dma->cookie); + if (ret) { + dev_err(dev, "failed to submit request: %d\n", + ret); + return ret; + } + } + + for (i = channels - 1; i >= 0; i--) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (dma->chan) + dma_async_issue_pending(dma->chan); + } + + ret = stream->compr_ops->start(stream_id); + break; + + case SNDRV_PCM_TRIGGER_STOP: + for (i = channels - 1; i >= 0; i--) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (dma->chan) + dmaengine_terminate_async(dma->chan); + } + + stream->copied_total = 0; + stream->stage1_pointer = 0; + stream->received_total = 0; + stream->received_stage0 = 0; + stream->received_stage1 = 0; + + ret = stream->compr_ops->stop(stream_id); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + for (i = channels - 1; i >= 0; i--) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (dma->chan) + dmaengine_pause(dma->chan); + } + + ret = stream->compr_ops->pause(stream_id); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + for (i = channels - 1; i >= 0; i--) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (dma->chan) + dmaengine_resume(dma->chan); + } + + ret = stream->compr_ops->pause_release(stream_id); + break; + + case SND_COMPR_TRIGGER_PARTIAL_DRAIN: + case SND_COMPR_TRIGGER_DRAIN: + ret = stream->compr_ops->drain(stream->received_total); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int sprd_platform_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct sprd_compr_playinfo *info = + (struct sprd_compr_playinfo *)stream->info_area; + + tstamp->copied_total = stream->copied_total; + tstamp->pcm_io_frames = info->current_data_offset; + + return 0; +} + +static int sprd_platform_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + int avail_bytes, data_count = count; + void *dst; + + /* + * We usually set fragment size as 32K, and the stage 0 IRAM buffer + * size is 32K too. So if now the received data size of the stage 0 + * IRAM buffer is less than 32K, that means we have some available + * spaces for the stage 0 IRAM buffer. + */ + if (stream->received_stage0 < runtime->fragment_size) { + avail_bytes = runtime->fragment_size - stream->received_stage0; + dst = stream->iram_buffer.area + stream->received_stage0; + + if (avail_bytes >= data_count) { + /* + * Copy data to the stage 0 IRAM buffer directly if + * spaces are enough. + */ + if (copy_from_user(dst, buf, data_count)) + return -EFAULT; + + stream->received_stage0 += data_count; + stream->copied_total += data_count; + goto copy_done; + } else { + /* + * If the data count is larger than the available spaces + * of the the stage 0 IRAM buffer, we should copy one + * partial data to the stage 0 IRAM buffer, and copy + * the left to the stage 1 DDR buffer. + */ + if (copy_from_user(dst, buf, avail_bytes)) + return -EFAULT; + + data_count -= avail_bytes; + stream->received_stage0 += avail_bytes; + stream->copied_total += avail_bytes; + buf += avail_bytes; + } + } + + /* + * Copy data to the stage 1 DDR buffer if no spaces for the stage 0 IRAM + * buffer. + */ + dst = stream->compr_buffer.area + stream->stage1_pointer; + if (data_count < stream->compr_buffer.bytes - stream->stage1_pointer) { + if (copy_from_user(dst, buf, data_count)) + return -EFAULT; + + stream->stage1_pointer += data_count; + } else { + avail_bytes = stream->compr_buffer.bytes - stream->stage1_pointer; + + if (copy_from_user(dst, buf, avail_bytes)) + return -EFAULT; + + if (copy_from_user(stream->compr_buffer.area, buf + avail_bytes, + data_count - avail_bytes)) + return -EFAULT; + + stream->stage1_pointer = data_count - avail_bytes; + } + + stream->received_stage1 += data_count; + +copy_done: + /* Update the copied data size. */ + stream->received_total += count; + return count; +} + +static int sprd_platform_compr_get_caps(struct snd_compr_stream *cstream, + struct snd_compr_caps *caps) +{ + caps->direction = cstream->direction; + caps->min_fragment_size = SPRD_COMPR_MIN_FRAGMENT_SIZE; + caps->max_fragment_size = SPRD_COMPR_MAX_FRAGMENT_SIZE; + caps->min_fragments = SPRD_COMPR_MIN_NUM_FRAGMENTS; + caps->max_fragments = SPRD_COMPR_MAX_NUM_FRAGMENTS; + caps->num_codecs = 2; + caps->codecs[0] = SND_AUDIOCODEC_MP3; + caps->codecs[1] = SND_AUDIOCODEC_AAC; + + return 0; +} + +static int +sprd_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + switch (codec->codec) { + case SND_AUDIOCODEC_MP3: + codec->num_descriptors = 2; + codec->descriptor[0].max_ch = 2; + codec->descriptor[0].bit_rate[0] = 320; + codec->descriptor[0].bit_rate[1] = 128; + codec->descriptor[0].num_bitrates = 2; + codec->descriptor[0].profiles = 0; + codec->descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO; + codec->descriptor[0].formats = 0; + break; + + case SND_AUDIOCODEC_AAC: + codec->num_descriptors = 2; + codec->descriptor[1].max_ch = 2; + codec->descriptor[1].bit_rate[0] = 320; + codec->descriptor[1].bit_rate[1] = 128; + codec->descriptor[1].num_bitrates = 2; + codec->descriptor[1].profiles = 0; + codec->descriptor[1].modes = 0; + codec->descriptor[1].formats = 0; + break; + + default: + return -EINVAL; + } + + return 0; +} + +const struct snd_compr_ops sprd_platform_compr_ops = { + .open = sprd_platform_compr_open, + .free = sprd_platform_compr_free, + .set_params = sprd_platform_compr_set_params, + .trigger = sprd_platform_compr_trigger, + .pointer = sprd_platform_compr_pointer, + .copy = sprd_platform_compr_copy, + .get_caps = sprd_platform_compr_get_caps, + .get_codec_caps = sprd_platform_compr_get_codec_caps, +}; + +MODULE_DESCRIPTION("Spreadtrum ASoC Compress Platform Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:compress-platform"); diff --git a/sound/soc/sprd/sprd-pcm-dma.c b/sound/soc/sprd/sprd-pcm-dma.c index cbb27c4abeba..9be6d4b2bf74 100644 --- a/sound/soc/sprd/sprd-pcm-dma.c +++ b/sound/soc/sprd/sprd-pcm-dma.c @@ -13,7 +13,6 @@ #include "sprd-pcm-dma.h" -#define DRV_NAME "sprd_pcm_dma" #define SPRD_PCM_DMA_LINKLIST_SIZE 64 #define SPRD_PCM_DMA_BRUST_LEN 640 @@ -524,6 +523,7 @@ static void sprd_pcm_free(struct snd_pcm *pcm) static const struct snd_soc_component_driver sprd_soc_component = { .name = DRV_NAME, .ops = &sprd_pcm_ops, + .compr_ops = &sprd_platform_compr_ops, .pcm_new = sprd_pcm_new, .pcm_free = sprd_pcm_free, }; diff --git a/sound/soc/sprd/sprd-pcm-dma.h b/sound/soc/sprd/sprd-pcm-dma.h index d85a34f1461d..08e9fdba82f1 100644 --- a/sound/soc/sprd/sprd-pcm-dma.h +++ b/sound/soc/sprd/sprd-pcm-dma.h @@ -3,8 +3,11 @@ #ifndef __SPRD_PCM_DMA_H #define __SPRD_PCM_DMA_H +#define DRV_NAME "sprd_pcm_dma" #define SPRD_PCM_CHANNEL_MAX 2 +extern const struct snd_compr_ops sprd_platform_compr_ops; + struct sprd_pcm_dma_params { dma_addr_t dev_phys[SPRD_PCM_CHANNEL_MAX]; u32 datawidth[SPRD_PCM_CHANNEL_MAX]; @@ -12,4 +15,44 @@ struct sprd_pcm_dma_params { const char *chan_name[SPRD_PCM_CHANNEL_MAX]; }; +struct sprd_compr_playinfo { + int total_time; + int current_time; + int total_data_length; + int current_data_offset; +}; + +struct sprd_compr_params { + u32 direction; + u32 rate; + u32 sample_rate; + u32 channels; + u32 format; + u32 period; + u32 periods; + u32 info_phys; + u32 info_size; +}; + +struct sprd_compr_callback { + void (*drain_notify)(void *data); + void *drain_data; +}; + +struct sprd_compr_ops { + int (*open)(int str_id, struct sprd_compr_callback *cb); + int (*close)(int str_id); + int (*start)(int str_id); + int (*stop)(int str_id); + int (*pause)(int str_id); + int (*pause_release)(int str_id); + int (*drain)(int received_total); + int (*set_params)(int str_id, struct sprd_compr_params *params); +}; + +struct sprd_compr_data { + struct sprd_compr_ops *ops; + struct sprd_pcm_dma_params *dma_params; +}; + #endif /* __SPRD_PCM_DMA_H */ -- cgit v1.2.3-59-g8ed1b From e194098bf9098083f8ae9687924a6878540f8561 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 1 Mar 2019 19:08:51 -0600 Subject: ASoC: core: support driver alias names for FE topology overrides When the same machine driver is reused between platforms but with a different alias, using the driver name is not enough. Add additional fallback case to use the card device name. Tested on GeminiLake with bxt_da7219_max98357a machine driver Suggested-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d88757659729..6f4842977b8d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1974,10 +1974,13 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) continue; /* for this machine ? */ + if (!strcmp(component->driver->ignore_machine, + card->dev->driver->name)) + goto match; if (strcmp(component->driver->ignore_machine, - card->dev->driver->name)) + dev_name(card->dev))) continue; - +match: /* machine matches, so override the rtd data */ for_each_card_prelinks(card, i, dai_link) { -- cgit v1.2.3-59-g8ed1b From 02b6424509e0c8a36d02b0ee36445be8d1fb128a Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 1 Mar 2019 19:08:52 -0600 Subject: ASoC: topology: Align tplg pointer increment across all kcontrols This aligns all kcontrol tplg pointer increments to be consistent in the respective create methods and ensures that the position is pointing to the next widget rather the current invalid widget. Signed-off-by: Liam Girdwood Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index c6a30d000f59..472f7705da93 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -996,8 +996,6 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count, for (i = 0; i < count; i++) { ec = (struct snd_soc_tplg_enum_control *)tplg->pos; - tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - ec->priv.size); /* validate kcontrol */ if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == @@ -1008,6 +1006,9 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count, if (se == NULL) return -ENOMEM; + tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + + ec->priv.size); + dev_dbg(tplg->dev, "ASoC: adding enum kcontrol %s size %d\n", ec->hdr.name, ec->items); @@ -1284,14 +1285,14 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( if (sm == NULL) goto err; - tplg->pos += (sizeof(struct snd_soc_tplg_mixer_control) + - mc->priv.size); - /* validate kcontrol */ if (strnlen(mc->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == SNDRV_CTL_ELEM_ID_NAME_MAXLEN) goto err_str; + tplg->pos += (sizeof(struct snd_soc_tplg_mixer_control) + + mc->priv.size); + dev_dbg(tplg->dev, " adding DAPM widget mixer control %s at %d\n", mc->hdr.name, i); @@ -1377,6 +1378,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( if (se == NULL) goto err; + tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + + ec->priv.size); + dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", ec->hdr.name); @@ -1441,9 +1445,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( ec->hdr.name); goto err_se; } - - tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - ec->priv.size); } return kc; -- cgit v1.2.3-59-g8ed1b From 64b92de9603f22b5455da925ee57268ef7fb4e80 Mon Sep 17 00:00:00 2001 From: Wen Yang Date: Tue, 26 Feb 2019 16:17:49 +0800 Subject: ASoC: wcd9335: fix a leaked reference by adding missing of_node_put The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/codecs/wcd9335.c:5193:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 5183, but without a correspon ding object release within this function. Signed-off-by: Wen Yang Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Srinivas Kandagatla Cc: Vinod Koul Cc: Dan Carpenter (commit_signer:1/11=9%,authored:1/11=9%) Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 981f88a5f615..a04a7cedd99d 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -5188,6 +5188,7 @@ static int wcd9335_slim_status(struct slim_device *sdev, wcd->slim = sdev; wcd->slim_ifc_dev = of_slim_get_device(sdev->ctrl, ifc_dev_np); + of_node_put(ifc_dev_np); if (!wcd->slim_ifc_dev) { dev_err(dev, "Unable to get SLIM Interface device\n"); return -EINVAL; -- cgit v1.2.3-59-g8ed1b From c705247136a523488eac806bd357c3e5d79a7acd Mon Sep 17 00:00:00 2001 From: Wen Yang Date: Tue, 26 Feb 2019 16:17:50 +0800 Subject: ASoC: fsl_utils: fix a leaked reference by adding missing of_node_put The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function. Signed-off-by: Wen Yang Cc: Timur Tabi Cc: Nicolin Chen Cc: Xiubo Li Cc: Fabio Estevam Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: alsa-devel@alsa-project.org Cc: linuxppc-dev@lists.ozlabs.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_utils.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 9981668ab590..040d06b89f00 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -71,6 +71,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, iprop = of_get_property(dma_np, "cell-index", NULL); if (!iprop) { of_node_put(dma_np); + of_node_put(dma_channel_np); return -EINVAL; } *dma_id = be32_to_cpup(iprop); -- cgit v1.2.3-59-g8ed1b From b820d52e7eed7b30b2dfef5f4213a2bc3cbea6f3 Mon Sep 17 00:00:00 2001 From: Wen Yang Date: Tue, 26 Feb 2019 16:17:51 +0800 Subject: ASoC: eukrea-tlv320: fix a leaked reference by adding missing of_node_put The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. ./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. Signed-off-by: Wen Yang Cc: Liam Girdwood Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown --- sound/soc/fsl/eukrea-tlv320.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 191426a6d9ad..30a3d68b5c03 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -118,13 +118,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "fsl,mux-int-port node missing or invalid.\n"); - return ret; + goto err; } ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "fsl,mux-ext-port node missing or invalid.\n"); - return ret; + goto err; } /* -- cgit v1.2.3-59-g8ed1b From 7742a5b418c91b1f27a08ebf90cb662cf4c06604 Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Tue, 2 Apr 2019 19:46:18 +0800 Subject: ASoC: tlv320aic31xx: aic31xx_set_jack() can be static Fixes: ebf3326cd969 ("ASoC: tlv320aic31xx: Add headphone/headset detection") Signed-off-by: kbuild test robot Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index fa47bfcb93e9..9b37e98da0db 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1264,8 +1264,8 @@ static int aic31xx_set_bias_level(struct snd_soc_component *component, return 0; } -int aic31xx_set_jack(struct snd_soc_component *component, - struct snd_soc_jack *jack, void *data) +static int aic31xx_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) { struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); -- cgit v1.2.3-59-g8ed1b From 26ffa016a3c1b34fbfcc8368edb315f2829504ae Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 2 Apr 2019 13:49:15 +0100 Subject: ASoC: wm_adsp: Remove redundant NULL check in wm_adsp_buffer_free wm_adsp_compr_detach is NULL aware so there is no need to check for NULL before calling it, remove the redundant check. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0aa62b26f61c..6145260d89bf 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3887,8 +3887,7 @@ static int wm_adsp_buffer_free(struct wm_adsp *dsp) struct wm_adsp_compr_buf *buf, *tmp; list_for_each_entry_safe(buf, tmp, &dsp->buffer_list, list) { - if (buf->compr) - wm_adsp_compr_detach(buf->compr); + wm_adsp_compr_detach(buf->compr); kfree(buf->name); kfree(buf->regions); -- cgit v1.2.3-59-g8ed1b From 789b930a8f0de609fd2ef9f2ebf73726b7087fea Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 2 Apr 2019 13:49:16 +0100 Subject: ASoC: wm_adsp: Make use of local variables Tidy up some instances of dereferencing to obtain things that are already stored in local variables. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6145260d89bf..4662a20cfb92 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3381,7 +3381,7 @@ static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) return -EINVAL; compr->buf = buf; - compr->buf->compr = compr; + buf->compr = compr; return 0; } @@ -4091,7 +4091,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, buf = compr->buf; - if (!compr->buf || compr->buf->error) { + if (!buf || buf->error) { snd_compr_stop_error(stream, SNDRV_PCM_STATE_XRUN); ret = -EIO; goto out; @@ -4111,7 +4111,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, if (buf->avail < wm_adsp_compr_frag_words(compr)) { ret = wm_adsp_buffer_get_error(buf); if (ret < 0) { - if (compr->buf->error) + if (buf->error) snd_compr_stop_error(stream, SNDRV_PCM_STATE_XRUN); goto out; -- cgit v1.2.3-59-g8ed1b From 2b7bcdad3af9fbd104d5f300dbb90f684c36aa09 Mon Sep 17 00:00:00 2001 From: Anders Roxell Date: Wed, 3 Apr 2019 05:05:01 +0200 Subject: ASoC: Mediatek: MT8183: TS3A227: fix build error MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When building CONFIG_SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A=m the following error pops up: ../sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c: In function ‘mt8183_mt6358_ts3a227_max98357_dev_probe’: ../sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:325:13: error: ‘struct snd_soc_dai_link’ has no member named ‘platform’; did you mean ‘platforms’? dai_link->platform = NULL; ^~~~~~~~ platforms Rework to use 'dai_link->platforms' instead of 'dai_link->platform'. Fixes: 11c0269017b2 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227") Signed-off-by: Anders Roxell Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 2da56232a9e1..a50c6af5c945 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -322,7 +322,7 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) * the "platform" will not null when probe is trying * again. It's not expected normally. */ - dai_link->platform = NULL; + dai_link->platforms = NULL; if (dai_link->platform_name) continue; -- cgit v1.2.3-59-g8ed1b From 52a30f3bb1b15c7d27233e9947370c1c14b5cbae Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Wed, 3 Apr 2019 12:15:52 +0800 Subject: ASoC: Mediatek: MT8183: Fix build error in mt8183_da7219_max98357_dev_probe When building CONFIG_SND_SOC_MT8183_DA7219_MAX98357A=m gcc warn this: sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c: In function mt8183_da7219_max98357_dev_probe: sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c:413:13: error: struct snd_soc_dai_link has no member named platform; did you mean platforms? dai_link->platform = NULL; ^~~~~~~~ platforms use 'dai_link->platforms' instead of 'dai_link->platform'. Fixes: 11c0269017b2 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227") Signed-off-by: YueHaibing Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 93536659a859..1e7e8aed8707 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -410,7 +410,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) * the "platform" will not null when probe is trying * again. It's not expected normally. */ - dai_link->platform = NULL; + dai_link->platforms = NULL; if (dai_link->platform_name) continue; -- cgit v1.2.3-59-g8ed1b From 0f9e47b3fd92c809a8c68719cd60ba863c4c48e0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Apr 2019 11:55:40 +0700 Subject: ASoC: sprd: Fix modular build Reported-by: Stephen Rothwell Suggested-by: Baolin Wang Tested-by: Baolin Wang Signed-off-by: Mark Brown --- sound/soc/sprd/Makefile | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sprd/Makefile b/sound/soc/sprd/Makefile index fa1d5476b8da..e6c2606da8e2 100644 --- a/sound/soc/sprd/Makefile +++ b/sound/soc/sprd/Makefile @@ -1,4 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 # Spreadtrum Audio Support -obj-$(CONFIG_SND_SOC_SPRD) += sprd-pcm-dma.o sprd-pcm-compress.o +snd-soc-sprd-platform-objs := sprd-pcm-dma.o sprd-pcm-compress.o + +obj-$(CONFIG_SND_SOC_SPRD) += snd-soc-sprd-platform.o -- cgit v1.2.3-59-g8ed1b From b6c573e5e8652330525d61b583203e7b71a09c53 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Wed, 3 Apr 2019 07:11:39 +0000 Subject: ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index a50c6af5c945..681bedb524f9 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -364,7 +364,6 @@ static const struct of_device_id mt8183_mt6358_ts3a227_max98357_dt_match[] = { static struct platform_driver mt8183_mt6358_ts3a227_max98357_driver = { .driver = { .name = "mt8183_mt6358_ts3a227_max98357", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt8183_mt6358_ts3a227_max98357_dt_match, #endif -- cgit v1.2.3-59-g8ed1b From d82e78b0a578a712629cdb6cb7bfc66fb3215306 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Wed, 3 Apr 2019 07:11:59 +0000 Subject: ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 1e7e8aed8707..31ea8632c397 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -454,7 +454,6 @@ static const struct of_device_id mt8183_da7219_max98357_dt_match[] = { static struct platform_driver mt8183_da7219_max98357_driver = { .driver = { .name = "mt8183_da7219_max98357", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt8183_da7219_max98357_dt_match, #endif -- cgit v1.2.3-59-g8ed1b From adcc9f2b064e60f569979fb1de1f158e8bed8a6d Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Wed, 3 Apr 2019 00:47:55 -0700 Subject: ASoC: sprd: Fix the smatch warning Remove the unnecessary validation of the 'cstream' variable to fix below smatch warning: sprd_platform_compr_drain_notify() warn: variable dereferenced before check 'cstream' (see line 105) Reported-by: Dan Carpenter Signed-off-by: Baolin Wang Signed-off-by: Mark Brown --- sound/soc/sprd/sprd-pcm-compress.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c index dc84257cffc2..424e9834d093 100644 --- a/sound/soc/sprd/sprd-pcm-compress.c +++ b/sound/soc/sprd/sprd-pcm-compress.c @@ -107,8 +107,7 @@ static void sprd_platform_compr_drain_notify(void *arg) memset(stream->info_area, 0, sizeof(struct sprd_compr_playinfo)); - if (cstream) - snd_compr_drain_notify(cstream); + snd_compr_drain_notify(cstream); } static void sprd_platform_compr_dma_complete(void *data) -- cgit v1.2.3-59-g8ed1b From 8a1d95c393d971e624fc28f11516b0bc3a7fa706 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Wed, 3 Apr 2019 21:17:16 -0700 Subject: ASoC: tlv320aic32x4: Update copyright and use SPDX identifier Update the copyright dates and use the SPDX identifier instead of reciting the license. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4-i2c.c | 14 ++------------ sound/soc/codecs/tlv320aic32x4-spi.c | 14 ++------------ 2 files changed, 4 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c index 385fa2e9525a..d8bede5c1f38 100644 --- a/sound/soc/codecs/tlv320aic32x4-i2c.c +++ b/sound/soc/codecs/tlv320aic32x4-i2c.c @@ -1,21 +1,11 @@ -/* - * linux/sound/soc/codecs/tlv320aic32x4-i2c.c +/* SPDX-License-Identifier: GPL-2.0 * - * Copyright 2011 NW Digital Radio + * Copyright 2011-2019 NW Digital Radio * * Author: Jeremy McDermond * * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. */ #include diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c index 07d78ae51e05..a72a61c4d785 100644 --- a/sound/soc/codecs/tlv320aic32x4-spi.c +++ b/sound/soc/codecs/tlv320aic32x4-spi.c @@ -1,21 +1,11 @@ -/* - * linux/sound/soc/codecs/tlv320aic32x4-spi.c +/* SPDX-License-Identifier: GPL-2.0 * - * Copyright 2011 NW Digital Radio + * Copyright 2011-2019 NW Digital Radio * * Author: Jeremy McDermond * * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. */ #include -- cgit v1.2.3-59-g8ed1b From fb20de606316a446efdfb0f775a892f782165783 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Wed, 3 Apr 2019 21:40:46 +0800 Subject: ASoC: rockchip: pdm: using left justified store mode This patch set left justified store mode default. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 1 + sound/soc/rockchip/rockchip_pdm.h | 3 +++ 2 files changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 400e29edb1c9..ba61a4b27d39 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -135,6 +135,7 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(pdm->regmap, PDM_HPF_CTRL, PDM_HPF_LE | PDM_HPF_RE, PDM_HPF_LE | PDM_HPF_RE); regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, PDM_CLK_EN, PDM_CLK_EN); + regmap_update_bits(pdm->regmap, PDM_CTRL0, PDM_MODE_MSK, PDM_MODE_LJ); val = 0; switch (params_format(params)) { diff --git a/sound/soc/rockchip/rockchip_pdm.h b/sound/soc/rockchip/rockchip_pdm.h index 886b48d128fd..00a8fa187d22 100644 --- a/sound/soc/rockchip/rockchip_pdm.h +++ b/sound/soc/rockchip/rockchip_pdm.h @@ -42,6 +42,9 @@ /* PDM CTRL0 */ #define PDM_PATH_MSK (0xf << 27) +#define PDM_MODE_MSK BIT(31) +#define PDM_MODE_RJ 0 +#define PDM_MODE_LJ BIT(31) #define PDM_PATH3_EN BIT(30) #define PDM_PATH2_EN BIT(29) #define PDM_PATH1_EN BIT(28) -- cgit v1.2.3-59-g8ed1b From a7980cd212cef9002fe9683f440d27609d1aae37 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Wed, 3 Apr 2019 21:40:47 +0800 Subject: ASoC: rockchip: pdm: add default regs This patch add default regs value for controller. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index ba61a4b27d39..984d99889a65 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -361,11 +361,19 @@ static bool rockchip_pdm_volatile_reg(struct device *dev, unsigned int reg) } } +static const struct reg_default rockchip_pdm_reg_defaults[] = { + {0x04, 0x78000017}, + {0x08, 0x0bb8ea60}, + {0x18, 0x0000001f}, +}; + static const struct regmap_config rockchip_pdm_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, .max_register = PDM_VERSION, + .reg_defaults = rockchip_pdm_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rockchip_pdm_reg_defaults), .writeable_reg = rockchip_pdm_wr_reg, .readable_reg = rockchip_pdm_rd_reg, .volatile_reg = rockchip_pdm_volatile_reg, -- cgit v1.2.3-59-g8ed1b From 252163a66a06859f468ef1e00fa5a1f377b16eed Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Wed, 3 Apr 2019 21:40:48 +0800 Subject: ASoC: rockchip: pdm: optimize clear logic There is no need to reset controller every time, do this once in pdm_probe. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 984d99889a65..eb5216dd7dbc 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -183,9 +183,6 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, val); regmap_update_bits(pdm->regmap, PDM_DMA_CTRL, PDM_DMA_RDL_MSK, PDM_DMA_RDL(16)); - regmap_update_bits(pdm->regmap, PDM_SYSCONFIG, - PDM_RX_MASK | PDM_RX_CLR_MASK, - PDM_RX_STOP | PDM_RX_CLR_WR); } return 0; @@ -353,6 +350,7 @@ static bool rockchip_pdm_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case PDM_SYSCONFIG: + case PDM_FIFO_CTRL: case PDM_INT_CLR: case PDM_INT_ST: return true; @@ -436,6 +434,7 @@ static int rockchip_pdm_probe(struct platform_device *pdev) goto err_suspend; } + rockchip_pdm_rxctrl(pdm, 0); ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "could not register pcm: %d\n", ret); -- cgit v1.2.3-59-g8ed1b From 624e8e00acafe3d31a7c31e67fa95ce06e324bf8 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Thu, 4 Apr 2019 11:51:09 +0800 Subject: ASoC: rockchip: pdm: fixup pdm fractional div This patch adds support fractional div for rk3308. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 172 ++++++++++++++++++++++++++++---------- sound/soc/rockchip/rockchip_pdm.h | 9 ++ 2 files changed, 139 insertions(+), 42 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index c50494b0ed0d..4f93a7454e85 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -17,14 +17,23 @@ #include #include #include +#include #include +#include #include +#include #include #include #include "rockchip_pdm.h" #define PDM_DMA_BURST_SIZE (8) /* size * width: 8*4 = 32 bytes */ +#define PDM_SIGNOFF_CLK_RATE (100000000) + +enum rk_pdm_version { + RK_PDM_RK3229, + RK_PDM_RK3308, +}; struct rk_pdm_dev { struct device *dev; @@ -32,22 +41,51 @@ struct rk_pdm_dev { struct clk *hclk; struct regmap *regmap; struct snd_dmaengine_dai_dma_data capture_dma_data; + struct reset_control *reset; + enum rk_pdm_version version; }; struct rk_pdm_clkref { unsigned int sr; unsigned int clk; + unsigned int clk_out; +}; + +struct rk_pdm_ds_ratio { + unsigned int ratio; + unsigned int sr; }; static struct rk_pdm_clkref clkref[] = { - { 8000, 40960000 }, - { 11025, 56448000 }, - { 12000, 61440000 }, + { 8000, 40960000, 2048000 }, + { 11025, 56448000, 2822400 }, + { 12000, 61440000, 3072000 }, + { 8000, 98304000, 2048000 }, + { 12000, 98304000, 3072000 }, +}; + +static struct rk_pdm_ds_ratio ds_ratio[] = { + { 0, 192000 }, + { 0, 176400 }, + { 0, 128000 }, + { 1, 96000 }, + { 1, 88200 }, + { 1, 64000 }, + { 2, 48000 }, + { 2, 44100 }, + { 2, 32000 }, + { 3, 24000 }, + { 3, 22050 }, + { 3, 16000 }, + { 4, 12000 }, + { 4, 11025 }, + { 4, 8000 }, }; -static unsigned int get_pdm_clk(unsigned int sr) +static unsigned int get_pdm_clk(struct rk_pdm_dev *pdm, unsigned int sr, + unsigned int *clk_src, unsigned int *clk_out) { - unsigned int i, count, clk, div; + unsigned int i, count, clk, div, rate; clk = 0; if (!sr) @@ -59,14 +97,39 @@ static unsigned int get_pdm_clk(unsigned int sr) continue; div = sr / clkref[i].sr; if ((div & (div - 1)) == 0) { + *clk_out = clkref[i].clk_out; + rate = clk_round_rate(pdm->clk, clkref[i].clk); + if (rate != clkref[i].clk) + continue; clk = clkref[i].clk; + *clk_src = clkref[i].clk; break; } } + if (!clk) { + clk = clk_round_rate(pdm->clk, PDM_SIGNOFF_CLK_RATE); + *clk_src = clk; + } return clk; } +static unsigned int get_pdm_ds_ratio(unsigned int sr) +{ + unsigned int i, count, ratio; + + ratio = 0; + if (!sr) + return ratio; + + count = ARRAY_SIZE(ds_ratio); + for (i = 0; i < count; i++) { + if (sr == ds_ratio[i].sr) + ratio = ds_ratio[i].ratio; + } + return ratio; +} + static inline struct rk_pdm_dev *to_info(struct snd_soc_dai *dai) { return snd_soc_dai_get_drvdata(dai); @@ -95,40 +158,52 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, struct rk_pdm_dev *pdm = to_info(dai); unsigned int val = 0; unsigned int clk_rate, clk_div, samplerate; + unsigned int clk_src, clk_out; + unsigned long m, n; + bool change; int ret; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return 0; + samplerate = params_rate(params); - clk_rate = get_pdm_clk(samplerate); + clk_rate = get_pdm_clk(pdm, samplerate, &clk_src, &clk_out); if (!clk_rate) return -EINVAL; - ret = clk_set_rate(pdm->clk, clk_rate); + ret = clk_set_rate(pdm->clk, clk_src); if (ret) return -EINVAL; - clk_div = DIV_ROUND_CLOSEST(clk_rate, samplerate); - - switch (clk_div) { - case 320: - val = PDM_CLK_320FS; - break; - case 640: - val = PDM_CLK_640FS; - break; - case 1280: - val = PDM_CLK_1280FS; - break; - case 2560: - val = PDM_CLK_2560FS; - break; - case 5120: - val = PDM_CLK_5120FS; - break; - default: - dev_err(pdm->dev, "unsupported div: %d\n", clk_div); - return -EINVAL; + if (pdm->version == RK_PDM_RK3308) { + rational_best_approximation(clk_out, clk_src, + GENMASK(16 - 1, 0), + GENMASK(16 - 1, 0), + &m, &n); + + val = (m << PDM_FD_NUMERATOR_SFT) | + (n << PDM_FD_DENOMINATOR_SFT); + regmap_update_bits_check(pdm->regmap, PDM_CTRL1, + PDM_FD_NUMERATOR_MSK | + PDM_FD_DENOMINATOR_MSK, + val, &change); + if (change) { + reset_control_assert(pdm->reset); + reset_control_deassert(pdm->reset); + rockchip_pdm_rxctrl(pdm, 0); + } + clk_div = n / m; + if (clk_div >= 40) + val = PDM_CLK_FD_RATIO_40; + else if (clk_div <= 35) + val = PDM_CLK_FD_RATIO_35; + else + return -EINVAL; + regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, + PDM_CLK_FD_RATIO_MSK, + val); } - + val = get_pdm_ds_ratio(samplerate); regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, PDM_DS_RATIO_MSK, val); regmap_update_bits(pdm->regmap, PDM_HPF_CTRL, PDM_HPF_CF_MSK, PDM_HPF_60HZ); @@ -177,13 +252,11 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - regmap_update_bits(pdm->regmap, PDM_CTRL0, - PDM_PATH_MSK | PDM_VDW_MSK, - val); - regmap_update_bits(pdm->regmap, PDM_DMA_CTRL, PDM_DMA_RDL_MSK, - PDM_DMA_RDL(16)); - } + regmap_update_bits(pdm->regmap, PDM_CTRL0, + PDM_PATH_MSK | PDM_VDW_MSK, + val); + regmap_update_bits(pdm->regmap, PDM_DMA_CTRL, PDM_DMA_RDL_MSK, + PDM_DMA_RDL(16)); return 0; } @@ -380,8 +453,19 @@ static const struct regmap_config rockchip_pdm_regmap_config = { .cache_type = REGCACHE_FLAT, }; +static const struct of_device_id rockchip_pdm_match[] = { + { .compatible = "rockchip,pdm", }, + { .compatible = "rockchip,px30-pdm", + .data = (void *)RK_PDM_RK3308 }, + { .compatible = "rockchip,rk3308-pdm", + .data = (void *)RK_PDM_RK3308 }, + {}, +}; +MODULE_DEVICE_TABLE(of, rockchip_pdm_match); + static int rockchip_pdm_probe(struct platform_device *pdev) { + const struct of_device_id *match; struct rk_pdm_dev *pdm; struct resource *res; void __iomem *regs; @@ -391,6 +475,16 @@ static int rockchip_pdm_probe(struct platform_device *pdev) if (!pdm) return -ENOMEM; + match = of_match_device(rockchip_pdm_match, &pdev->dev); + if (match) + pdm->version = (enum rk_pdm_version)match->data; + + if (pdm->version == RK_PDM_RK3308) { + pdm->reset = devm_reset_control_get(&pdev->dev, "pdm-m"); + if (IS_ERR(pdm->reset)) + return PTR_ERR(pdm->reset); + } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(regs)) @@ -503,12 +597,6 @@ static const struct dev_pm_ops rockchip_pdm_pm_ops = { SET_SYSTEM_SLEEP_PM_OPS(rockchip_pdm_suspend, rockchip_pdm_resume) }; -static const struct of_device_id rockchip_pdm_match[] = { - { .compatible = "rockchip,pdm", }, - {}, -}; -MODULE_DEVICE_TABLE(of, rockchip_pdm_match); - static struct platform_driver rockchip_pdm_driver = { .probe = rockchip_pdm_probe, .remove = rockchip_pdm_remove, diff --git a/sound/soc/rockchip/rockchip_pdm.h b/sound/soc/rockchip/rockchip_pdm.h index 00a8fa187d22..ae88644aa334 100644 --- a/sound/soc/rockchip/rockchip_pdm.h +++ b/sound/soc/rockchip/rockchip_pdm.h @@ -53,7 +53,16 @@ #define PDM_VDW_MSK (0x1f << 0) #define PDM_VDW(X) ((X - 1) << 0) +/* PDM CTRL1 */ +#define PDM_FD_NUMERATOR_SFT 16 +#define PDM_FD_NUMERATOR_MSK GENMASK(31, 16) +#define PDM_FD_DENOMINATOR_SFT 0 +#define PDM_FD_DENOMINATOR_MSK GENMASK(15, 0) + /* PDM CLK CTRL */ +#define PDM_CLK_FD_RATIO_MSK BIT(6) +#define PDM_CLK_FD_RATIO_40 (0X0 << 6) +#define PDM_CLK_FD_RATIO_35 BIT(6) #define PDM_CLK_MSK BIT(5) #define PDM_CLK_EN BIT(5) #define PDM_CLK_DIS (0x0 << 5) -- cgit v1.2.3-59-g8ed1b From 717d97879b346d56fc47a6e3fe0ad44fea9949b2 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Thu, 4 Apr 2019 11:54:30 +0800 Subject: ASoC: rockchip: pdm: add compatible for rk1808 This patch adds support for rk1808, the pdm controller is the same as rk3308. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 4f93a7454e85..3e1c5fd5b32b 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -457,6 +457,8 @@ static const struct of_device_id rockchip_pdm_match[] = { { .compatible = "rockchip,pdm", }, { .compatible = "rockchip,px30-pdm", .data = (void *)RK_PDM_RK3308 }, + { .compatible = "rockchip,rk1808-pdm", + .data = (void *)RK_PDM_RK3308 }, { .compatible = "rockchip,rk3308-pdm", .data = (void *)RK_PDM_RK3308 }, {}, -- cgit v1.2.3-59-g8ed1b From 430f5da69b9323d5e7faca95bc30e61be4fcbbbf Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Thu, 4 Apr 2019 11:56:29 +0800 Subject: ASoC: rockchip: pdm: adjust waterlevel in frame unit This patch make the waterlevel more reasonable, because the pdm controller share the single FIFO(128 entries) with each channel. adjust waterlevel in frame to meet the vad or dma frames request. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 3e1c5fd5b32b..955cdc2b8fc5 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -255,8 +255,9 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(pdm->regmap, PDM_CTRL0, PDM_PATH_MSK | PDM_VDW_MSK, val); + /* all channels share the single FIFO */ regmap_update_bits(pdm->regmap, PDM_DMA_CTRL, PDM_DMA_RDL_MSK, - PDM_DMA_RDL(16)); + PDM_DMA_RDL(8 * params_channels(params))); return 0; } -- cgit v1.2.3-59-g8ed1b From 54cd97cfe02e9bb93db53e7cbfa9b672bf4a5ac8 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Thu, 4 Apr 2019 11:57:57 +0800 Subject: ASoC: rockchip: pdm: Correct PDM_CTRL0 reg value This patch fix the wrong reg value for rk322x/rk322xh, cuz there is no STORE JUSTIFIED MODE on it. on rk322x/rk322xh, the same bit means PDM_MODE/RESERVED, if the bit is set to RESERVED, the controller will not work. Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 955cdc2b8fc5..068453596470 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -210,7 +210,9 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(pdm->regmap, PDM_HPF_CTRL, PDM_HPF_LE | PDM_HPF_RE, PDM_HPF_LE | PDM_HPF_RE); regmap_update_bits(pdm->regmap, PDM_CLK_CTRL, PDM_CLK_EN, PDM_CLK_EN); - regmap_update_bits(pdm->regmap, PDM_CTRL0, PDM_MODE_MSK, PDM_MODE_LJ); + if (pdm->version != RK_PDM_RK3229) + regmap_update_bits(pdm->regmap, PDM_CTRL0, + PDM_MODE_MSK, PDM_MODE_LJ); val = 0; switch (params_format(params)) { @@ -455,7 +457,8 @@ static const struct regmap_config rockchip_pdm_regmap_config = { }; static const struct of_device_id rockchip_pdm_match[] = { - { .compatible = "rockchip,pdm", }, + { .compatible = "rockchip,pdm", + .data = (void *)RK_PDM_RK3229 }, { .compatible = "rockchip,px30-pdm", .data = (void *)RK_PDM_RK3308 }, { .compatible = "rockchip,rk1808-pdm", -- cgit v1.2.3-59-g8ed1b From 47fa5773f11c7cc9117d74ea5a18a22502d8f0f3 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 4 Apr 2019 08:56:03 +0100 Subject: ASoC: sprd: Fix spelling mistake "faied" -> "failed" There are two identical spelling mistakes in dev_err messages. Fix them. Signed-off-by: Colin Ian King Reviewed-by: Mukesh Ojha Reviewed-by: Baolin Wang Signed-off-by: Mark Brown --- sound/soc/sprd/sprd-pcm-compress.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c index 424e9834d093..6cddf551bc11 100644 --- a/sound/soc/sprd/sprd-pcm-compress.c +++ b/sound/soc/sprd/sprd-pcm-compress.c @@ -281,13 +281,13 @@ static int sprd_platform_compr_set_params(struct snd_compr_stream *cstream, */ ret = sprd_platform_compr_dma_config(cstream, params, 1); if (ret) { - dev_err(dev, "faied to config stage 1 DMA: %d\n", ret); + dev_err(dev, "failed to config stage 1 DMA: %d\n", ret); return ret; } ret = sprd_platform_compr_dma_config(cstream, params, 0); if (ret) { - dev_err(dev, "faied to config stage 0 DMA: %d\n", ret); + dev_err(dev, "failed to config stage 0 DMA: %d\n", ret); goto config_err; } -- cgit v1.2.3-59-g8ed1b From b126fc07328749269021ac53173f9afdd6947248 Mon Sep 17 00:00:00 2001 From: Sugar Zhang Date: Thu, 4 Apr 2019 11:57:08 +0800 Subject: ASoC: rockchip: pdm: Mark RXFIFO_DATA as volatile and precious This patch marks RXFIFO_DATA as precious to avoid being read outside a call from the driver, such as regmap debugfs Signed-off-by: Sugar Zhang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 068453596470..6c0f242db5ef 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -417,6 +417,7 @@ static bool rockchip_pdm_rd_reg(struct device *dev, unsigned int reg) case PDM_INT_CLR: case PDM_INT_ST: case PDM_DATA_VALID: + case PDM_RXFIFO_DATA: case PDM_VERSION: return true; default: @@ -431,6 +432,17 @@ static bool rockchip_pdm_volatile_reg(struct device *dev, unsigned int reg) case PDM_FIFO_CTRL: case PDM_INT_CLR: case PDM_INT_ST: + case PDM_RXFIFO_DATA: + return true; + default: + return false; + } +} + +static bool rockchip_pdm_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PDM_RXFIFO_DATA: return true; default: return false; @@ -453,6 +465,7 @@ static const struct regmap_config rockchip_pdm_regmap_config = { .writeable_reg = rockchip_pdm_wr_reg, .readable_reg = rockchip_pdm_rd_reg, .volatile_reg = rockchip_pdm_volatile_reg, + .precious_reg = rockchip_pdm_precious_reg, .cache_type = REGCACHE_FLAT, }; -- cgit v1.2.3-59-g8ed1b From 44ceee847e27c828f2f1ef4e400e6bc0c8d04de3 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Wed, 3 Apr 2019 21:01:54 -0700 Subject: ASoC: tlv320aic32x4: Add Switch for Setting Common Mode Voltage Add a switch for setting common mode voltage. This can allow for higher drive levels on the amplifier outputs. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index aec86fd6ac15..77df2cbdf5f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -242,6 +242,12 @@ static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0); /* -12dB min, 0.5dB steps */ static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0); +static const char * const lo_cm_text[] = { + "Full Chip", "1.65V", +}; + +static SOC_ENUM_SINGLE_DECL(lo_cm_enum, AIC32X4_CMMODE, 3, lo_cm_text); + static const struct snd_kcontrol_new aic32x4_snd_controls[] = { SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL, AIC32X4_RDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm), @@ -255,6 +261,7 @@ static const struct snd_kcontrol_new aic32x4_snd_controls[] = { AIC32X4_HPRGAIN, 6, 0x01, 1), SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN, AIC32X4_LORGAIN, 6, 0x01, 1), + SOC_ENUM("LO Playback Common Mode Switch", lo_cm_enum), SOC_DOUBLE_R("Mic PGA Switch", AIC32X4_LMICPGAVOL, AIC32X4_RMICPGAVOL, 7, 0x01, 1), -- cgit v1.2.3-59-g8ed1b From d3e6e374566e1154820a9a3dc82f7eef646fcf95 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Wed, 3 Apr 2019 21:01:55 -0700 Subject: ASoC: tlv320aic32x4: Add Playback PowerTune Controls PowerTune controls the power level of the chip. On playback this indirectly controls things like the gain of the various output amplifiers. This can allow for the decrease of output levels from the codec. This adds controls for those power levels to the driver. Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 9 +++++++++ sound/soc/codecs/tlv320aic32x4.h | 2 ++ 2 files changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 77df2cbdf5f4..6edee05ff9f0 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -248,9 +248,18 @@ static const char * const lo_cm_text[] = { static SOC_ENUM_SINGLE_DECL(lo_cm_enum, AIC32X4_CMMODE, 3, lo_cm_text); +static const char * const ptm_text[] = { + "P3", "P2", "P1", +}; + +static SOC_ENUM_SINGLE_DECL(l_ptm_enum, AIC32X4_LPLAYBACK, 2, ptm_text); +static SOC_ENUM_SINGLE_DECL(r_ptm_enum, AIC32X4_RPLAYBACK, 2, ptm_text); + static const struct snd_kcontrol_new aic32x4_snd_controls[] = { SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL, AIC32X4_RDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm), + SOC_ENUM("DAC Left Playback PowerTune Switch", l_ptm_enum), + SOC_ENUM("DAC Right Playback PowerTune Switch", r_ptm_enum), SOC_DOUBLE_R_S_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN, AIC32X4_HPRGAIN, 0, -0x6, 0x1d, 5, 0, tlv_driver_gain), diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index 88205bc97198..40734211bc0e 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -78,6 +78,8 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name); #define AIC32X4_PWRCFG AIC32X4_REG(1, 1) #define AIC32X4_LDOCTL AIC32X4_REG(1, 2) +#define AIC32X4_LPLAYBACK AIC32X4_REG(1, 3) +#define AIC32X4_RPLAYBACK AIC32X4_REG(1, 4) #define AIC32X4_OUTPWRCTL AIC32X4_REG(1, 9) #define AIC32X4_CMMODE AIC32X4_REG(1, 10) #define AIC32X4_HPLROUTE AIC32X4_REG(1, 12) -- cgit v1.2.3-59-g8ed1b From fdf34366d3242d5eeffa1b4d9a3497ebf30a4ecb Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 4 Apr 2019 13:50:15 +0200 Subject: ASoC: max98357a: add missing supported rates According the publicly available datasheet (and some test) the max98357a also supports 32, 44.1 and 88.2 kHz sample rate. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index d469576b5a7b..d037a3e4d323 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -97,7 +97,10 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { SNDRV_PCM_FMTBIT_S32, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 8000, .rate_max = 96000, -- cgit v1.2.3-59-g8ed1b From aa612f2b006aa3552871dabcd6a8e90e33f65e09 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 4 Apr 2019 13:56:01 +0100 Subject: ASoC: wm_adsp: Avoid calling snd_compr_stop_error from WDT expiry It is unsafe to call snd_compr_stop_error from outside of the compressed ops. Firstly the compressed device lock needs to be held and secondly it queues error work to issue a trigger stop which should not happen after the stream has been freed. To avoid these issues use the same trick used for the IRQ handling, simply send a snd_compr_fragment_elapsed to cause user-space to wake on the poll, then report the error when user-space issues the pointer request after it wakes. Fixes: a2bcbc1b9ac2f ("ASoC: wm_adsp: Shutdown any compressed streams on DSP watchdog timeout") Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm_adsp.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index c8c49d5b8ac9..a9298bfddd9c 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -4092,7 +4092,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, buf = compr->buf; - if (!buf || buf->error) { + if (dsp->fatal_error || !buf || buf->error) { snd_compr_stop_error(stream, SNDRV_PCM_STATE_XRUN); ret = -EIO; goto out; @@ -4196,12 +4196,13 @@ static int wm_adsp_buffer_capture_block(struct wm_adsp_compr *compr, int target) static int wm_adsp_compr_read(struct wm_adsp_compr *compr, char __user *buf, size_t count) { + struct wm_adsp *dsp = compr->dsp; int ntotal = 0; int nwords, nbytes; compr_dbg(compr, "Requested read of %zu bytes\n", count); - if (!compr->buf || compr->buf->error) { + if (dsp->fatal_error || !compr->buf || compr->buf->error) { snd_compr_stop_error(compr->stream, SNDRV_PCM_STATE_XRUN); return -EIO; } @@ -4262,11 +4263,8 @@ static void wm_adsp_fatal_error(struct wm_adsp *dsp) dsp->fatal_error = true; list_for_each_entry(compr, &dsp->compr_list, list) { - if (compr->stream) { - snd_compr_stop_error(compr->stream, - SNDRV_PCM_STATE_XRUN); + if (compr->stream) snd_compr_fragment_elapsed(compr->stream); - } } } -- cgit v1.2.3-59-g8ed1b From f77b6ea70f3a672092e2d7eca2e67be09c0b8c54 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 3 Apr 2019 15:23:32 +0200 Subject: ASoC: cs42l51: add regulator management Add cs42l51 audio codec power supply management through regulator framework. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51-i2c.c | 6 ++++++ sound/soc/codecs/cs42l51.c | 44 +++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/cs42l51.h | 1 + 3 files changed, 50 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 4b5731a41876..8333dbf18ea2 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -35,12 +35,18 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c, return cs42l51_probe(&i2c->dev, devm_regmap_init_i2c(i2c, &config)); } +static int cs42l51_i2c_remove(struct i2c_client *i2c) +{ + return cs42l51_remove(&i2c->dev); +} + static struct i2c_driver cs42l51_i2c_driver = { .driver = { .name = "cs42l51", .of_match_table = cs42l51_of_match, }, .probe = cs42l51_i2c_probe, + .remove = cs42l51_i2c_remove, .id_table = cs42l51_i2c_id, }; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 80da3cd73e04..f43eb51d2d8d 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -31,6 +31,7 @@ #include #include #include +#include #include "cs42l51.h" @@ -40,11 +41,19 @@ enum master_slave_mode { MODE_MASTER, }; +static const char * const cs42l51_supply_names[] = { + "VL", + "VD", + "VA", + "VAHP", +}; + struct cs42l51_private { unsigned int mclk; struct clk *mclk_handle; unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; + struct regulator_bulk_data supplies[ARRAY_SIZE(cs42l51_supply_names)]; }; #define CS42L51_FORMATS ( \ @@ -550,7 +559,7 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) { struct cs42l51_private *cs42l51; unsigned int val; - int ret; + int ret, i; if (IS_ERR(regmap)) return PTR_ERR(regmap); @@ -569,6 +578,23 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) cs42l51->mclk_handle = NULL; } + for (i = 0; i < ARRAY_SIZE(cs42l51->supplies); i++) + cs42l51->supplies[i].supply = cs42l51_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(cs42l51->supplies), + cs42l51->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42l51->supplies), + cs42l51->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + /* Verify that we have a CS42L51 */ ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { @@ -587,11 +613,27 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) ret = devm_snd_soc_register_component(dev, &soc_component_device_cs42l51, &cs42l51_dai, 1); + if (ret < 0) + goto error; + + return 0; + error: + regulator_bulk_disable(ARRAY_SIZE(cs42l51->supplies), + cs42l51->supplies); return ret; } EXPORT_SYMBOL_GPL(cs42l51_probe); +int cs42l51_remove(struct device *dev) +{ + struct cs42l51_private *cs42l51 = dev_get_drvdata(dev); + + return regulator_bulk_disable(ARRAY_SIZE(cs42l51->supplies), + cs42l51->supplies); +} +EXPORT_SYMBOL_GPL(cs42l51_remove); + const struct of_device_id cs42l51_of_match[] = { { .compatible = "cirrus,cs42l51", }, { } diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 0ca805492ac4..aef0ede82c7b 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -22,6 +22,7 @@ struct device; extern const struct regmap_config cs42l51_regmap; int cs42l51_probe(struct device *dev, struct regmap *regmap); +int cs42l51_remove(struct device *dev); extern const struct of_device_id cs42l51_of_match[]; #define CS42L51_CHIP_ID 0x1B -- cgit v1.2.3-59-g8ed1b From 11b9cd748e3107df5c5e040a4daa986a3c6d8b75 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 3 Apr 2019 15:23:33 +0200 Subject: ASoC: cs42l51: add reset management Manage cs42l51 audio codec reset pin. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index f43eb51d2d8d..9b3ffa16b204 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -30,6 +30,7 @@ #include #include #include +#include #include #include @@ -54,6 +55,7 @@ struct cs42l51_private { unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; struct regulator_bulk_data supplies[ARRAY_SIZE(cs42l51_supply_names)]; + struct gpio_desc *reset_gpio; }; #define CS42L51_FORMATS ( \ @@ -595,6 +597,17 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) return ret; } + cs42l51->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs42l51->reset_gpio)) + return PTR_ERR(cs42l51->reset_gpio); + + if (cs42l51->reset_gpio) { + dev_dbg(dev, "Release reset gpio\n"); + gpiod_set_value_cansleep(cs42l51->reset_gpio, 0); + mdelay(2); + } + /* Verify that we have a CS42L51 */ ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { @@ -629,6 +642,8 @@ int cs42l51_remove(struct device *dev) { struct cs42l51_private *cs42l51 = dev_get_drvdata(dev); + gpiod_set_value_cansleep(cs42l51->reset_gpio, 1); + return regulator_bulk_disable(ARRAY_SIZE(cs42l51->supplies), cs42l51->supplies); } -- cgit v1.2.3-59-g8ed1b From 2f7c4ce09a43457666592f36e18a5db58cca86a0 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 3 Apr 2019 15:23:34 +0200 Subject: ASoC: cs42l51: add support of master mode Add support of master mode for cs42l51 cirrus audio codec. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 30 +++++++++++++++++++++++++++--- 1 file changed, 27 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 9b3ffa16b204..397b68901d1c 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -340,6 +340,19 @@ static struct cs42l51_ratios slave_auto_ratios[] = { { 256, CS42L51_DSM_MODE, 1 }, { 384, CS42L51_DSM_MODE, 1 }, }; +/* + * Master mode mclk/fs ratios. + * Recommended configurations are SSM for 4-50khz and DSM for 50-100kHz ranges + * The table below provides support of following ratios: + * 128: SSM (%128) with div2 disabled + * 256: SSM (%128) with div2 enabled + * In both cases, if sampling rate is above 50kHz, SSM is overridden + * with DSM (%128) configuration + */ +static struct cs42l51_ratios master_ratios[] = { + { 128, CS42L51_SSM_MODE, 0 }, { 256, CS42L51_SSM_MODE, 1 }, +}; + static int cs42l51_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -362,11 +375,13 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, unsigned int ratio; struct cs42l51_ratios *ratios = NULL; int nr_ratios = 0; - int intf_ctl, power_ctl, fmt; + int intf_ctl, power_ctl, fmt, mode; switch (cs42l51->func) { case MODE_MASTER: - return -EINVAL; + ratios = master_ratios; + nr_ratios = ARRAY_SIZE(master_ratios); + break; case MODE_SLAVE: ratios = slave_ratios; nr_ratios = ARRAY_SIZE(slave_ratios); @@ -402,7 +417,16 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, switch (cs42l51->func) { case MODE_MASTER: intf_ctl |= CS42L51_INTF_CTL_MASTER; - power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode); + mode = ratios[i].speed_mode; + /* Force DSM mode if sampling rate is above 50kHz */ + if (rate > 50000) + mode = CS42L51_DSM_MODE; + power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(mode); + /* + * Auto detect mode is not applicable for master mode and has to + * be disabled. Otherwise SPEED[1:0] bits will be ignored. + */ + power_ctl &= ~CS42L51_MIC_POWER_CTL_AUTO; break; case MODE_SLAVE: power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode); -- cgit v1.2.3-59-g8ed1b From 75a714823e89ceeb168092b4c4d0a3f52b415461 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 3 Apr 2019 15:23:36 +0200 Subject: ASoC: cs42l51: add power management Add sleep PM callbacks to support system low power modes. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51-i2c.c | 7 ++- sound/soc/codecs/cs42l51.c | 122 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l51.h | 2 + 3 files changed, 129 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 8333dbf18ea2..116221e581ce 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -29,8 +29,6 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c, struct regmap_config config; config = cs42l51_regmap; - config.val_bits = 8; - config.reg_bits = 8; return cs42l51_probe(&i2c->dev, devm_regmap_init_i2c(i2c, &config)); } @@ -40,10 +38,15 @@ static int cs42l51_i2c_remove(struct i2c_client *i2c) return cs42l51_remove(&i2c->dev); } +static const struct dev_pm_ops cs42l51_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(cs42l51_suspend, cs42l51_resume) +}; + static struct i2c_driver cs42l51_i2c_driver = { .driver = { .name = "cs42l51", .of_match_table = cs42l51_of_match, + .pm = &cs42l51_pm_ops, }, .probe = cs42l51_i2c_probe, .remove = cs42l51_i2c_remove, diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 397b68901d1c..b8f43858876a 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -56,6 +56,7 @@ struct cs42l51_private { enum master_slave_mode func; struct regulator_bulk_data supplies[ARRAY_SIZE(cs42l51_supply_names)]; struct gpio_desc *reset_gpio; + struct regmap *regmap; }; #define CS42L51_FORMATS ( \ @@ -575,7 +576,106 @@ static const struct snd_soc_component_driver soc_component_device_cs42l51 = { .non_legacy_dai_naming = 1, }; +static bool cs42l51_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L51_POWER_CTL1: + case CS42L51_MIC_POWER_CTL: + case CS42L51_INTF_CTL: + case CS42L51_MIC_CTL: + case CS42L51_ADC_CTL: + case CS42L51_ADC_INPUT: + case CS42L51_DAC_OUT_CTL: + case CS42L51_DAC_CTL: + case CS42L51_ALC_PGA_CTL: + case CS42L51_ALC_PGB_CTL: + case CS42L51_ADCA_ATT: + case CS42L51_ADCB_ATT: + case CS42L51_ADCA_VOL: + case CS42L51_ADCB_VOL: + case CS42L51_PCMA_VOL: + case CS42L51_PCMB_VOL: + case CS42L51_BEEP_FREQ: + case CS42L51_BEEP_VOL: + case CS42L51_BEEP_CONF: + case CS42L51_TONE_CTL: + case CS42L51_AOUTA_VOL: + case CS42L51_AOUTB_VOL: + case CS42L51_PCM_MIXER: + case CS42L51_LIMIT_THRES_DIS: + case CS42L51_LIMIT_REL: + case CS42L51_LIMIT_ATT: + case CS42L51_ALC_EN: + case CS42L51_ALC_REL: + case CS42L51_ALC_THRES: + case CS42L51_NOISE_CONF: + case CS42L51_CHARGE_FREQ: + return true; + default: + return false; + } +} + +static bool cs42l51_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L51_STATUS: + return true; + default: + return false; + } +} + +static bool cs42l51_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42L51_CHIP_REV_ID: + case CS42L51_POWER_CTL1: + case CS42L51_MIC_POWER_CTL: + case CS42L51_INTF_CTL: + case CS42L51_MIC_CTL: + case CS42L51_ADC_CTL: + case CS42L51_ADC_INPUT: + case CS42L51_DAC_OUT_CTL: + case CS42L51_DAC_CTL: + case CS42L51_ALC_PGA_CTL: + case CS42L51_ALC_PGB_CTL: + case CS42L51_ADCA_ATT: + case CS42L51_ADCB_ATT: + case CS42L51_ADCA_VOL: + case CS42L51_ADCB_VOL: + case CS42L51_PCMA_VOL: + case CS42L51_PCMB_VOL: + case CS42L51_BEEP_FREQ: + case CS42L51_BEEP_VOL: + case CS42L51_BEEP_CONF: + case CS42L51_TONE_CTL: + case CS42L51_AOUTA_VOL: + case CS42L51_AOUTB_VOL: + case CS42L51_PCM_MIXER: + case CS42L51_LIMIT_THRES_DIS: + case CS42L51_LIMIT_REL: + case CS42L51_LIMIT_ATT: + case CS42L51_ALC_EN: + case CS42L51_ALC_REL: + case CS42L51_ALC_THRES: + case CS42L51_NOISE_CONF: + case CS42L51_STATUS: + case CS42L51_CHARGE_FREQ: + return true; + default: + return false; + } +} + const struct regmap_config cs42l51_regmap = { + .reg_bits = 8, + .reg_stride = 1, + .val_bits = 8, + .use_single_write = true, + .readable_reg = cs42l51_readable_reg, + .volatile_reg = cs42l51_volatile_reg, + .writeable_reg = cs42l51_writeable_reg, .max_register = CS42L51_CHARGE_FREQ, .cache_type = REGCACHE_RBTREE, }; @@ -596,6 +696,7 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; dev_set_drvdata(dev, cs42l51); + cs42l51->regmap = regmap; cs42l51->mclk_handle = devm_clk_get(dev, "MCLK"); if (IS_ERR(cs42l51->mclk_handle)) { @@ -673,6 +774,27 @@ int cs42l51_remove(struct device *dev) } EXPORT_SYMBOL_GPL(cs42l51_remove); +int __maybe_unused cs42l51_suspend(struct device *dev) +{ + struct cs42l51_private *cs42l51 = dev_get_drvdata(dev); + + regcache_cache_only(cs42l51->regmap, true); + regcache_mark_dirty(cs42l51->regmap); + + return 0; +} +EXPORT_SYMBOL_GPL(cs42l51_suspend); + +int __maybe_unused cs42l51_resume(struct device *dev) +{ + struct cs42l51_private *cs42l51 = dev_get_drvdata(dev); + + regcache_cache_only(cs42l51->regmap, false); + + return regcache_sync(cs42l51->regmap); +} +EXPORT_SYMBOL_GPL(cs42l51_resume); + const struct of_device_id cs42l51_of_match[] = { { .compatible = "cirrus,cs42l51", }, { } diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index aef0ede82c7b..79dee01137c8 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -23,6 +23,8 @@ struct device; extern const struct regmap_config cs42l51_regmap; int cs42l51_probe(struct device *dev, struct regmap *regmap); int cs42l51_remove(struct device *dev); +int __maybe_unused cs42l51_suspend(struct device *dev); +int __maybe_unused cs42l51_resume(struct device *dev); extern const struct of_device_id cs42l51_of_match[]; #define CS42L51_CHIP_ID 0x1B -- cgit v1.2.3-59-g8ed1b From e04232c38190ec4485c62ceff94cf2b1c0caba20 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 3 Apr 2019 15:23:37 +0200 Subject: ASoC: cs42l51: add adc volume control Add ADC boost volume control for CS42L51 codec. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index b8f43858876a..58ece037d944 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -123,6 +123,7 @@ static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(aout_tlv, -10200, 50, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0); +static const DECLARE_TLV_DB_SCALE(adc_boost_tlv, 2000, 2000, 0); static const char *chan_mix[] = { "L R", "L+R", @@ -151,6 +152,8 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_SINGLE("Zero Cross Switch", CS42L51_DAC_CTL, 0, 0, 0), SOC_DOUBLE_TLV("Mic Boost Volume", CS42L51_MIC_CTL, 0, 1, 1, 0, boost_tlv), + SOC_DOUBLE_TLV("ADC Boost Volume", + CS42L51_MIC_CTL, 5, 6, 1, 0, adc_boost_tlv), SOC_SINGLE_TLV("Bass Volume", CS42L51_TONE_CTL, 0, 0xf, 1, tone_tlv), SOC_SINGLE_TLV("Treble Volume", CS42L51_TONE_CTL, 4, 0xf, 1, tone_tlv), SOC_ENUM_EXT("PCM channel mixer", -- cgit v1.2.3-59-g8ed1b From f96fb7d198ca624fe33c4145a004eb5a3d0eddec Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Thu, 4 Apr 2019 17:30:40 -0700 Subject: ASoC: core: remove link components before cleaning up card resources When the card is registered by the machine driver, dai link components are probed after the snd_card is created. This is done in snd_soc_bind_card() which calls snd_soc_instantiate_card() to first create the snd_card and then probes the link components by calling soc_probe_link_components(). The snd_card is used by the component driver to add the kcontrols associated with dapm widgets to the card. When the machine driver is unregistered, the snd_card is freed when the card resources are cleaned up. But the snd_card needs to be valid while unloading the topology dapm widgets in order to remove the kcontrols from the card. Since, unloading topology is done when the component driver is removed, the link components should be removed in snd_soc_unbind_card(). This will ensure that the kcontrols are removed before the card resources are cleaned up and the snd_card itself is freed. Signed-off-by: Ranjani Sridharan Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6f4842977b8d..75f6a8085a76 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2831,10 +2831,21 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) { + struct snd_soc_pcm_runtime *rtd; + int order; + if (card->instantiated) { card->instantiated = false; snd_soc_dapm_shutdown(card); snd_soc_flush_all_delayed_work(card); + + /* remove all components used by DAI links on this card */ + for_each_comp_order(order) { + for_each_card_rtds(card, rtd) { + soc_remove_link_components(card, rtd, order); + } + } + soc_cleanup_card_resources(card); if (!unregister) list_add(&card->list, &unbind_card_list); -- cgit v1.2.3-59-g8ed1b From 7c02509a8a9981fb2c16b75904423e7ab2f9f43a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 4 Apr 2019 13:17:29 +0200 Subject: ASoC: meson: axg-fifo: add g12a support The g12a fifos gained the ability to set the initial address of the pointer within the buffer, instead of defaulting to the buffer start address. It is not very useful to us (yet) but we need to put a copy the buffer start address in the related register for the fifo to work properly on the g12a SoC family Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-fifo.c | 34 +++++++++++++++++++++++++++++++--- sound/soc/meson/axg-fifo.h | 2 ++ 2 files changed, 33 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 75e5e480fda2..01c1c7db2510 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -19,7 +19,7 @@ * This file implements the platform operations common to the playback and * capture frontend DAI. The logic behind this two types of fifo is very * similar but some difference exist. - * These differences the respective DAI drivers + * These differences are handled in the respective DAI drivers */ static struct snd_pcm_hardware axg_fifo_hw = { @@ -133,6 +133,23 @@ static int axg_fifo_pcm_hw_params(struct snd_pcm_substream *ss, return 0; } +static int g12a_fifo_pcm_hw_params(struct snd_pcm_substream *ss, + struct snd_pcm_hw_params *params) +{ + struct axg_fifo *fifo = axg_fifo_data(ss); + struct snd_pcm_runtime *runtime = ss->runtime; + int ret; + + ret = axg_fifo_pcm_hw_params(ss, params); + if (ret) + return ret; + + /* Set the initial memory address of the DMA */ + regmap_write(fifo->map, FIFO_INIT_ADDR, runtime->dma_addr); + + return 0; +} + static int axg_fifo_pcm_hw_free(struct snd_pcm_substream *ss) { struct axg_fifo *fifo = axg_fifo_data(ss); @@ -262,6 +279,17 @@ const struct snd_pcm_ops axg_fifo_pcm_ops = { }; EXPORT_SYMBOL_GPL(axg_fifo_pcm_ops); +const struct snd_pcm_ops g12a_fifo_pcm_ops = { + .open = axg_fifo_pcm_open, + .close = axg_fifo_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = g12a_fifo_pcm_hw_params, + .hw_free = axg_fifo_pcm_hw_free, + .pointer = axg_fifo_pcm_pointer, + .trigger = axg_fifo_pcm_trigger, +}; +EXPORT_SYMBOL_GPL(g12a_fifo_pcm_ops); + int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type) { struct snd_card *card = rtd->card->snd_card; @@ -278,7 +306,7 @@ static const struct regmap_config axg_fifo_regmap_cfg = { .reg_bits = 32, .val_bits = 32, .reg_stride = 4, - .max_register = FIFO_STATUS2, + .max_register = FIFO_INIT_ADDR, }; int axg_fifo_probe(struct platform_device *pdev) @@ -339,6 +367,6 @@ int axg_fifo_probe(struct platform_device *pdev) } EXPORT_SYMBOL_GPL(axg_fifo_probe); -MODULE_DESCRIPTION("Amlogic AXG fifo driver"); +MODULE_DESCRIPTION("Amlogic AXG/G12A fifo driver"); MODULE_AUTHOR("Jerome Brunet "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h index d9f516cfbeda..5caf81241dfe 100644 --- a/sound/soc/meson/axg-fifo.h +++ b/sound/soc/meson/axg-fifo.h @@ -60,6 +60,7 @@ struct snd_soc_pcm_runtime; #define FIFO_STATUS1 0x14 #define STATUS1_INT_STS(x) ((x) << 0) #define FIFO_STATUS2 0x18 +#define FIFO_INIT_ADDR 0x24 struct axg_fifo { struct regmap *map; @@ -74,6 +75,7 @@ struct axg_fifo_match_data { }; extern const struct snd_pcm_ops axg_fifo_pcm_ops; +extern const struct snd_pcm_ops g12a_fifo_pcm_ops; int axg_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, unsigned int type); int axg_fifo_probe(struct platform_device *pdev); -- cgit v1.2.3-59-g8ed1b From a3c23a8ad4dc07100d916d75ca30c982288b868d Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 4 Apr 2019 13:17:30 +0200 Subject: ASoC: meson: axg-toddr: add g12a support Since the g12a SoC fifo can set the fifo initial start address, we must make sure to actually reset the write pointer to this address when starting a capture. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-toddr.c | 53 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) (limited to 'sound') diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index 0e9ca3882ae5..4f63e434fad4 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -24,6 +24,7 @@ #define CTRL0_TODDR_MSB_POS(x) ((x) << 8) #define CTRL0_TODDR_LSB_POS_MASK GENMASK(7, 3) #define CTRL0_TODDR_LSB_POS(x) ((x) << 3) +#define CTRL1_TODDR_FORCE_FINISH BIT(25) #define TODDR_MSB_POS 31 @@ -33,6 +34,22 @@ static int axg_toddr_pcm_new(struct snd_soc_pcm_runtime *rtd, return axg_fifo_pcm_new(rtd, SNDRV_PCM_STREAM_CAPTURE); } +static int g12a_toddr_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + + /* Reset the write pointer to the FIFO_INIT_ADDR */ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_TODDR_FORCE_FINISH, 0); + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_TODDR_FORCE_FINISH, CTRL1_TODDR_FORCE_FINISH); + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_TODDR_FORCE_FINISH, 0); + + return 0; +} + static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -172,10 +189,46 @@ static const struct axg_fifo_match_data axg_toddr_match_data = { .dai_drv = &axg_toddr_dai_drv }; +static const struct snd_soc_dai_ops g12a_toddr_ops = { + .prepare = g12a_toddr_dai_prepare, + .hw_params = axg_toddr_dai_hw_params, + .startup = axg_toddr_dai_startup, + .shutdown = axg_toddr_dai_shutdown, +}; + +static struct snd_soc_dai_driver g12a_toddr_dai_drv = { + .name = "TODDR", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .rates = AXG_FIFO_RATES, + .formats = AXG_FIFO_FORMATS, + }, + .ops = &g12a_toddr_ops, + .pcm_new = axg_toddr_pcm_new, +}; + +static const struct snd_soc_component_driver g12a_toddr_component_drv = { + .dapm_widgets = axg_toddr_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_toddr_dapm_widgets), + .dapm_routes = axg_toddr_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_toddr_dapm_routes), + .ops = &g12a_fifo_pcm_ops +}; + +static const struct axg_fifo_match_data g12a_toddr_match_data = { + .component_drv = &g12a_toddr_component_drv, + .dai_drv = &g12a_toddr_dai_drv +}; + static const struct of_device_id axg_toddr_of_match[] = { { .compatible = "amlogic,axg-toddr", .data = &axg_toddr_match_data, + }, { + .compatible = "amlogic,g12a-toddr", + .data = &g12a_toddr_match_data, }, {} }; MODULE_DEVICE_TABLE(of, axg_toddr_of_match); -- cgit v1.2.3-59-g8ed1b From fcced66f208d778aa2dea05910161689503c16bf Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 4 Apr 2019 13:17:31 +0200 Subject: ASoC: meson: axg-frddr: add g12a support On the axg, frddr could only be connected to 1 downstream element, so the playback was possible on 1 interface only at a time. On the g12a, the frddr may connect and wait for the request of up to 3 downstream elements. With this, it possible for single playback to be played on several interfaces at the same time. Like the toddr fifo, the g12a frddr also need to take care of resetting the read pointer to the initial fifo address when preparing a playback. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-frddr.c | 143 +++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 140 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index a6f6f6a2eca8..2b8807737b2b 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -3,7 +3,9 @@ // Copyright (c) 2018 BayLibre, SAS. // Author: Jerome Brunet -/* This driver implements the frontend playback DAI of AXG based SoCs */ +/* + * This driver implements the frontend playback DAI of AXG and G12A based SoCs + */ #include #include @@ -14,7 +16,29 @@ #include "axg-fifo.h" -#define CTRL0_FRDDR_PP_MODE BIT(30) +#define CTRL0_FRDDR_PP_MODE BIT(30) +#define CTRL0_SEL1_EN_SHIFT 3 +#define CTRL0_SEL2_SHIFT 4 +#define CTRL0_SEL2_EN_SHIFT 7 +#define CTRL0_SEL3_SHIFT 8 +#define CTRL0_SEL3_EN_SHIFT 11 +#define CTRL1_FRDDR_FORCE_FINISH BIT(12) + +static int g12a_frddr_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + + /* Reset the read pointer to the FIFO_INIT_ADDR */ + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_FRDDR_FORCE_FINISH, 0); + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_FRDDR_FORCE_FINISH, CTRL1_FRDDR_FORCE_FINISH); + regmap_update_bits(fifo->map, FIFO_CTRL1, + CTRL1_FRDDR_FORCE_FINISH, 0); + + return 0; +} static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -119,10 +143,123 @@ static const struct axg_fifo_match_data axg_frddr_match_data = { .dai_drv = &axg_frddr_dai_drv }; +static const struct snd_soc_dai_ops g12a_frddr_ops = { + .prepare = g12a_frddr_dai_prepare, + .startup = axg_frddr_dai_startup, + .shutdown = axg_frddr_dai_shutdown, +}; + +static struct snd_soc_dai_driver g12a_frddr_dai_drv = { + .name = "FRDDR", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = AXG_FIFO_CH_MAX, + .rates = AXG_FIFO_RATES, + .formats = AXG_FIFO_FORMATS, + }, + .ops = &g12a_frddr_ops, + .pcm_new = axg_frddr_pcm_new, +}; + +static const char * const g12a_frddr_sel_texts[] = { + "OUT 0", "OUT 1", "OUT 2", "OUT 3", "OUT 4", +}; + +static SOC_ENUM_SINGLE_DECL(g12a_frddr_sel1_enum, FIFO_CTRL0, CTRL0_SEL_SHIFT, + g12a_frddr_sel_texts); +static SOC_ENUM_SINGLE_DECL(g12a_frddr_sel2_enum, FIFO_CTRL0, CTRL0_SEL2_SHIFT, + g12a_frddr_sel_texts); +static SOC_ENUM_SINGLE_DECL(g12a_frddr_sel3_enum, FIFO_CTRL0, CTRL0_SEL3_SHIFT, + g12a_frddr_sel_texts); + +static const struct snd_kcontrol_new g12a_frddr_out1_demux = + SOC_DAPM_ENUM("Output Src 1", g12a_frddr_sel1_enum); +static const struct snd_kcontrol_new g12a_frddr_out2_demux = + SOC_DAPM_ENUM("Output Src 2", g12a_frddr_sel2_enum); +static const struct snd_kcontrol_new g12a_frddr_out3_demux = + SOC_DAPM_ENUM("Output Src 3", g12a_frddr_sel3_enum); + +static const struct snd_kcontrol_new g12a_frddr_out1_enable = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", FIFO_CTRL0, + CTRL0_SEL1_EN_SHIFT, 1, 0); +static const struct snd_kcontrol_new g12a_frddr_out2_enable = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", FIFO_CTRL0, + CTRL0_SEL2_EN_SHIFT, 1, 0); +static const struct snd_kcontrol_new g12a_frddr_out3_enable = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", FIFO_CTRL0, + CTRL0_SEL3_EN_SHIFT, 1, 0); + +static const struct snd_soc_dapm_widget g12a_frddr_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("SRC 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SRC 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SRC 3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SWITCH("SRC 1 EN", SND_SOC_NOPM, 0, 0, + &g12a_frddr_out1_enable), + SND_SOC_DAPM_SWITCH("SRC 2 EN", SND_SOC_NOPM, 0, 0, + &g12a_frddr_out2_enable), + SND_SOC_DAPM_SWITCH("SRC 3 EN", SND_SOC_NOPM, 0, 0, + &g12a_frddr_out3_enable), + SND_SOC_DAPM_DEMUX("SINK 1 SEL", SND_SOC_NOPM, 0, 0, + &g12a_frddr_out1_demux), + SND_SOC_DAPM_DEMUX("SINK 2 SEL", SND_SOC_NOPM, 0, 0, + &g12a_frddr_out2_demux), + SND_SOC_DAPM_DEMUX("SINK 3 SEL", SND_SOC_NOPM, 0, 0, + &g12a_frddr_out3_demux), + SND_SOC_DAPM_AIF_OUT("OUT 0", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("OUT 4", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route g12a_frddr_dapm_routes[] = { + { "SRC 1", NULL, "Playback" }, + { "SRC 2", NULL, "Playback" }, + { "SRC 3", NULL, "Playback" }, + { "SRC 1 EN", "Switch", "SRC 1" }, + { "SRC 2 EN", "Switch", "SRC 2" }, + { "SRC 3 EN", "Switch", "SRC 3" }, + { "SINK 1 SEL", NULL, "SRC 1 EN" }, + { "SINK 2 SEL", NULL, "SRC 2 EN" }, + { "SINK 3 SEL", NULL, "SRC 3 EN" }, + { "OUT 0", "OUT 0", "SINK 1 SEL" }, + { "OUT 1", "OUT 1", "SINK 1 SEL" }, + { "OUT 2", "OUT 2", "SINK 1 SEL" }, + { "OUT 3", "OUT 3", "SINK 1 SEL" }, + { "OUT 4", "OUT 4", "SINK 1 SEL" }, + { "OUT 0", "OUT 0", "SINK 2 SEL" }, + { "OUT 1", "OUT 1", "SINK 2 SEL" }, + { "OUT 2", "OUT 2", "SINK 2 SEL" }, + { "OUT 3", "OUT 3", "SINK 2 SEL" }, + { "OUT 4", "OUT 4", "SINK 2 SEL" }, + { "OUT 0", "OUT 0", "SINK 3 SEL" }, + { "OUT 1", "OUT 1", "SINK 3 SEL" }, + { "OUT 2", "OUT 2", "SINK 3 SEL" }, + { "OUT 3", "OUT 3", "SINK 3 SEL" }, + { "OUT 4", "OUT 4", "SINK 3 SEL" }, +}; + +static const struct snd_soc_component_driver g12a_frddr_component_drv = { + .dapm_widgets = g12a_frddr_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(g12a_frddr_dapm_widgets), + .dapm_routes = g12a_frddr_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(g12a_frddr_dapm_routes), + .ops = &g12a_fifo_pcm_ops +}; + +static const struct axg_fifo_match_data g12a_frddr_match_data = { + .component_drv = &g12a_frddr_component_drv, + .dai_drv = &g12a_frddr_dai_drv +}; + static const struct of_device_id axg_frddr_of_match[] = { { .compatible = "amlogic,axg-frddr", .data = &axg_frddr_match_data, + }, { + .compatible = "amlogic,g12a-frddr", + .data = &g12a_frddr_match_data, }, {} }; MODULE_DEVICE_TABLE(of, axg_frddr_of_match); @@ -136,6 +273,6 @@ static struct platform_driver axg_frddr_pdrv = { }; module_platform_driver(axg_frddr_pdrv); -MODULE_DESCRIPTION("Amlogic AXG playback fifo driver"); +MODULE_DESCRIPTION("Amlogic AXG/G12A playback fifo driver"); MODULE_AUTHOR("Jerome Brunet "); MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-59-g8ed1b From f01bc67f58fde599b48d2dde5d0f48dccd84c4f1 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 4 Apr 2019 13:17:32 +0200 Subject: ASoC: meson: axg-tdm-formatter: rework quirks settings The g12a tdmout requires a different signal skew offset than the axg. With this change, the skew offset is added as a parameter of the tdm formatters to prepare the addition of the g12a support. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdm-formatter.c | 6 ++++-- sound/soc/meson/axg-tdm-formatter.h | 11 +++++++++-- sound/soc/meson/axg-tdmin.c | 16 +++++++++++----- sound/soc/meson/axg-tdmout.c | 16 +++++++++++----- 4 files changed, 35 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index 43e390f9358a..0c6cce5c5773 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -68,7 +68,7 @@ EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks); static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter) { struct axg_tdm_stream *ts = formatter->stream; - bool invert = formatter->drv->invert_sclk; + bool invert = formatter->drv->quirks->invert_sclk; int ret; /* Do nothing if the formatter is already enabled */ @@ -85,7 +85,9 @@ static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter) return ret; /* Setup the stream parameter in the formatter */ - ret = formatter->drv->ops->prepare(formatter->map, formatter->stream); + ret = formatter->drv->ops->prepare(formatter->map, + formatter->drv->quirks, + formatter->stream); if (ret) return ret; diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h index cf947caf3cb1..9ef98e955cb2 100644 --- a/sound/soc/meson/axg-tdm-formatter.h +++ b/sound/soc/meson/axg-tdm-formatter.h @@ -14,18 +14,25 @@ struct regmap; struct snd_soc_dapm_widget; struct snd_kcontrol; +struct axg_tdm_formatter_hw { + unsigned int skew_offset; + bool invert_sclk; +}; + struct axg_tdm_formatter_ops { struct axg_tdm_stream *(*get_stream)(struct snd_soc_dapm_widget *w); void (*enable)(struct regmap *map); void (*disable)(struct regmap *map); - int (*prepare)(struct regmap *map, struct axg_tdm_stream *ts); + int (*prepare)(struct regmap *map, + const struct axg_tdm_formatter_hw *quirks, + struct axg_tdm_stream *ts); }; struct axg_tdm_formatter_driver { const struct snd_soc_component_driver *component_drv; const struct regmap_config *regmap_cfg; const struct axg_tdm_formatter_ops *ops; - bool invert_sclk; + const struct axg_tdm_formatter_hw *quirks; }; int axg_tdm_formatter_set_channel_masks(struct regmap *map, diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index bbac44c81688..a790f925a4ef 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -107,21 +107,22 @@ static void axg_tdmin_disable(struct regmap *map) regmap_update_bits(map, TDMIN_CTRL, TDMIN_CTRL_ENABLE, 0); } -static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts) +static int axg_tdmin_prepare(struct regmap *map, + const struct axg_tdm_formatter_hw *quirks, + struct axg_tdm_stream *ts) { - unsigned int val = 0; + unsigned int val, skew = quirks->skew_offset; /* Set stream skew */ switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_DSP_A: - val |= TDMIN_CTRL_IN_BIT_SKEW(3); + skew += 1; break; case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_DSP_B: - val = TDMIN_CTRL_IN_BIT_SKEW(2); break; default: @@ -130,6 +131,8 @@ static int axg_tdmin_prepare(struct regmap *map, struct axg_tdm_stream *ts) return -EINVAL; } + val = TDMIN_CTRL_IN_BIT_SKEW(skew); + /* Set stream format mode */ switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -204,7 +207,10 @@ static const struct axg_tdm_formatter_driver axg_tdmin_drv = { .component_drv = &axg_tdmin_component_drv, .regmap_cfg = &axg_tdmin_regmap_cfg, .ops = &axg_tdmin_ops, - .invert_sclk = false, + .quirks = &(const struct axg_tdm_formatter_hw) { + .invert_sclk = false, + .skew_offset = 2, + }, }; static const struct of_device_id axg_tdmin_of_match[] = { diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c index f73368ee1088..3984818e2a7c 100644 --- a/sound/soc/meson/axg-tdmout.c +++ b/sound/soc/meson/axg-tdmout.c @@ -124,21 +124,22 @@ static void axg_tdmout_disable(struct regmap *map) regmap_update_bits(map, TDMOUT_CTRL0, TDMOUT_CTRL0_ENABLE, 0); } -static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts) +static int axg_tdmout_prepare(struct regmap *map, + const struct axg_tdm_formatter_hw *quirks, + struct axg_tdm_stream *ts) { - unsigned int val = 0; + unsigned int val, skew = quirks->skew_offset; /* Set the stream skew */ switch (ts->iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_DSP_A: - val |= TDMOUT_CTRL0_INIT_BITNUM(1); break; case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_DSP_B: - val |= TDMOUT_CTRL0_INIT_BITNUM(2); + skew += 1; break; default: @@ -147,6 +148,8 @@ static int axg_tdmout_prepare(struct regmap *map, struct axg_tdm_stream *ts) return -EINVAL; } + val = TDMOUT_CTRL0_INIT_BITNUM(skew); + /* Set the slot width */ val |= TDMOUT_CTRL0_BITNUM(ts->iface->slot_width - 1); @@ -234,7 +237,10 @@ static const struct axg_tdm_formatter_driver axg_tdmout_drv = { .component_drv = &axg_tdmout_component_drv, .regmap_cfg = &axg_tdmout_regmap_cfg, .ops = &axg_tdmout_ops, - .invert_sclk = true, + .quirks = &(const struct axg_tdm_formatter_hw) { + .invert_sclk = true, + .skew_offset = 1, + }, }; static const struct of_device_id axg_tdmout_of_match[] = { -- cgit v1.2.3-59-g8ed1b From aa191a37b801be6c5abebe77e67dcec7c5c0faee Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 4 Apr 2019 13:17:33 +0200 Subject: ASoC: meson: axg-tdmout: add g12a support The axg tdmout driver just need a different skew offset to operate correctly on the g12a SoC family. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdmout.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c index 3984818e2a7c..527bfc4487e0 100644 --- a/sound/soc/meson/axg-tdmout.c +++ b/sound/soc/meson/axg-tdmout.c @@ -243,10 +243,23 @@ static const struct axg_tdm_formatter_driver axg_tdmout_drv = { }, }; +static const struct axg_tdm_formatter_driver g12a_tdmout_drv = { + .component_drv = &axg_tdmout_component_drv, + .regmap_cfg = &axg_tdmout_regmap_cfg, + .ops = &axg_tdmout_ops, + .quirks = &(const struct axg_tdm_formatter_hw) { + .invert_sclk = true, + .skew_offset = 2, + }, +}; + static const struct of_device_id axg_tdmout_of_match[] = { { .compatible = "amlogic,axg-tdmout", .data = &axg_tdmout_drv, + }, { + .compatible = "amlogic,g12a-tdmout", + .data = &g12a_tdmout_drv, }, {} }; MODULE_DEVICE_TABLE(of, axg_tdmout_of_match); -- cgit v1.2.3-59-g8ed1b From 5aebe7c7f9c20ef225c0c3a25c06a20c3938e390 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 4 Apr 2019 14:13:57 -0500 Subject: ASoC: topology: fix endianness issues Use le16/32/64_to_cpu() as needed to make Sparse happy. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 291 +++++++++++++++++++++++++++-------------------- 1 file changed, 170 insertions(+), 121 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 472f7705da93..03c4dbdfc584 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -197,8 +197,8 @@ static int tplc_chan_get_reg(struct soc_tplg *tplg, int i; for (i = 0; i < SND_SOC_TPLG_MAX_CHAN; i++) { - if (chan[i].id == map) - return chan[i].reg; + if (le32_to_cpu(chan[i].id) == map) + return le32_to_cpu(chan[i].reg); } return -EINVAL; @@ -210,8 +210,8 @@ static int tplc_chan_get_shift(struct soc_tplg *tplg, int i; for (i = 0; i < SND_SOC_TPLG_MAX_CHAN; i++) { - if (chan[i].id == map) - return chan[i].shift; + if (le32_to_cpu(chan[i].id) == map) + return le32_to_cpu(chan[i].shift); } return -EINVAL; @@ -592,7 +592,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, const struct snd_soc_tplg_bytes_ext_ops *ext_ops; int num_ops, i; - if (hdr->ops.info == SND_SOC_TPLG_CTL_BYTES + if (le32_to_cpu(hdr->ops.info) == SND_SOC_TPLG_CTL_BYTES && k->iface & SNDRV_CTL_ELEM_IFACE_MIXER && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { @@ -708,9 +708,9 @@ static int soc_tplg_create_tlv_db_scale(struct soc_tplg *tplg, p[0] = SNDRV_CTL_TLVT_DB_SCALE; p[1] = item_len; - p[2] = scale->min; - p[3] = (scale->step & TLV_DB_SCALE_MASK) - | (scale->mute ? TLV_DB_SCALE_MUTE : 0); + p[2] = le32_to_cpu(scale->min); + p[3] = (le32_to_cpu(scale->step) & TLV_DB_SCALE_MASK) + | (le32_to_cpu(scale->mute) ? TLV_DB_SCALE_MUTE : 0); kc->tlv.p = (void *)p; return 0; @@ -720,13 +720,14 @@ static int soc_tplg_create_tlv(struct soc_tplg *tplg, struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_hdr *tc) { struct snd_soc_tplg_ctl_tlv *tplg_tlv; + u32 access = le32_to_cpu(tc->access); - if (!(tc->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE)) + if (!(access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE)) return 0; - if (!(tc->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)) { + if (!(access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)) { tplg_tlv = &tc->tlv; - switch (tplg_tlv->type) { + switch (le32_to_cpu(tplg_tlv->type)) { case SNDRV_CTL_TLVT_DB_SCALE: return soc_tplg_create_tlv_db_scale(tplg, kc, &tplg_tlv->scale); @@ -777,7 +778,7 @@ static int soc_tplg_dbytes_create(struct soc_tplg *tplg, unsigned int count, return -ENOMEM; tplg->pos += (sizeof(struct snd_soc_tplg_bytes_control) + - be->priv.size); + le32_to_cpu(be->priv.size)); dev_dbg(tplg->dev, "ASoC: adding bytes kcontrol %s with access 0x%x\n", @@ -787,9 +788,9 @@ static int soc_tplg_dbytes_create(struct soc_tplg *tplg, unsigned int count, kc.name = be->hdr.name; kc.private_value = (long)sbe; kc.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc.access = be->hdr.access; + kc.access = le32_to_cpu(be->hdr.access); - sbe->max = be->max; + sbe->max = le32_to_cpu(be->max); sbe->dobj.type = SND_SOC_DOBJ_BYTES; sbe->dobj.ops = tplg->ops; INIT_LIST_HEAD(&sbe->dobj.list); @@ -857,7 +858,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, if (sm == NULL) return -ENOMEM; tplg->pos += (sizeof(struct snd_soc_tplg_mixer_control) + - mc->priv.size); + le32_to_cpu(mc->priv.size)); dev_dbg(tplg->dev, "ASoC: adding mixer kcontrol %s with access 0x%x\n", @@ -867,7 +868,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, kc.name = mc->hdr.name; kc.private_value = (long)sm; kc.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc.access = mc->hdr.access; + kc.access = le32_to_cpu(mc->hdr.access); /* we only support FL/FR channel mapping atm */ sm->reg = tplc_chan_get_reg(tplg, mc->channel, @@ -879,10 +880,10 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, sm->rshift = tplc_chan_get_shift(tplg, mc->channel, SNDRV_CHMAP_FR); - sm->max = mc->max; - sm->min = mc->min; - sm->invert = mc->invert; - sm->platform_max = mc->platform_max; + sm->max = le32_to_cpu(mc->max); + sm->min = le32_to_cpu(mc->min); + sm->invert = le32_to_cpu(mc->invert); + sm->platform_max = le32_to_cpu(mc->platform_max); sm->dobj.index = tplg->index; sm->dobj.ops = tplg->ops; sm->dobj.type = SND_SOC_DOBJ_MIXER; @@ -933,7 +934,7 @@ static int soc_tplg_denum_create_texts(struct soc_enum *se, int i, ret; se->dobj.control.dtexts = - kcalloc(ec->items, sizeof(char *), GFP_KERNEL); + kcalloc(le32_to_cpu(ec->items), sizeof(char *), GFP_KERNEL); if (se->dobj.control.dtexts == NULL) return -ENOMEM; @@ -965,15 +966,22 @@ err: static int soc_tplg_denum_create_values(struct soc_enum *se, struct snd_soc_tplg_enum_control *ec) { - if (ec->items > sizeof(*ec->values)) + int i; + + if (le32_to_cpu(ec->items) > sizeof(*ec->values)) return -EINVAL; - se->dobj.control.dvalues = kmemdup(ec->values, - ec->items * sizeof(u32), + se->dobj.control.dvalues = kzalloc(le32_to_cpu(ec->items) * + sizeof(u32), GFP_KERNEL); if (!se->dobj.control.dvalues) return -ENOMEM; + /* convert from little-endian */ + for (i = 0; i < le32_to_cpu(ec->items); i++) { + se->dobj.control.dvalues[i] = le32_to_cpu(ec->values[i]); + } + return 0; } @@ -1007,7 +1015,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count, return -ENOMEM; tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - ec->priv.size); + le32_to_cpu(ec->priv.size)); dev_dbg(tplg->dev, "ASoC: adding enum kcontrol %s size %d\n", ec->hdr.name, ec->items); @@ -1016,7 +1024,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count, kc.name = ec->hdr.name; kc.private_value = (long)se; kc.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc.access = ec->hdr.access; + kc.access = le32_to_cpu(ec->hdr.access); se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); se->shift_l = tplc_chan_get_shift(tplg, ec->channel, @@ -1024,14 +1032,14 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count, se->shift_r = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FL); - se->items = ec->items; - se->mask = ec->mask; + se->items = le32_to_cpu(ec->items); + se->mask = le32_to_cpu(ec->mask); se->dobj.index = tplg->index; se->dobj.type = SND_SOC_DOBJ_ENUM; se->dobj.ops = tplg->ops; INIT_LIST_HEAD(&se->dobj.list); - switch (ec->hdr.ops.info) { + switch (le32_to_cpu(ec->hdr.ops.info)) { case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: case SND_SOC_TPLG_CTL_ENUM_VALUE: err = soc_tplg_denum_create_values(se, ec); @@ -1104,23 +1112,24 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, int i; if (tplg->pass != SOC_TPLG_PASS_MIXER) { - tplg->pos += hdr->size + hdr->payload_size; + tplg->pos += le32_to_cpu(hdr->size) + + le32_to_cpu(hdr->payload_size); return 0; } dev_dbg(tplg->dev, "ASoC: adding %d kcontrols at 0x%lx\n", hdr->count, soc_tplg_get_offset(tplg)); - for (i = 0; i < hdr->count; i++) { + for (i = 0; i < le32_to_cpu(hdr->count); i++) { control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; - if (control_hdr->size != sizeof(*control_hdr)) { + if (le32_to_cpu(control_hdr->size) != sizeof(*control_hdr)) { dev_err(tplg->dev, "ASoC: invalid control size\n"); return -EINVAL; } - switch (control_hdr->ops.info) { + switch (le32_to_cpu(control_hdr->ops.info)) { case SND_SOC_TPLG_CTL_VOLSW: case SND_SOC_TPLG_CTL_STROBE: case SND_SOC_TPLG_CTL_VOLSW_SX: @@ -1128,17 +1137,20 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, case SND_SOC_TPLG_CTL_RANGE: case SND_SOC_TPLG_DAPM_CTL_VOLSW: case SND_SOC_TPLG_DAPM_CTL_PIN: - soc_tplg_dmixer_create(tplg, 1, hdr->payload_size); + soc_tplg_dmixer_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; case SND_SOC_TPLG_CTL_ENUM: case SND_SOC_TPLG_CTL_ENUM_VALUE: case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - soc_tplg_denum_create(tplg, 1, hdr->payload_size); + soc_tplg_denum_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; case SND_SOC_TPLG_CTL_BYTES: - soc_tplg_dbytes_create(tplg, 1, hdr->payload_size); + soc_tplg_dbytes_create(tplg, 1, + le32_to_cpu(hdr->payload_size)); break; default: soc_bind_err(tplg, control_hdr, i); @@ -1166,17 +1178,22 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, struct snd_soc_dapm_context *dapm = &tplg->comp->dapm; struct snd_soc_tplg_dapm_graph_elem *elem; struct snd_soc_dapm_route **routes; - int count = hdr->count, i, j; + int count, i, j; int ret = 0; + count = le32_to_cpu(hdr->count); + if (tplg->pass != SOC_TPLG_PASS_GRAPH) { - tplg->pos += hdr->size + hdr->payload_size; + tplg->pos += + le32_to_cpu(hdr->size) + + le32_to_cpu(hdr->payload_size); + return 0; } if (soc_tplg_check_elem_count(tplg, sizeof(struct snd_soc_tplg_dapm_graph_elem), - count, hdr->payload_size, "graph")) { + count, le32_to_cpu(hdr->payload_size), "graph")) { dev_err(tplg->dev, "ASoC: invalid count %d for DAPM routes\n", count); @@ -1291,7 +1308,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( goto err_str; tplg->pos += (sizeof(struct snd_soc_tplg_mixer_control) + - mc->priv.size); + le32_to_cpu(mc->priv.size)); dev_dbg(tplg->dev, " adding DAPM widget mixer control %s at %d\n", mc->hdr.name, i); @@ -1404,7 +1421,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( se->mask = ec->mask; se->dobj.index = tplg->index; - switch (ec->hdr.ops.info) { + switch (le32_to_cpu(ec->hdr.ops.info)) { case SND_SOC_TPLG_CTL_ENUM_VALUE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: err = soc_tplg_denum_create_values(se, ec); @@ -1495,7 +1512,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( goto err; tplg->pos += (sizeof(struct snd_soc_tplg_bytes_control) + - be->priv.size); + le32_to_cpu(be->priv.size)); dev_dbg(tplg->dev, "ASoC: adding bytes kcontrol %s with access 0x%x\n", @@ -1567,7 +1584,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, memset(&template, 0, sizeof(template)); /* map user to kernel widget ID */ - template.id = get_widget_id(w->id); + template.id = get_widget_id(le32_to_cpu(w->id)); if (template.id < 0) return template.id; @@ -1580,18 +1597,20 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, ret = -ENOMEM; goto err; } - template.reg = w->reg; - template.shift = w->shift; - template.mask = w->mask; - template.subseq = w->subseq; + template.reg = le32_to_cpu(w->reg); + template.shift = le32_to_cpu(w->shift); + template.mask = le32_to_cpu(w->mask); + template.subseq = le32_to_cpu(w->subseq); template.on_val = w->invert ? 0 : 1; template.off_val = w->invert ? 1 : 0; - template.ignore_suspend = w->ignore_suspend; - template.event_flags = w->event_flags; + template.ignore_suspend = le32_to_cpu(w->ignore_suspend); + template.event_flags = le16_to_cpu(w->event_flags); template.dobj.index = tplg->index; tplg->pos += - (sizeof(struct snd_soc_tplg_dapm_widget) + w->priv.size); + (sizeof(struct snd_soc_tplg_dapm_widget) + + le32_to_cpu(w->priv.size)); + if (w->num_kcontrols == 0) { kcontrol_type = 0; template.num_kcontrols = 0; @@ -1602,7 +1621,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, dev_dbg(tplg->dev, "ASoC: template %s has %d controls of type %x\n", w->name, w->num_kcontrols, control_hdr->type); - switch (control_hdr->ops.info) { + switch (le32_to_cpu(control_hdr->ops.info)) { case SND_SOC_TPLG_CTL_VOLSW: case SND_SOC_TPLG_CTL_STROBE: case SND_SOC_TPLG_CTL_VOLSW_SX: @@ -1610,7 +1629,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_CTL_RANGE: case SND_SOC_TPLG_DAPM_CTL_VOLSW: kcontrol_type = SND_SOC_TPLG_TYPE_MIXER; /* volume mixer */ - template.num_kcontrols = w->num_kcontrols; + template.num_kcontrols = le32_to_cpu(w->num_kcontrols); template.kcontrol_news = soc_tplg_dapm_widget_dmixer_create(tplg, template.num_kcontrols); @@ -1625,7 +1644,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: kcontrol_type = SND_SOC_TPLG_TYPE_ENUM; /* enumerated mixer */ - template.num_kcontrols = w->num_kcontrols; + template.num_kcontrols = le32_to_cpu(w->num_kcontrols); template.kcontrol_news = soc_tplg_dapm_widget_denum_create(tplg, template.num_kcontrols); @@ -1636,7 +1655,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, break; case SND_SOC_TPLG_CTL_BYTES: kcontrol_type = SND_SOC_TPLG_TYPE_BYTES; /* bytes control */ - template.num_kcontrols = w->num_kcontrols; + template.num_kcontrols = le32_to_cpu(w->num_kcontrols); template.kcontrol_news = soc_tplg_dapm_widget_dbytes_create(tplg, template.num_kcontrols); @@ -1648,7 +1667,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, default: dev_err(tplg->dev, "ASoC: invalid widget control type %d:%d:%d\n", control_hdr->ops.get, control_hdr->ops.put, - control_hdr->ops.info); + le32_to_cpu(control_hdr->ops.info)); ret = -EINVAL; goto hdr_err; } @@ -1698,7 +1717,9 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { struct snd_soc_tplg_dapm_widget *widget; - int ret, count = hdr->count, i; + int ret, count, i; + + count = le32_to_cpu(hdr->count); if (tplg->pass != SOC_TPLG_PASS_WIDGET) return 0; @@ -1707,7 +1728,7 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; - if (widget->size != sizeof(*widget)) { + if (le32_to_cpu(widget->size) != sizeof(*widget)) { dev_err(tplg->dev, "ASoC: invalid widget size\n"); return -EINVAL; } @@ -1749,13 +1770,13 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream, struct snd_soc_tplg_stream_caps *caps) { stream->stream_name = kstrdup(caps->name, GFP_KERNEL); - stream->channels_min = caps->channels_min; - stream->channels_max = caps->channels_max; - stream->rates = caps->rates; - stream->rate_min = caps->rate_min; - stream->rate_max = caps->rate_max; - stream->formats = caps->formats; - stream->sig_bits = caps->sig_bits; + stream->channels_min = le32_to_cpu(caps->channels_min); + stream->channels_max = le32_to_cpu(caps->channels_max); + stream->rates = le32_to_cpu(caps->rates); + stream->rate_min = le32_to_cpu(caps->rate_min); + stream->rate_max = le32_to_cpu(caps->rate_max); + stream->formats = le64_to_cpu(caps->formats); + stream->sig_bits = le32_to_cpu(caps->sig_bits); } static void set_dai_flags(struct snd_soc_dai_driver *dai_drv, @@ -1790,7 +1811,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (strlen(pcm->dai_name)) dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); - dai_drv->id = pcm->dai_id; + dai_drv->id = le32_to_cpu(pcm->dai_id); if (pcm->playback) { stream = &dai_drv->playback; @@ -1865,7 +1886,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, link->name = kstrdup(pcm->pcm_name, GFP_KERNEL); link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL); } - link->id = pcm->pcm_id; + link->id = le32_to_cpu(pcm->pcm_id); if (strlen(pcm->dai_name)) link->cpu_dai_name = kstrdup(pcm->dai_name, GFP_KERNEL); @@ -1875,10 +1896,12 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, /* enable DPCM */ link->dynamic = 1; - link->dpcm_playback = pcm->playback; - link->dpcm_capture = pcm->capture; + link->dpcm_playback = le32_to_cpu(pcm->playback); + link->dpcm_capture = le32_to_cpu(pcm->capture); if (pcm->flag_mask) - set_link_flags(link, pcm->flag_mask, pcm->flags); + set_link_flags(link, + le32_to_cpu(pcm->flag_mask), + le32_to_cpu(pcm->flags)); /* pass control to component driver for optional further init */ ret = soc_tplg_dai_link_load(tplg, link, NULL); @@ -1917,7 +1940,7 @@ static int soc_tplg_pcm_create(struct soc_tplg *tplg, static void stream_caps_new_ver(struct snd_soc_tplg_stream_caps *dest, struct snd_soc_tplg_stream_caps_v4 *src) { - dest->size = sizeof(*dest); + dest->size = cpu_to_le32(sizeof(*dest)); memcpy(dest->name, src->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dest->formats = src->formats; dest->rates = src->rates; @@ -1951,7 +1974,7 @@ static int pcm_new_ver(struct soc_tplg *tplg, *pcm = NULL; - if (src->size != sizeof(*src_v4)) { + if (le32_to_cpu(src->size) != sizeof(*src_v4)) { dev_err(tplg->dev, "ASoC: invalid PCM size\n"); return -EINVAL; } @@ -1962,7 +1985,7 @@ static int pcm_new_ver(struct soc_tplg *tplg, if (!dest) return -ENOMEM; - dest->size = sizeof(*dest); /* size of latest abi version */ + dest->size = cpu_to_le32(sizeof(*dest)); /* size of latest abi version */ memcpy(dest->pcm_name, src_v4->pcm_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); memcpy(dest->dai_name, src_v4->dai_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dest->pcm_id = src_v4->pcm_id; @@ -1971,7 +1994,7 @@ static int pcm_new_ver(struct soc_tplg *tplg, dest->capture = src_v4->capture; dest->compress = src_v4->compress; dest->num_streams = src_v4->num_streams; - for (i = 0; i < dest->num_streams; i++) + for (i = 0; i < le32_to_cpu(dest->num_streams); i++) memcpy(&dest->stream[i], &src_v4->stream[i], sizeof(struct snd_soc_tplg_stream)); @@ -1986,25 +2009,30 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { struct snd_soc_tplg_pcm *pcm, *_pcm; - int count = hdr->count; + int count; + int size; int i; bool abi_match; + count = le32_to_cpu(hdr->count); + if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) return 0; /* check the element size and count */ pcm = (struct snd_soc_tplg_pcm *)tplg->pos; - if (pcm->size > sizeof(struct snd_soc_tplg_pcm) - || pcm->size < sizeof(struct snd_soc_tplg_pcm_v4)) { + size = le32_to_cpu(pcm->size); + if (size > sizeof(struct snd_soc_tplg_pcm) + || size < sizeof(struct snd_soc_tplg_pcm_v4)) { dev_err(tplg->dev, "ASoC: invalid size %d for PCM elems\n", - pcm->size); + size); return -EINVAL; } if (soc_tplg_check_elem_count(tplg, - pcm->size, count, - hdr->payload_size, "PCM DAI")) { + size, count, + le32_to_cpu(hdr->payload_size), + "PCM DAI")) { dev_err(tplg->dev, "ASoC: invalid count %d for PCM DAI elems\n", count); return -EINVAL; @@ -2012,11 +2040,12 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { pcm = (struct snd_soc_tplg_pcm *)tplg->pos; + size = le32_to_cpu(pcm->size); /* check ABI version by size, create a new version of pcm * if abi not match. */ - if (pcm->size == sizeof(*pcm)) { + if (size == sizeof(*pcm)) { abi_match = true; _pcm = pcm; } else { @@ -2030,7 +2059,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, /* offset by version-specific struct size and * real priv data size */ - tplg->pos += pcm->size + _pcm->priv.size; + tplg->pos += size + le32_to_cpu(_pcm->priv.size); if (!abi_match) kfree(_pcm); /* free the duplicated one */ @@ -2058,12 +2087,13 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, unsigned char invert_bclk, invert_fsync; int i; - for (i = 0; i < cfg->num_hw_configs; i++) { + for (i = 0; i < le32_to_cpu(cfg->num_hw_configs); i++) { hw_config = &cfg->hw_config[i]; if (hw_config->id != cfg->default_hw_config_id) continue; - link->dai_fmt = hw_config->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + link->dai_fmt = le32_to_cpu(hw_config->fmt) & + SND_SOC_DAIFMT_FORMAT_MASK; /* clock gating */ switch (hw_config->clock_gated) { @@ -2127,7 +2157,8 @@ static int link_new_ver(struct soc_tplg *tplg, *link = NULL; - if (src->size != sizeof(struct snd_soc_tplg_link_config_v4)) { + if (le32_to_cpu(src->size) != + sizeof(struct snd_soc_tplg_link_config_v4)) { dev_err(tplg->dev, "ASoC: invalid physical link config size\n"); return -EINVAL; } @@ -2139,10 +2170,10 @@ static int link_new_ver(struct soc_tplg *tplg, if (!dest) return -ENOMEM; - dest->size = sizeof(*dest); + dest->size = cpu_to_le32(sizeof(*dest)); dest->id = src_v4->id; dest->num_streams = src_v4->num_streams; - for (i = 0; i < dest->num_streams; i++) + for (i = 0; i < le32_to_cpu(dest->num_streams); i++) memcpy(&dest->stream[i], &src_v4->stream[i], sizeof(struct snd_soc_tplg_stream)); @@ -2175,7 +2206,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, else stream_name = NULL; - link = snd_soc_find_dai_link(tplg->comp->card, cfg->id, + link = snd_soc_find_dai_link(tplg->comp->card, le32_to_cpu(cfg->id), name, stream_name); if (!link) { dev_err(tplg->dev, "ASoC: physical link %s (id %d) not exist\n", @@ -2189,7 +2220,9 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, /* flags */ if (cfg->flag_mask) - set_link_flags(link, cfg->flag_mask, cfg->flags); + set_link_flags(link, + le32_to_cpu(cfg->flag_mask), + le32_to_cpu(cfg->flags)); /* pass control to component driver for optional further init */ ret = soc_tplg_dai_link_load(tplg, link, cfg); @@ -2213,27 +2246,33 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { struct snd_soc_tplg_link_config *link, *_link; - int count = hdr->count; + int count; + int size; int i, ret; bool abi_match; + count = le32_to_cpu(hdr->count); + if (tplg->pass != SOC_TPLG_PASS_LINK) { - tplg->pos += hdr->size + hdr->payload_size; + tplg->pos += le32_to_cpu(hdr->size) + + le32_to_cpu(hdr->payload_size); return 0; }; /* check the element size and count */ link = (struct snd_soc_tplg_link_config *)tplg->pos; - if (link->size > sizeof(struct snd_soc_tplg_link_config) - || link->size < sizeof(struct snd_soc_tplg_link_config_v4)) { + size = le32_to_cpu(link->size); + if (size > sizeof(struct snd_soc_tplg_link_config) + || size < sizeof(struct snd_soc_tplg_link_config_v4)) { dev_err(tplg->dev, "ASoC: invalid size %d for physical link elems\n", - link->size); + size); return -EINVAL; } if (soc_tplg_check_elem_count(tplg, - link->size, count, - hdr->payload_size, "physical link config")) { + size, count, + le32_to_cpu(hdr->payload_size), + "physical link config")) { dev_err(tplg->dev, "ASoC: invalid count %d for physical link elems\n", count); return -EINVAL; @@ -2242,7 +2281,8 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, /* config physical DAI links */ for (i = 0; i < count; i++) { link = (struct snd_soc_tplg_link_config *)tplg->pos; - if (link->size == sizeof(*link)) { + size = le32_to_cpu(link->size); + if (size == sizeof(*link)) { abi_match = true; _link = link; } else { @@ -2259,7 +2299,7 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, /* offset by version-specific struct size and * real priv data size */ - tplg->pos += link->size + _link->priv.size; + tplg->pos += size + le32_to_cpu(_link->priv.size); if (!abi_match) kfree(_link); /* free the duplicated one */ @@ -2279,13 +2319,15 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, static int soc_tplg_dai_config(struct soc_tplg *tplg, struct snd_soc_tplg_dai *d) { - struct snd_soc_dai_link_component dai_component = {0}; + struct snd_soc_dai_link_component dai_component; struct snd_soc_dai *dai; struct snd_soc_dai_driver *dai_drv; struct snd_soc_pcm_stream *stream; struct snd_soc_tplg_stream_caps *caps; int ret; + memset(&dai_component, 0, sizeof(dai_component)); + dai_component.dai_name = d->dai_name; dai = snd_soc_find_dai(&dai_component); if (!dai) { @@ -2294,7 +2336,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, return -EINVAL; } - if (d->dai_id != dai->id) { + if (le32_to_cpu(d->dai_id) != dai->id) { dev_err(tplg->dev, "ASoC: physical DAI %s id mismatch\n", d->dai_name); return -EINVAL; @@ -2317,7 +2359,9 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, } if (d->flag_mask) - set_dai_flags(dai_drv, d->flag_mask, d->flags); + set_dai_flags(dai_drv, + le32_to_cpu(d->flag_mask), + le32_to_cpu(d->flags)); /* pass control to component driver for optional further init */ ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); @@ -2334,22 +2378,24 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { struct snd_soc_tplg_dai *dai; - int count = hdr->count; + int count; int i; + count = le32_to_cpu(hdr->count); + if (tplg->pass != SOC_TPLG_PASS_BE_DAI) return 0; /* config the existing BE DAIs */ for (i = 0; i < count; i++) { dai = (struct snd_soc_tplg_dai *)tplg->pos; - if (dai->size != sizeof(*dai)) { + if (le32_to_cpu(dai->size) != sizeof(*dai)) { dev_err(tplg->dev, "ASoC: invalid physical DAI size\n"); return -EINVAL; } soc_tplg_dai_config(tplg, dai); - tplg->pos += (sizeof(*dai) + dai->priv.size); + tplg->pos += (sizeof(*dai) + le32_to_cpu(dai->priv.size)); } dev_dbg(tplg->dev, "ASoC: Configure %d BE DAIs\n", count); @@ -2371,25 +2417,28 @@ static int manifest_new_ver(struct soc_tplg *tplg, { struct snd_soc_tplg_manifest *dest; struct snd_soc_tplg_manifest_v4 *src_v4; + int size; *manifest = NULL; - if (src->size != sizeof(*src_v4)) { + size = le32_to_cpu(src->size); + if (size != sizeof(*src_v4)) { dev_warn(tplg->dev, "ASoC: invalid manifest size %d\n", - src->size); - if (src->size) + size); + if (size) return -EINVAL; - src->size = sizeof(*src_v4); + src->size = cpu_to_le32(sizeof(*src_v4)); } dev_warn(tplg->dev, "ASoC: old version of manifest\n"); src_v4 = (struct snd_soc_tplg_manifest_v4 *)src; - dest = kzalloc(sizeof(*dest) + src_v4->priv.size, GFP_KERNEL); + dest = kzalloc(sizeof(*dest) + le32_to_cpu(src_v4->priv.size), + GFP_KERNEL); if (!dest) return -ENOMEM; - dest->size = sizeof(*dest); /* size of latest abi version */ + dest->size = cpu_to_le32(sizeof(*dest)); /* size of latest abi version */ dest->control_elems = src_v4->control_elems; dest->widget_elems = src_v4->widget_elems; dest->graph_elems = src_v4->graph_elems; @@ -2398,7 +2447,7 @@ static int manifest_new_ver(struct soc_tplg *tplg, dest->priv.size = src_v4->priv.size; if (dest->priv.size) memcpy(dest->priv.data, src_v4->priv.data, - src_v4->priv.size); + le32_to_cpu(src_v4->priv.size)); *manifest = dest; return 0; @@ -2417,7 +2466,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, manifest = (struct snd_soc_tplg_manifest *)tplg->pos; /* check ABI version by size, create a new manifest if abi not match */ - if (manifest->size == sizeof(*manifest)) { + if (le32_to_cpu(manifest->size) == sizeof(*manifest)) { abi_match = true; _manifest = manifest; } else { @@ -2444,10 +2493,10 @@ static int soc_valid_header(struct soc_tplg *tplg, if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size) return 0; - if (hdr->size != sizeof(*hdr)) { + if (le32_to_cpu(hdr->size) != sizeof(*hdr)) { dev_err(tplg->dev, "ASoC: invalid header size for type %d at offset 0x%lx size 0x%zx.\n", - hdr->type, soc_tplg_get_hdr_offset(tplg), + le32_to_cpu(hdr->type), soc_tplg_get_hdr_offset(tplg), tplg->fw->size); return -EINVAL; } @@ -2461,7 +2510,7 @@ static int soc_valid_header(struct soc_tplg *tplg, return -EINVAL; } - if (hdr->magic != SND_SOC_TPLG_MAGIC) { + if (le32_to_cpu(hdr->magic) != SND_SOC_TPLG_MAGIC) { dev_err(tplg->dev, "ASoC: pass %d does not have a valid header got %x at offset 0x%lx size 0x%zx.\n", tplg->pass, hdr->magic, @@ -2470,8 +2519,8 @@ static int soc_valid_header(struct soc_tplg *tplg, } /* Support ABI from version 4 */ - if (hdr->abi > SND_SOC_TPLG_ABI_VERSION - || hdr->abi < SND_SOC_TPLG_ABI_VERSION_MIN) { + if (le32_to_cpu(hdr->abi) > SND_SOC_TPLG_ABI_VERSION || + le32_to_cpu(hdr->abi) < SND_SOC_TPLG_ABI_VERSION_MIN) { dev_err(tplg->dev, "ASoC: pass %d invalid ABI version got 0x%x need 0x%x at offset 0x%lx size 0x%zx.\n", tplg->pass, hdr->abi, @@ -2486,7 +2535,7 @@ static int soc_valid_header(struct soc_tplg *tplg, return -EINVAL; } - if (tplg->pass == hdr->type) + if (tplg->pass == le32_to_cpu(hdr->type)) dev_dbg(tplg->dev, "ASoC: Got 0x%x bytes of type %d version %d vendor %d at pass %d\n", hdr->payload_size, hdr->type, hdr->version, @@ -2502,13 +2551,13 @@ static int soc_tplg_load_header(struct soc_tplg *tplg, tplg->pos = tplg->hdr_pos + sizeof(struct snd_soc_tplg_hdr); /* check for matching ID */ - if (hdr->index != tplg->req_index && + if (le32_to_cpu(hdr->index) != tplg->req_index && tplg->req_index != SND_SOC_TPLG_INDEX_ALL) return 0; - tplg->index = hdr->index; + tplg->index = le32_to_cpu(hdr->index); - switch (hdr->type) { + switch (le32_to_cpu(hdr->type)) { case SND_SOC_TPLG_TYPE_MIXER: case SND_SOC_TPLG_TYPE_ENUM: case SND_SOC_TPLG_TYPE_BYTES: @@ -2564,7 +2613,7 @@ static int soc_tplg_process_headers(struct soc_tplg *tplg) return ret; /* goto next header */ - tplg->hdr_pos += hdr->payload_size + + tplg->hdr_pos += le32_to_cpu(hdr->payload_size) + sizeof(struct snd_soc_tplg_hdr); hdr = (struct snd_soc_tplg_hdr *)tplg->hdr_pos; } -- cgit v1.2.3-59-g8ed1b From 2114171d9cce1a897bee394b06f6c224247f095c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 4 Apr 2019 14:13:58 -0500 Subject: ASoC: topology: fix big-endian check Use an explicit define to avoid Sparse issues coming from the use of cpu_to_be32 Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 03c4dbdfc584..51903ca7614b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -30,6 +30,8 @@ #include #include +#define SOC_TPLG_MAGIC_BIG_ENDIAN 0x436F5341 /* ASoC in reverse */ + /* * We make several passes over the data (since it wont necessarily be ordered) * and process objects in the following order. This guarantees the component @@ -2502,7 +2504,7 @@ static int soc_valid_header(struct soc_tplg *tplg, } /* big endian firmware objects not supported atm */ - if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) { + if (hdr->magic == SOC_TPLG_MAGIC_BIG_ENDIAN) { dev_err(tplg->dev, "ASoC: pass %d big endian not supported header got %x at offset 0x%lx size 0x%zx.\n", tplg->pass, hdr->magic, -- cgit v1.2.3-59-g8ed1b From 1997ee89f36da906efb8e700f1d08368c73883be Mon Sep 17 00:00:00 2001 From: "S.j. Wang" Date: Thu, 4 Apr 2019 09:40:56 +0000 Subject: ASoC: fsl_esai: Support synchronous mode In ESAI synchronous mode, the clock is generated by Tx, So we should always set registers of Tx which relate with the bit clock and frame clock generation (TCCR, TCR, ECR), even there is only Rx is working. Signed-off-by: Shengjiu Wang Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 3623aa9a6f2e..c7410bbfd2af 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -218,7 +218,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); struct clk *clksrc = esai_priv->extalclk; - bool tx = clk_id <= ESAI_HCKT_EXTAL; + bool tx = (clk_id <= ESAI_HCKT_EXTAL || esai_priv->synchronous); bool in = dir == SND_SOC_CLOCK_IN; u32 ratio, ecr = 0; unsigned long clk_rate; @@ -253,7 +253,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, ecr |= ESAI_ECR_ETI; /* fall through */ case ESAI_HCKR_EXTAL: - ecr |= ESAI_ECR_ERI; + ecr |= esai_priv->synchronous ? ESAI_ECR_ETI : ESAI_ECR_ERI; break; default: return -EINVAL; @@ -537,10 +537,18 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, bclk = params_rate(params) * slot_width * esai_priv->slots; - ret = fsl_esai_set_bclk(dai, tx, bclk); + ret = fsl_esai_set_bclk(dai, esai_priv->synchronous || tx, bclk); if (ret) return ret; + mask = ESAI_xCR_xSWS_MASK; + val = ESAI_xCR_xSWS(slot_width, width); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + /* Recording in synchronous mode needs to set TCR also */ + if (!tx && esai_priv->synchronous) + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, val); + /* Use Normal mode to support monaural audio */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ? @@ -556,10 +564,9 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); - mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); - val = ESAI_xCR_xSWS(slot_width, width) | (tx ? ESAI_xCR_PADC : 0); - - regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + if (tx) + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_PADC, ESAI_xCR_PADC); /* Remove ESAI personal reset by configuring ESAI_PCRC and ESAI_PRRC */ regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, -- cgit v1.2.3-59-g8ed1b From 4110e9a1868332e305f64d5c5b32368597caeb8f Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Wed, 3 Apr 2019 15:23:35 +0200 Subject: ASoC: cs42l51: change mic bias DAPM Use SND_SOC_DAPM_SUPPLY for mic bias DAPM instead of deprecated SND_SOC_DAPM_MICBIAS. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 58ece037d944..991e4ebd7a04 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -210,7 +210,8 @@ static const struct snd_kcontrol_new cs42l51_adcr_mux_controls = SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum); static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { - SND_SOC_DAPM_MICBIAS("Mic Bias", CS42L51_MIC_POWER_CTL, 1, 1), + SND_SOC_DAPM_SUPPLY("Mic Bias", CS42L51_MIC_POWER_CTL, 1, 1, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_PGA_E("Left PGA", CS42L51_POWER_CTL1, 3, 1, NULL, 0, cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), SND_SOC_DAPM_PGA_E("Right PGA", CS42L51_POWER_CTL1, 4, 1, NULL, 0, -- cgit v1.2.3-59-g8ed1b From 1c05c1ed6f4dd267a2ca30bacd63870cd5ecc860 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:49:51 +0300 Subject: ASoC: imx-pcm: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm.h | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 133c4470acad..5dd406774d3e 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -1,13 +1,9 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ /* * Copyright 2009 Sascha Hauer * * This code is based on code copyrighted by Freescale, * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. */ #ifndef _IMX_PCM_H -- cgit v1.2.3-59-g8ed1b From ecf4630f1b9857141e38ea798bc30b8fd7c11dfc Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:49:53 +0300 Subject: ASoC: eukrea-tlv320: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/eukrea-tlv320.c | 26 ++++++++++---------------- 1 file changed, 10 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 30a3d68b5c03..d648268cb454 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -1,19 +1,13 @@ -/* - * eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode - * - * Copyright 2010 Eric Bénard, Eukréa Electromatique - * - * based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c - * which is Copyright 2009 Simtec Electronics - * and on sound/soc/imx/phycore-ac97.c which is - * Copyright 2009 Sascha Hauer, Pengutronix - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// eukrea-tlv320.c -- SoC audio for eukrea_cpuimxXX in I2S mode +// +// Copyright 2010 Eric Bénard, Eukréa Electromatique +// +// based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +// which is Copyright 2009 Simtec Electronics +// and on sound/soc/imx/phycore-ac97.c which is +// Copyright 2009 Sascha Hauer, Pengutronix #include #include -- cgit v1.2.3-59-g8ed1b From e073564fe653eacbef6723a0a5210131b3998000 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:49:54 +0300 Subject: ASoC: fsl-dma: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 26 +++++++++++--------------- sound/soc/fsl/fsl_dma.h | 5 +---- 2 files changed, 12 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 78871de35086..e22508301412 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -1,18 +1,14 @@ -/* - * Freescale DMA ALSA SoC PCM driver - * - * Author: Timur Tabi - * - * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - * - * This driver implements ASoC support for the Elo DMA controller, which is - * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, - * the PCM driver is what handles the DMA buffer. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale DMA ALSA SoC PCM driver +// +// Author: Timur Tabi +// +// Copyright 2007-2010 Freescale Semiconductor, Inc. +// +// This driver implements ASoC support for the Elo DMA controller, which is +// the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, +// the PCM driver is what handles the DMA buffer. #include #include diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 78fee97e8036..f19ae765b656 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -1,9 +1,6 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * mpc8610-pcm.h - ALSA PCM interface for the Freescale MPC8610 SoC - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef _MPC8610_PCM_H -- cgit v1.2.3-59-g8ed1b From 47bbcbfeb8ac99164067485a8322d92e69044211 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:49:55 +0300 Subject: ASoC: imx-audmux: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 99e07b01a2ce..04e59e66711d 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -1,21 +1,11 @@ -/* - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * Copyright 2009 Pengutronix, Sascha Hauer - * - * Initial development of this code was funded by - * Phytec Messtechnik GmbH, http://www.phytec.de - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright 2012 Freescale Semiconductor, Inc. +// Copyright 2012 Linaro Ltd. +// Copyright 2009 Pengutronix, Sascha Hauer +// +// Initial development of this code was funded by +// Phytec Messtechnik GmbH, http://www.phytec.de #include #include -- cgit v1.2.3-59-g8ed1b From 0fe61fc0c243b965d941f24e8b25cfc94697757a Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:49:56 +0300 Subject: ASoC: imx-es8328: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/imx-es8328.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 9953438086e4..c9d8739b04a9 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -1,14 +1,7 @@ -/* - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright 2012 Freescale Semiconductor, Inc. +// Copyright 2012 Linaro Ltd. #include #include -- cgit v1.2.3-59-g8ed1b From a134f42eaacaa4d31785d51cf19a45845721b212 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:49:57 +0300 Subject: ASoC: imx-mc13783: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/imx-mc13783.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 9d19b808f634..545815a27074 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -1,17 +1,11 @@ -/* - * imx-mc13783.c -- SoC audio for imx based boards with mc13783 codec - * - * Copyright 2012 Philippe Retornaz, - * - * Heavly based on phycore-mc13783: - * Copyright 2009 Sascha Hauer, Pengutronix - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// imx-mc13783.c -- SoC audio for imx based boards with mc13783 codec +// +// Copyright 2012 Philippe Retornaz, +// +// Heavly based on phycore-mc13783: +// Copyright 2009 Sascha Hauer, Pengutronix #include #include -- cgit v1.2.3-59-g8ed1b From d4451c159a0b63efd5028d62cae501737d8679fa Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:49:58 +0300 Subject: ASoC: imx-pcm-fiq: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-fiq.c | 21 ++++++++------------- 1 file changed, 8 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 0578f3486847..c49aea4fba56 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -1,16 +1,11 @@ -/* - * imx-pcm-fiq.c -- ALSA Soc Audio Layer - * - * Copyright 2009 Sascha Hauer - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// imx-pcm-fiq.c -- ALSA Soc Audio Layer +// +// Copyright 2009 Sascha Hauer +// +// This code is based on code copyrighted by Freescale, +// Liam Girdwood, Javier Martin and probably others. + #include #include #include -- cgit v1.2.3-59-g8ed1b From 58dbd1015d3ed47d53a99e7c4da0bbd3543177da Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:49:59 +0300 Subject: ASoC: imx-spdif: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/imx-spdif.c | 13 +++---------- 1 file changed, 3 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 797d66e43d49..4f7f210beb18 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -1,13 +1,6 @@ -/* - * Copyright (C) 2013 Freescale Semiconductor, Inc. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright (C) 2013 Freescale Semiconductor, Inc. #include #include -- cgit v1.2.3-59-g8ed1b From bdc16b5748926412ca90cf0521c718a46e2ca2b7 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:00 +0300 Subject: ASoC: imx-ssi: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 57 ++++++++++++++++++++++--------------------------- sound/soc/fsl/imx-ssi.h | 6 +----- 2 files changed, 26 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 06790615e04e..9038b61317be 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -1,35 +1,28 @@ -/* - * imx-ssi.c -- ALSA Soc Audio Layer - * - * Copyright 2009 Sascha Hauer - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * - * The i.MX SSI core has some nasty limitations in AC97 mode. While most - * sane processor vendors have a FIFO per AC97 slot, the i.MX has only - * one FIFO which combines all valid receive slots. We cannot even select - * which slots we want to receive. The WM9712 with which this driver - * was developed with always sends GPIO status data in slot 12 which - * we receive in our (PCM-) data stream. The only chance we have is to - * manually skip this data in the FIQ handler. With sampling rates different - * from 48000Hz not every frame has valid receive data, so the ratio - * between pcm data and GPIO status data changes. Our FIQ handler is not - * able to handle this, hence this driver only works with 48000Hz sampling - * rate. - * Reading and writing AC97 registers is another challenge. The core - * provides us status bits when the read register is updated with *another* - * value. When we read the same register two times (and the register still - * contains the same value) these status bits are not set. We work - * around this by not polling these bits but only wait a fixed delay. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// imx-ssi.c -- ALSA Soc Audio Layer +// +// Copyright 2009 Sascha Hauer +// +// This code is based on code copyrighted by Freescale, +// Liam Girdwood, Javier Martin and probably others. +// +// The i.MX SSI core has some nasty limitations in AC97 mode. While most +// sane processor vendors have a FIFO per AC97 slot, the i.MX has only +// one FIFO which combines all valid receive slots. We cannot even select +// which slots we want to receive. The WM9712 with which this driver +// was developed with always sends GPIO status data in slot 12 which +// we receive in our (PCM-) data stream. The only chance we have is to +// manually skip this data in the FIQ handler. With sampling rates different +// from 48000Hz not every frame has valid receive data, so the ratio +// between pcm data and GPIO status data changes. Our FIQ handler is not +// able to handle this, hence this driver only works with 48000Hz sampling +// rate. +// Reading and writing AC97 registers is another challenge. The core +// provides us status bits when the read register is updated with *another* +// value. When we read the same register two times (and the register still +// contains the same value) these status bits are not set. We work +// around this by not polling these bits but only wait a fixed delay. #include #include diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index be6562365b6a..19cd0937e740 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -1,8 +1,4 @@ -/* - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +/* SPDX-License-Identifier: GPL-2.0 */ #ifndef _IMX_SSI_H #define _IMX_SSI_H -- cgit v1.2.3-59-g8ed1b From 1edfc2485d8dcf4c766a5be8049cab1cddc6700d Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:01 +0300 Subject: ASoC: mpc5200_dma: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index c1a4544eb16b..4396442c2fdd 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -1,10 +1,10 @@ -/* - * Freescale MPC5200 PSC DMA - * ALSA SoC Platform driver - * - * Copyright (C) 2008 Secret Lab Technologies Ltd. - * Copyright (C) 2009 Jon Smirl, Digispeaker - */ +// SPDX-License-Identifier: GPL +// +// Freescale MPC5200 PSC DMA +// ALSA SoC Platform driver +// +// Copyright (C) 2008 Secret Lab Technologies Ltd. +// Copyright (C) 2009 Jon Smirl, Digispeaker #include #include -- cgit v1.2.3-59-g8ed1b From 1af301d46246942a00e15294603dbd3abfb41a82 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:02 +0300 Subject: ASoC: mpc5200_psc_ac97: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_psc_ac97.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 07ee355ee385..e5b9c04d1565 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -1,13 +1,9 @@ -/* - * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. - * - * Copyright (C) 2009 Jon Smirl, Digispeaker - * Author: Jon Smirl - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. +// +// Copyright (C) 2009 Jon Smirl, Digispeaker +// Author: Jon Smirl #include #include -- cgit v1.2.3-59-g8ed1b From 864a8472c4412dd9bedb6484ce41ec906e812d57 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:03 +0300 Subject: ASoC: mpc5200_psc_i2s: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_psc_i2s.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index d8232943ccb6..6de97461ba25 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -1,10 +1,10 @@ -/* - * Freescale MPC5200 PSC in I2S mode - * ALSA SoC Digital Audio Interface (DAI) driver - * - * Copyright (C) 2008 Secret Lab Technologies Ltd. - * Copyright (C) 2009 Jon Smirl, Digispeaker - */ +// SPDX-License-Identifier: GPL +// +// Freescale MPC5200 PSC in I2S mode +// ALSA SoC Digital Audio Interface (DAI) driver +// +// Copyright (C) 2008 Secret Lab Technologies Ltd. +// Copyright (C) 2009 Jon Smirl, Digispeaker #include #include -- cgit v1.2.3-59-g8ed1b From eed9bb238e08619547585e563499f79a700109b8 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:04 +0300 Subject: ASoC: mpc8610_hpcd: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/mpc8610_hpcd.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a639b52c16f6..f6261a3eeb0f 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -1,14 +1,10 @@ -/** - * Freescale MPC8610HPCD ALSA SoC Machine driver - * - * Author: Timur Tabi - * - * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale MPC8610HPCD ALSA SoC Machine driver +// +// Author: Timur Tabi +// +// Copyright 2007-2010 Freescale Semiconductor, Inc. #include #include -- cgit v1.2.3-59-g8ed1b From 1021ec41a909ee7961f63f38afd90b5ccc592c49 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:05 +0300 Subject: ASoC: mx27vis-aic32x4: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/mx27vis-aic32x4.c | 29 +++++++---------------------- 1 file changed, 7 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index d7ec3d20065c..37a4520aef62 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -1,25 +1,10 @@ -/* - * mx27vis-aic32x4.c - * - * Copyright 2011 Vista Silicon S.L. - * - * Author: Javier Martin - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, - * MA 02110-1301, USA. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// mx27vis-aic32x4.c +// +// Copyright 2011 Vista Silicon S.L. +// +// Author: Javier Martin #include #include -- cgit v1.2.3-59-g8ed1b From ab78529cfec09bf1a4327d48c4ba6d3d159a1540 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:06 +0300 Subject: ASoC: p1022_ds: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/p1022_ds.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 41c623c55c16..80384f70878d 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -1,14 +1,10 @@ -/** - * Freescale P1022DS ALSA SoC Machine driver - * - * Author: Timur Tabi - * - * Copyright 2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale P1022DS ALSA SoC Machine driver +// +// Author: Timur Tabi +// +// Copyright 2010 Freescale Semiconductor, Inc. #include #include -- cgit v1.2.3-59-g8ed1b From 8acfeb65cbaa92ff2e4f3305c7a18a6edabd1d92 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:07 +0300 Subject: ASoC: p1022_rdk: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/p1022_rdk.c | 32 ++++++++++++++------------------ 1 file changed, 14 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index 4afbdd610bfa..1c32c2d8c6b0 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -1,21 +1,17 @@ -/** - * Freescale P1022RDK ALSA SoC Machine driver - * - * Author: Timur Tabi - * - * Copyright 2012 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - * - * Note: in order for audio to work correctly, the output controls need - * to be enabled, because they control the clock. So for playback, for - * example: - * - * amixer sset 'Left Output Mixer PCM' on - * amixer sset 'Right Output Mixer PCM' on - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale P1022RDK ALSA SoC Machine driver +// +// Author: Timur Tabi +// +// Copyright 2012 Freescale Semiconductor, Inc. +// +// Note: in order for audio to work correctly, the output controls need +// to be enabled, because they control the clock. So for playback, for +// example: +// +// amixer sset 'Left Output Mixer PCM' on +// amixer sset 'Right Output Mixer PCM' on #include #include -- cgit v1.2.3-59-g8ed1b From 6e3398c67e682486b1f18672519a8e607d902172 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:08 +0300 Subject: ASoC: pcm030-audio-fabric: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/pcm030-audio-fabric.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index e339f36cea95..a7fe4ad25c52 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -1,14 +1,10 @@ -/* - * Phytec pcm030 driver for the PSC of the Freescale MPC52xx - * configured as AC97 interface - * - * Copyright 2008 Jon Smirl, Digispeaker - * Author: Jon Smirl - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Phytec pcm030 driver for the PSC of the Freescale MPC52xx +// configured as AC97 interface +// +// Copyright 2008 Jon Smirl, Digispeaker +// Author: Jon Smirl #include #include -- cgit v1.2.3-59-g8ed1b From e5eec5ee98a6379b6a515beb27db4f9a80bd9276 Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:09 +0300 Subject: ASoC: phycore-ac97: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/phycore-ac97.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index 66fb6c4614d2..fe7ba6db7c96 100644 --- a/sound/soc/fsl/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c @@ -1,14 +1,8 @@ -/* - * phycore-ac97.c -- SoC audio for imx_phycore in AC97 mode - * - * Copyright 2009 Sascha Hauer, Pengutronix - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// phycore-ac97.c -- SoC audio for imx_phycore in AC97 mode +// +// Copyright 2009 Sascha Hauer, Pengutronix #include #include -- cgit v1.2.3-59-g8ed1b From a03584b75a28450a2a64646e3c1b3d6c48aacf3b Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Fri, 5 Apr 2019 14:50:10 +0300 Subject: ASoC: wm1133-ev1: Switch to SPDX identifier Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/wm1133-ev1.c | 21 ++++++++------------- 1 file changed, 8 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index 2f80b21b2921..aad24ccbef90 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -1,16 +1,11 @@ -/* - * wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS - * - * Copyright (c) 2010 Wolfson Microelectronics plc - * Author: Mark Brown - * - * Based on an earlier driver for the same hardware by Liam Girdwood. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS +// +// Copyright (c) 2010 Wolfson Microelectronics plc +// Author: Mark Brown +// +// Based on an earlier driver for the same hardware by Liam Girdwood. #include #include -- cgit v1.2.3-59-g8ed1b From ba49cf6f8e4ad44ee752eb13413a26a7191893ae Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 29 Mar 2019 22:50:44 +0100 Subject: ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect Add a quirk to support boards whose jack detection mechanism is inverted. It will set the 'everest,jack-detect-inverted' boolean device property for the es8316 codec driver. Signed-off-by: Paul Cercueil Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 6937c00cf63d..ccf552e92d53 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -40,6 +40,9 @@ #include "../atom/sst-atom-controls.h" #include "../common/sst-dsp.h" +/* jd-inv + terminating entry */ +#define MAX_NO_PROPS 2 + struct byt_cht_es8316_private { struct clk *mclk; struct snd_soc_jack jack; @@ -55,6 +58,7 @@ enum { #define BYT_CHT_ES8316_MAP(quirk) ((quirk) & GENMASK(3, 0)) #define BYT_CHT_ES8316_SSP0 BIT(16) #define BYT_CHT_ES8316_MONO_SPEAKER BIT(17) +#define BYT_CHT_ES8316_JD_INVERTED BIT(18) static int quirk; @@ -72,6 +76,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0 enabled"); if (quirk & BYT_CHT_ES8316_MONO_SPEAKER) dev_info(dev, "quirk MONO_SPEAKER enabled\n"); + if (quirk & BYT_CHT_ES8316_JD_INVERTED) + dev_info(dev, "quirk JD_INVERTED enabled\n"); } static int byt_cht_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, @@ -438,12 +444,14 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { static const char * const mic_name[] = { "in1", "in2" }; + struct property_entry props[MAX_NO_PROPS] = {}; struct byt_cht_es8316_private *priv; struct device *dev = &pdev->dev; struct snd_soc_acpi_mach *mach; const char *platform_name; struct acpi_device *adev; struct device *codec_dev; + unsigned int cnt = 0; int dai_index = 0; int i; int ret = 0; @@ -513,6 +521,15 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) if (!codec_dev) return -EPROBE_DEFER; + if (quirk & BYT_CHT_ES8316_JD_INVERTED) + props[cnt++] = PROPERTY_ENTRY_BOOL("everest,jack-detect-inverted"); + + if (cnt) { + ret = device_add_properties(codec_dev, props); + if (ret) + return ret; + } + devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios); priv->speaker_en_gpio = gpiod_get_index(codec_dev, "speaker-enable", 0, -- cgit v1.2.3-59-g8ed1b From a8d218f4fe8116395e4c4a13b63029ecf00d4035 Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 29 Mar 2019 22:50:45 +0100 Subject: ASoC: Intel: bytcht_es8316: Add quirk for the Teclast X98+ II The Teclast X98+ II is a Cherrytrail tablet, which require two quirks: - it has stereo speakers, - its jack detection mechanism is inverted. Signed-off-by: Paul Cercueil Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index ccf552e92d53..38975827e276 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include #include @@ -441,11 +442,25 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = { { }, }; +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { + { /* Teclast X98 Plus II */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "TECLAST"), + DMI_MATCH(DMI_PRODUCT_NAME, "X98 Plus II"), + }, + .driver_data = (void *)(BYT_CHT_ES8316_INTMIC_IN1_MAP + | BYT_CHT_ES8316_JD_INVERTED), + }, + {} +}; + static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { static const char * const mic_name[] = { "in1", "in2" }; struct property_entry props[MAX_NO_PROPS] = {}; struct byt_cht_es8316_private *priv; + const struct dmi_system_id *dmi_id; struct device *dev = &pdev->dev; struct snd_soc_acpi_mach *mach; const char *platform_name; @@ -488,7 +503,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) return ret; /* Check for BYTCR or other platform and setup quirks */ - if (x86_match_cpu(baytrail_cpu_ids) && + dmi_id = dmi_first_match(byt_cht_es8316_quirk_table); + if (dmi_id) { + quirk = (int)dmi_id->driver_data; + } else if (x86_match_cpu(baytrail_cpu_ids) && mach->mach_params.acpi_ipc_irq_index == 0) { /* On BYTCR default to SSP0, internal-mic-in2-map, mono-spk */ quirk = BYT_CHT_ES8316_SSP0 | BYT_CHT_ES8316_INTMIC_IN2_MAP | -- cgit v1.2.3-59-g8ed1b From 2390c9b473da217ee086e5382869ade82c217a30 Mon Sep 17 00:00:00 2001 From: Jenny TC Date: Tue, 26 Mar 2019 22:50:56 +0530 Subject: ASoC: Intel: kbl_da7219_max98927: remove headset kbl_da7219_max98927 supports two configurations - One with da7219 & max98927/373 and other one with max98927/373 alone. This patch removes the headset dais from max98927/373 configuration. Since the snd_soc_dapm_ignore_suspend for DMIC is called from da7219_codec_init, the code is re-arranged to invoke snd_soc_dapm_ignore_suspend from kabylake_dmic_init. Signed-off-by: Jenny TC Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_da7219_max98927.c | 52 ++++++++++++---------------- 1 file changed, 23 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 2768a572d065..f72a7bf028d7 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -52,7 +52,6 @@ struct kbl_codec_private { enum { KBL_DPCM_AUDIO_PB = 0, - KBL_DPCM_AUDIO_CP, KBL_DPCM_AUDIO_ECHO_REF_CP, KBL_DPCM_AUDIO_REF_CP, KBL_DPCM_AUDIO_DMIC_CP, @@ -60,6 +59,7 @@ enum { KBL_DPCM_AUDIO_HDMI2_PB, KBL_DPCM_AUDIO_HDMI3_PB, KBL_DPCM_AUDIO_HS_PB, + KBL_DPCM_AUDIO_CP, }; static int platform_clock_control(struct snd_soc_dapm_widget *w, @@ -311,6 +311,12 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) da7219_aad_jack_det(component, &ctx->kabylake_headset); + return 0; +} + +static int kabylake_dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); if (ret) dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret); @@ -581,20 +587,6 @@ static struct snd_soc_dai_link kabylake_dais[] = { .dpcm_playback = 1, .ops = &kabylake_da7219_fe_ops, }, - [KBL_DPCM_AUDIO_CP] = { - .name = "Kbl Audio Capture Port", - .stream_name = "Audio Record", - .cpu_dai_name = "System Pin", - .platform_name = "0000:00:1f.3", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &kabylake_da7219_fe_ops, - }, [KBL_DPCM_AUDIO_ECHO_REF_CP] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", @@ -690,6 +682,20 @@ static struct snd_soc_dai_link kabylake_dais[] = { .ops = &kabylake_da7219_fe_ops, }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, + }, /* Back End DAI links */ { @@ -733,6 +739,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", + .init = kabylake_dmic_init, .platform_name = "0000:00:1f.3", .be_hw_params_fixup = kabylake_dmic_fixup, .ignore_suspend = 1, @@ -792,20 +799,6 @@ static struct snd_soc_dai_link kabylake_max98_927_373_dais[] = { .dpcm_playback = 1, .ops = &kabylake_da7219_fe_ops, }, - [KBL_DPCM_AUDIO_CP] = { - .name = "Kbl Audio Capture Port", - .stream_name = "Audio Record", - .cpu_dai_name = "System Pin", - .platform_name = "0000:00:1f.3", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &kabylake_da7219_fe_ops, - }, [KBL_DPCM_AUDIO_ECHO_REF_CP] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", @@ -911,6 +904,7 @@ static struct snd_soc_dai_link kabylake_max98_927_373_dais[] = { .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", + .init = kabylake_dmic_init, .platform_name = "0000:00:1f.3", .be_hw_params_fixup = kabylake_dmic_fixup, .ignore_suspend = 1, -- cgit v1.2.3-59-g8ed1b From daae4c583534a3e20f08701ef6659066ebf60ca7 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Fri, 5 Apr 2019 14:54:30 -0700 Subject: ASoC: tlv320aic32x4: Fix spacing Fix some cosmetic spacing issues reported by Julia Lawall . Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4-clk.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 667ec2c03508..9e4899eb1d8e 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -165,8 +165,8 @@ static int clk_aic32x4_pll_calc_muldiv(struct clk_aic32x4_pll_muldiv *settings, * of the multiplier. This is because we can't do floating point * math in the kernel. */ - multiplier = (u64) rate * settings->p * 10000; - do_div(multiplier, parent_rate); + multiplier = (u64) rate * settings->p * 10000; + do_div(multiplier, parent_rate); /* * J can't be over 64, so R can scale this. -- cgit v1.2.3-59-g8ed1b From df55531b8b0eea9d2473f5697ae4f38d0df6bec7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Apr 2019 12:25:32 +0200 Subject: ALSA: timer: Revert active callback sync check at close This is essentially a revert of the commit a7588c896b05 ("ALSA: timer: Check ack_list emptiness instead of bit flag"). The intended change by the commit turns out to be insufficient, as snd_timer_close*() always calls snd_timer_stop() that deletes the ack_list beforehand. In theory, we can change the behavior of snd_timer_stop() to sync the pending ack_list, but this will become a deadlock for the callback like sequencer that calls again snd_timer_stop() from itself. So, reverting the change is a more straightforward solution. Fixes: a7588c896b05 ("ALSA: timer: Check ack_list emptiness instead of bit flag") Reported-by: syzbot+58813d77154713f4de15@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai --- include/sound/timer.h | 1 + sound/core/timer.c | 11 ++++++----- 2 files changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/include/sound/timer.h b/include/sound/timer.h index bcfee20ea226..7ae226ab6990 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -43,6 +43,7 @@ #define SNDRV_TIMER_IFLG_START 0x00000004 #define SNDRV_TIMER_IFLG_AUTO 0x00000008 /* auto restart */ #define SNDRV_TIMER_IFLG_FAST 0x00000010 /* fast callback (do not use tasklet) */ +#define SNDRV_TIMER_IFLG_CALLBACK 0x00000020 /* timer callback is active */ #define SNDRV_TIMER_IFLG_EXCLUSIVE 0x00000040 /* exclusive owner - no more instances */ #define SNDRV_TIMER_IFLG_EARLY_EVENT 0x00000080 /* write early event to the poll queue */ diff --git a/sound/core/timer.c b/sound/core/timer.c index bb7e90ab90f8..df52d2960179 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -372,7 +372,7 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri) timer->num_instances--; /* wait, until the active callback is finished */ spin_lock_irq(&timer->lock); - while (!list_empty(&timeri->ack_list)) { + while (timeri->flags & SNDRV_TIMER_IFLG_CALLBACK) { spin_unlock_irq(&timer->lock); udelay(10); spin_lock_irq(&timer->lock); @@ -748,19 +748,20 @@ static void snd_timer_process_callbacks(struct snd_timer *timer, ti = list_first_entry(head, struct snd_timer_instance, ack_list); + /* remove from ack_list and make empty */ + list_del_init(&ti->ack_list); + if (!(ti->flags & SNDRV_TIMER_IFLG_DEAD)) { ticks = ti->pticks; ti->pticks = 0; resolution = ti->resolution; - + ti->flags |= SNDRV_TIMER_IFLG_CALLBACK; spin_unlock(&timer->lock); if (ti->callback) ti->callback(ti, resolution, ticks); spin_lock(&timer->lock); + ti->flags &= ~SNDRV_TIMER_IFLG_CALLBACK; } - - /* remove from ack_list and make empty */ - list_del_init(&ti->ack_list); } } -- cgit v1.2.3-59-g8ed1b From fd7ae83de11a597cbe15770382f101c784a79b1c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Mar 2019 16:09:45 +0100 Subject: ALSA: seq: Use kvmalloc() for cell pools Use kvmalloc() for allocating cell pools since the pool size can be relatively small that may be covered better by slab. Signed-off-by: Takashi Iwai --- sound/core/seq/seq_memory.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 5b0388202bac..6ea4d8a5a71e 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -24,7 +24,7 @@ #include #include #include -#include +#include #include #include @@ -389,8 +389,8 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) if (snd_BUG_ON(!pool)) return -EINVAL; - cellptr = vmalloc(array_size(sizeof(struct snd_seq_event_cell), - pool->size)); + cellptr = kvmalloc_array(sizeof(struct snd_seq_event_cell), pool->size, + GFP_KERNEL); if (!cellptr) return -ENOMEM; @@ -398,7 +398,7 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) spin_lock_irqsave(&pool->lock, flags); if (pool->ptr) { spin_unlock_irqrestore(&pool->lock, flags); - vfree(cellptr); + kvfree(cellptr); return 0; } @@ -456,7 +456,7 @@ int snd_seq_pool_done(struct snd_seq_pool *pool) pool->total_elements = 0; spin_unlock_irqrestore(&pool->lock, flags); - vfree(ptr); + kvfree(ptr); spin_lock_irqsave(&pool->lock, flags); pool->closing = 0; -- cgit v1.2.3-59-g8ed1b From 4b24b960b10b6a4e30beba3ce097fa867b4a085f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Mar 2019 15:55:08 +0100 Subject: ALSA: seq: Align temporary re-locking with irqsave version In a few places in sequencer core, we temporarily unlock / re-lock the pool spin lock while waiting for the allocation in the blocking mode. There spin_unlock_irq() / spin_lock_irq() pairs are called while initially spin_lock_irqsave() is used (and spin_lock_irqrestore() at the end of the function again). This is likely OK for now, but it's a bit confusing and error-prone. This patch replaces these temporary relocking lines with the irqsave variant to make the lock/unlock sequence more consistently. Signed-off-by: Takashi Iwai --- sound/core/seq/seq_fifo.c | 4 ++-- sound/core/seq/seq_memory.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 72c0302a55d2..613ae10d33b8 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -195,9 +195,9 @@ int snd_seq_fifo_cell_out(struct snd_seq_fifo *f, } set_current_state(TASK_INTERRUPTIBLE); add_wait_queue(&f->input_sleep, &wait); - spin_unlock_irq(&f->lock); + spin_unlock_irqrestore(&f->lock, flags); schedule(); - spin_lock_irq(&f->lock); + spin_lock_irqsave(&f->lock, flags); remove_wait_queue(&f->input_sleep, &wait); if (signal_pending(current)) { spin_unlock_irqrestore(&f->lock, flags); diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 6ea4d8a5a71e..ae0b8971f6ce 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -244,13 +244,13 @@ static int snd_seq_cell_alloc(struct snd_seq_pool *pool, set_current_state(TASK_INTERRUPTIBLE); add_wait_queue(&pool->output_sleep, &wait); - spin_unlock_irq(&pool->lock); + spin_unlock_irqrestore(&pool->lock, flags); if (mutexp) mutex_unlock(mutexp); schedule(); if (mutexp) mutex_lock(mutexp); - spin_lock_irq(&pool->lock); + spin_lock_irqsave(&pool->lock, flags); remove_wait_queue(&pool->output_sleep, &wait); /* interrupted? */ if (signal_pending(current)) { -- cgit v1.2.3-59-g8ed1b From f823b8a75527dca0b93cf577bbabbe47fd79b2a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Mar 2019 16:21:01 +0100 Subject: ALSA: seq: Remove superfluous irqsave flags spin_lock_irqsave() is used unnecessarily in various places in sequencer core code although it's pretty obvious that the context is sleepable. Remove irqsave and use the plain spin_lock_irq() in such places for simplicity. Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 19 ++++++++----------- sound/core/seq/seq_fifo.c | 10 ++++------ sound/core/seq/seq_memory.c | 16 +++++++--------- sound/core/seq/seq_ports.c | 15 ++++++--------- 4 files changed, 25 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 7d4640d1fe9f..933bde3843d9 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -203,7 +203,6 @@ int __init client_init_data(void) static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) { - unsigned long flags; int c; struct snd_seq_client *client; @@ -224,7 +223,7 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) mutex_init(&client->ioctl_mutex); /* find free slot in the client table */ - spin_lock_irqsave(&clients_lock, flags); + spin_lock_irq(&clients_lock); if (client_index < 0) { for (c = SNDRV_SEQ_DYNAMIC_CLIENTS_BEGIN; c < SNDRV_SEQ_MAX_CLIENTS; @@ -232,17 +231,17 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) if (clienttab[c] || clienttablock[c]) continue; clienttab[client->number = c] = client; - spin_unlock_irqrestore(&clients_lock, flags); + spin_unlock_irq(&clients_lock); return client; } } else { if (clienttab[client_index] == NULL && !clienttablock[client_index]) { clienttab[client->number = client_index] = client; - spin_unlock_irqrestore(&clients_lock, flags); + spin_unlock_irq(&clients_lock); return client; } } - spin_unlock_irqrestore(&clients_lock, flags); + spin_unlock_irq(&clients_lock); snd_seq_pool_delete(&client->pool); kfree(client); return NULL; /* no free slot found or busy, return failure code */ @@ -251,23 +250,21 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) static int seq_free_client1(struct snd_seq_client *client) { - unsigned long flags; - if (!client) return 0; - spin_lock_irqsave(&clients_lock, flags); + spin_lock_irq(&clients_lock); clienttablock[client->number] = 1; clienttab[client->number] = NULL; - spin_unlock_irqrestore(&clients_lock, flags); + spin_unlock_irq(&clients_lock); snd_seq_delete_all_ports(client); snd_seq_queue_client_leave(client->number); snd_use_lock_sync(&client->use_lock); snd_seq_queue_client_termination(client->number); if (client->pool) snd_seq_pool_delete(&client->pool); - spin_lock_irqsave(&clients_lock, flags); + spin_lock_irq(&clients_lock); clienttablock[client->number] = 0; - spin_unlock_irqrestore(&clients_lock, flags); + spin_unlock_irq(&clients_lock); return 0; } diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 613ae10d33b8..97ee89cb6426 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -98,18 +98,17 @@ static struct snd_seq_event_cell *fifo_cell_out(struct snd_seq_fifo *f); void snd_seq_fifo_clear(struct snd_seq_fifo *f) { struct snd_seq_event_cell *cell; - unsigned long flags; /* clear overflow flag */ atomic_set(&f->overflow, 0); snd_use_lock_sync(&f->use_lock); - spin_lock_irqsave(&f->lock, flags); + spin_lock_irq(&f->lock); /* drain the fifo */ while ((cell = fifo_cell_out(f)) != NULL) { snd_seq_cell_free(cell); } - spin_unlock_irqrestore(&f->lock, flags); + spin_unlock_irq(&f->lock); } @@ -239,7 +238,6 @@ int snd_seq_fifo_poll_wait(struct snd_seq_fifo *f, struct file *file, /* change the size of pool; all old events are removed */ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) { - unsigned long flags; struct snd_seq_pool *newpool, *oldpool; struct snd_seq_event_cell *cell, *next, *oldhead; @@ -255,7 +253,7 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) return -ENOMEM; } - spin_lock_irqsave(&f->lock, flags); + spin_lock_irq(&f->lock); /* remember old pool */ oldpool = f->pool; oldhead = f->head; @@ -265,7 +263,7 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) f->tail = NULL; f->cells = 0; /* NOTE: overflow flag is not cleared */ - spin_unlock_irqrestore(&f->lock, flags); + spin_unlock_irq(&f->lock); /* close the old pool and wait until all users are gone */ snd_seq_pool_mark_closing(oldpool); diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index ae0b8971f6ce..19b718e871c5 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -384,7 +384,6 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) { int cell; struct snd_seq_event_cell *cellptr; - unsigned long flags; if (snd_BUG_ON(!pool)) return -EINVAL; @@ -395,9 +394,9 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) return -ENOMEM; /* add new cells to the free cell list */ - spin_lock_irqsave(&pool->lock, flags); + spin_lock_irq(&pool->lock); if (pool->ptr) { - spin_unlock_irqrestore(&pool->lock, flags); + spin_unlock_irq(&pool->lock); kvfree(cellptr); return 0; } @@ -416,7 +415,7 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) /* init statistics */ pool->max_used = 0; pool->total_elements = pool->size; - spin_unlock_irqrestore(&pool->lock, flags); + spin_unlock_irq(&pool->lock); return 0; } @@ -435,7 +434,6 @@ void snd_seq_pool_mark_closing(struct snd_seq_pool *pool) /* remove events */ int snd_seq_pool_done(struct snd_seq_pool *pool) { - unsigned long flags; struct snd_seq_event_cell *ptr; if (snd_BUG_ON(!pool)) @@ -449,18 +447,18 @@ int snd_seq_pool_done(struct snd_seq_pool *pool) schedule_timeout_uninterruptible(1); /* release all resources */ - spin_lock_irqsave(&pool->lock, flags); + spin_lock_irq(&pool->lock); ptr = pool->ptr; pool->ptr = NULL; pool->free = NULL; pool->total_elements = 0; - spin_unlock_irqrestore(&pool->lock, flags); + spin_unlock_irq(&pool->lock); kvfree(ptr); - spin_lock_irqsave(&pool->lock, flags); + spin_lock_irq(&pool->lock); pool->closing = 0; - spin_unlock_irqrestore(&pool->lock, flags); + spin_unlock_irq(&pool->lock); return 0; } diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 24d90abfc64d..1e2239240f21 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -128,7 +128,6 @@ static void port_subs_info_init(struct snd_seq_port_subs_info *grp) struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, int port) { - unsigned long flags; struct snd_seq_client_port *new_port, *p; int num = -1; @@ -157,7 +156,7 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, num = port >= 0 ? port : 0; mutex_lock(&client->ports_mutex); - write_lock_irqsave(&client->ports_lock, flags); + write_lock_irq(&client->ports_lock); list_for_each_entry(p, &client->ports_list_head, list) { if (p->addr.port > num) break; @@ -169,7 +168,7 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, client->num_ports++; new_port->addr.port = num; /* store the port number in the port */ sprintf(new_port->name, "port-%d", num); - write_unlock_irqrestore(&client->ports_lock, flags); + write_unlock_irq(&client->ports_lock); mutex_unlock(&client->ports_mutex); return new_port; @@ -283,11 +282,10 @@ static int port_delete(struct snd_seq_client *client, /* delete a port with the given port id */ int snd_seq_delete_port(struct snd_seq_client *client, int port) { - unsigned long flags; struct snd_seq_client_port *found = NULL, *p; mutex_lock(&client->ports_mutex); - write_lock_irqsave(&client->ports_lock, flags); + write_lock_irq(&client->ports_lock); list_for_each_entry(p, &client->ports_list_head, list) { if (p->addr.port == port) { /* ok found. delete from the list at first */ @@ -297,7 +295,7 @@ int snd_seq_delete_port(struct snd_seq_client *client, int port) break; } } - write_unlock_irqrestore(&client->ports_lock, flags); + write_unlock_irq(&client->ports_lock); mutex_unlock(&client->ports_mutex); if (found) return port_delete(client, found); @@ -308,7 +306,6 @@ int snd_seq_delete_port(struct snd_seq_client *client, int port) /* delete the all ports belonging to the given client */ int snd_seq_delete_all_ports(struct snd_seq_client *client) { - unsigned long flags; struct list_head deleted_list; struct snd_seq_client_port *port, *tmp; @@ -316,7 +313,7 @@ int snd_seq_delete_all_ports(struct snd_seq_client *client) * clear the port list in the client data. */ mutex_lock(&client->ports_mutex); - write_lock_irqsave(&client->ports_lock, flags); + write_lock_irq(&client->ports_lock); if (! list_empty(&client->ports_list_head)) { list_add(&deleted_list, &client->ports_list_head); list_del_init(&client->ports_list_head); @@ -324,7 +321,7 @@ int snd_seq_delete_all_ports(struct snd_seq_client *client) INIT_LIST_HEAD(&deleted_list); } client->num_ports = 0; - write_unlock_irqrestore(&client->ports_lock, flags); + write_unlock_irq(&client->ports_lock); /* remove each port in deleted_list */ list_for_each_entry_safe(port, tmp, &deleted_list, list) { -- cgit v1.2.3-59-g8ed1b From feb689025fbb6f0aa6297d3ddf97de945ea4ad32 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Apr 2019 17:35:22 +0200 Subject: ALSA: seq: Protect in-kernel ioctl calls with mutex ALSA OSS sequencer calls the ioctl function indirectly via snd_seq_kernel_client_ctl(). While we already applied the protection against races between the normal ioctls and writes via the client's ioctl_mutex, this code path was left untouched. And this seems to be the cause of still remaining some rare UAF as spontaneously triggered by syzkaller. For the sake of robustness, wrap the ioctl_mutex also for the call via snd_seq_kernel_client_ctl(), too. Reported-by: syzbot+e4c8abb920efa77bace9@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 933bde3843d9..976404691261 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2340,14 +2340,19 @@ int snd_seq_kernel_client_ctl(int clientid, unsigned int cmd, void *arg) { const struct ioctl_handler *handler; struct snd_seq_client *client; + int err; client = clientptr(clientid); if (client == NULL) return -ENXIO; for (handler = ioctl_handlers; handler->cmd > 0; ++handler) { - if (handler->cmd == cmd) - return handler->func(client, arg); + if (handler->cmd == cmd) { + mutex_lock(&client->ioctl_mutex); + err = handler->func(client, arg); + mutex_unlock(&client->ioctl_mutex); + return err; + } } pr_debug("ALSA: seq unknown ioctl() 0x%x (type='%c', number=0x%02x)\n", -- cgit v1.2.3-59-g8ed1b From 2eabc5ec8ab4d4748a82050dfcb994119b983750 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Apr 2019 18:04:17 +0200 Subject: ALSA: seq: Fix race of get-subscription call vs port-delete ioctls The snd_seq_ioctl_get_subscription() retrieves the port subscriber information as a pointer, while the object isn't protected, hence it may be deleted before the actual reference. This race was spotted by syzkaller and may lead to a UAF. The fix is simply copying the data in the lookup function that performs in the rwsem to protect against the deletion. Reported-by: syzbot+9437020c82413d00222d@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 10 ++-------- sound/core/seq/seq_ports.c | 13 ++++++++----- sound/core/seq/seq_ports.h | 5 +++-- 3 files changed, 13 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 976404691261..f256704dc401 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1897,20 +1897,14 @@ static int snd_seq_ioctl_get_subscription(struct snd_seq_client *client, int result; struct snd_seq_client *sender = NULL; struct snd_seq_client_port *sport = NULL; - struct snd_seq_subscribers *p; result = -EINVAL; if ((sender = snd_seq_client_use_ptr(subs->sender.client)) == NULL) goto __end; if ((sport = snd_seq_port_use_ptr(sender, subs->sender.port)) == NULL) goto __end; - p = snd_seq_port_get_subscription(&sport->c_src, &subs->dest); - if (p) { - result = 0; - *subs = p->info; - } else - result = -ENOENT; - + result = snd_seq_port_get_subscription(&sport->c_src, &subs->dest, + subs); __end: if (sport) snd_seq_port_unlock(sport); diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 1e2239240f21..d964d728681e 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -632,20 +632,23 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector, /* get matched subscriber */ -struct snd_seq_subscribers *snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp, - struct snd_seq_addr *dest_addr) +int snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp, + struct snd_seq_addr *dest_addr, + struct snd_seq_port_subscribe *subs) { - struct snd_seq_subscribers *s, *found = NULL; + struct snd_seq_subscribers *s; + int err = -ENOENT; down_read(&src_grp->list_mutex); list_for_each_entry(s, &src_grp->list_head, src_list) { if (addr_match(dest_addr, &s->info.dest)) { - found = s; + *subs = s->info; + err = 0; break; } } up_read(&src_grp->list_mutex); - return found; + return err; } /* diff --git a/sound/core/seq/seq_ports.h b/sound/core/seq/seq_ports.h index 26bd71f36c41..06003b36652e 100644 --- a/sound/core/seq/seq_ports.h +++ b/sound/core/seq/seq_ports.h @@ -135,7 +135,8 @@ int snd_seq_port_subscribe(struct snd_seq_client_port *port, struct snd_seq_port_subscribe *info); /* get matched subscriber */ -struct snd_seq_subscribers *snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp, - struct snd_seq_addr *dest_addr); +int snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp, + struct snd_seq_addr *dest_addr, + struct snd_seq_port_subscribe *subs); #endif -- cgit v1.2.3-59-g8ed1b From 41672c0c24a62699d20aab53b98d843b16483053 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Mar 2019 17:11:10 +0100 Subject: ALSA: timer: Simplify error path in snd_timer_open() Just a minor refactoring to use the standard goto for error paths in snd_timer_open() instead of open code. The first mutex_lock() is moved to the beginning of the function to make the code clearer. Signed-off-by: Takashi Iwai --- sound/core/timer.c | 39 ++++++++++++++++++++------------------- 1 file changed, 20 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index df52d2960179..0eed4fe0da21 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -255,19 +255,20 @@ int snd_timer_open(struct snd_timer_instance **ti, struct snd_timer_instance *timeri = NULL; int err; + mutex_lock(®ister_mutex); if (tid->dev_class == SNDRV_TIMER_CLASS_SLAVE) { /* open a slave instance */ if (tid->dev_sclass <= SNDRV_TIMER_SCLASS_NONE || tid->dev_sclass > SNDRV_TIMER_SCLASS_OSS_SEQUENCER) { pr_debug("ALSA: timer: invalid slave class %i\n", tid->dev_sclass); - return -EINVAL; + err = -EINVAL; + goto unlock; } - mutex_lock(®ister_mutex); timeri = snd_timer_instance_new(owner, NULL); if (!timeri) { - mutex_unlock(®ister_mutex); - return -ENOMEM; + err = -ENOMEM; + goto unlock; } timeri->slave_class = tid->dev_sclass; timeri->slave_id = tid->device; @@ -278,13 +279,10 @@ int snd_timer_open(struct snd_timer_instance **ti, snd_timer_close_locked(timeri); timeri = NULL; } - mutex_unlock(®ister_mutex); - *ti = timeri; - return err; + goto unlock; } /* open a master instance */ - mutex_lock(®ister_mutex); timer = snd_timer_find(tid); #ifdef CONFIG_MODULES if (!timer) { @@ -295,25 +293,26 @@ int snd_timer_open(struct snd_timer_instance **ti, } #endif if (!timer) { - mutex_unlock(®ister_mutex); - return -ENODEV; + err = -ENODEV; + goto unlock; } if (!list_empty(&timer->open_list_head)) { timeri = list_entry(timer->open_list_head.next, struct snd_timer_instance, open_list); if (timeri->flags & SNDRV_TIMER_IFLG_EXCLUSIVE) { - mutex_unlock(®ister_mutex); - return -EBUSY; + err = -EBUSY; + timeri = NULL; + goto unlock; } } if (timer->num_instances >= timer->max_instances) { - mutex_unlock(®ister_mutex); - return -EBUSY; + err = -EBUSY; + goto unlock; } timeri = snd_timer_instance_new(owner, timer); if (!timeri) { - mutex_unlock(®ister_mutex); - return -ENOMEM; + err = -ENOMEM; + goto unlock; } /* take a card refcount for safe disconnection */ if (timer->card) @@ -322,16 +321,16 @@ int snd_timer_open(struct snd_timer_instance **ti, timeri->slave_id = slave_id; if (list_empty(&timer->open_list_head) && timer->hw.open) { - int err = timer->hw.open(timer); + err = timer->hw.open(timer); if (err) { kfree(timeri->owner); kfree(timeri); + timeri = NULL; if (timer->card) put_device(&timer->card->card_dev); module_put(timer->module); - mutex_unlock(®ister_mutex); - return err; + goto unlock; } } @@ -342,6 +341,8 @@ int snd_timer_open(struct snd_timer_instance **ti, snd_timer_close_locked(timeri); timeri = NULL; } + + unlock: mutex_unlock(®ister_mutex); *ti = timeri; return err; -- cgit v1.2.3-59-g8ed1b From 5d704b0d3b4855734f029337e516b829c473801c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Mar 2019 17:44:53 +0100 Subject: ALSA: timer: Coding style fixes Avoid old school C style but do plain and clear way. Signed-off-by: Takashi Iwai --- sound/core/timer.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/timer.c b/sound/core/timer.c index 0eed4fe0da21..d23efec35660 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1909,7 +1909,10 @@ static int snd_timer_user_start(struct file *file) snd_timer_stop(tu->timeri); tu->timeri->lost = 0; tu->last_resolution = 0; - return (err = snd_timer_start(tu->timeri, tu->ticks)) < 0 ? err : 0; + err = snd_timer_start(tu->timeri, tu->ticks); + if (err < 0) + return err; + return 0; } static int snd_timer_user_stop(struct file *file) @@ -1920,7 +1923,10 @@ static int snd_timer_user_stop(struct file *file) tu = file->private_data; if (!tu->timeri) return -EBADFD; - return (err = snd_timer_stop(tu->timeri)) < 0 ? err : 0; + err = snd_timer_stop(tu->timeri); + if (err < 0) + return err; + return 0; } static int snd_timer_user_continue(struct file *file) @@ -1935,7 +1941,10 @@ static int snd_timer_user_continue(struct file *file) if (!(tu->timeri->flags & SNDRV_TIMER_IFLG_PAUSED)) return snd_timer_user_start(file); tu->timeri->lost = 0; - return (err = snd_timer_continue(tu->timeri)) < 0 ? err : 0; + err = snd_timer_continue(tu->timeri); + if (err < 0) + return err; + return 0; } static int snd_timer_user_pause(struct file *file) @@ -1946,7 +1955,10 @@ static int snd_timer_user_pause(struct file *file) tu = file->private_data; if (!tu->timeri) return -EBADFD; - return (err = snd_timer_pause(tu->timeri)) < 0 ? err : 0; + err = snd_timer_pause(tu->timeri); + if (err < 0) + return err; + return 0; } enum { -- cgit v1.2.3-59-g8ed1b From f0654ba94e33699b295ce4f3dc73094db6209035 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Apr 2019 19:58:32 +0200 Subject: Revert "ALSA: seq: Protect in-kernel ioctl calls with mutex" This reverts commit feb689025fbb6f0aa6297d3ddf97de945ea4ad32. The fix attempt was incorrect, leading to the mutex deadlock through the close of OSS sequencer client. The proper fix needs more consideration, so let's revert it now. Fixes: feb689025fbb ("ALSA: seq: Protect in-kernel ioctl calls with mutex") Reported-by: syzbot+47ded6c0f23016cde310@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index f256704dc401..de320b1b90de 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2334,19 +2334,14 @@ int snd_seq_kernel_client_ctl(int clientid, unsigned int cmd, void *arg) { const struct ioctl_handler *handler; struct snd_seq_client *client; - int err; client = clientptr(clientid); if (client == NULL) return -ENXIO; for (handler = ioctl_handlers; handler->cmd > 0; ++handler) { - if (handler->cmd == cmd) { - mutex_lock(&client->ioctl_mutex); - err = handler->func(client, arg); - mutex_unlock(&client->ioctl_mutex); - return err; - } + if (handler->cmd == cmd) + return handler->func(client, arg); } pr_debug("ALSA: seq unknown ioctl() 0x%x (type='%c', number=0x%02x)\n", -- cgit v1.2.3-59-g8ed1b From 7c32ae35fbf9cffb7aa3736f44dec10c944ca18e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Apr 2019 11:37:19 +0200 Subject: ALSA: seq: Cover unsubscribe_port() in list_mutex The call of unsubscribe_port() which manages the group count and module refcount from delete_and_unsubscribe_port() looks racy; it's not covered by the group list lock, and it's likely a cause of the reported unbalance at port deletion. Let's move the call inside the group list_mutex to plug the hole. Reported-by: syzbot+e4c8abb920efa77bace9@syzkaller.appspotmail.com Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ports.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index d964d728681e..ac7556ab531c 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -547,10 +547,10 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, list_del_init(list); grp->exclusive = 0; write_unlock_irq(&grp->list_lock); - up_write(&grp->list_mutex); if (!empty) unsubscribe_port(client, port, grp, &subs->info, ack); + up_write(&grp->list_mutex); } /* connect two ports */ -- cgit v1.2.3-59-g8ed1b From 6740ea6776e97274627e3c261469d1c4ba0a0cb5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Apr 2019 12:10:14 +0200 Subject: ALSA: seq: Simplify snd_seq_kernel_client_enqueue() helper We have two helpers for queuing a sequencer event from the kernel client, and both are used only from OSS sequencer layer without any hop and atomic set. Let's simplify and unify two helpers into one. No functional change, just a call pattern change. Signed-off-by: Takashi Iwai --- include/sound/seq_kernel.h | 3 ++- sound/core/seq/oss/seq_oss_rw.c | 11 ++++------- sound/core/seq/oss/seq_oss_writeq.c | 2 +- sound/core/seq/seq_clientmgr.c | 37 +++++++------------------------------ sound/core/seq/seq_clientmgr.h | 4 ---- 5 files changed, 14 insertions(+), 43 deletions(-) (limited to 'sound') diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h index 4b9ee3009aa0..c7a5433e109a 100644 --- a/include/sound/seq_kernel.h +++ b/include/sound/seq_kernel.h @@ -73,7 +73,8 @@ __printf(3, 4) int snd_seq_create_kernel_client(struct snd_card *card, int client_index, const char *name_fmt, ...); int snd_seq_delete_kernel_client(int client); -int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event *ev, int atomic, int hop); +int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event *ev, + struct file *file, bool blocking); int snd_seq_kernel_client_dispatch(int client, struct snd_seq_event *ev, int atomic, int hop); int snd_seq_kernel_client_ctl(int client, unsigned int cmd, void *arg); diff --git a/sound/core/seq/oss/seq_oss_rw.c b/sound/core/seq/oss/seq_oss_rw.c index 30886f5fb100..eb1ef12181f3 100644 --- a/sound/core/seq/oss/seq_oss_rw.c +++ b/sound/core/seq/oss/seq_oss_rw.c @@ -180,14 +180,11 @@ insert_queue(struct seq_oss_devinfo *dp, union evrec *rec, struct file *opt) return 0; /* invalid event - no need to insert queue */ event.time.tick = snd_seq_oss_timer_cur_tick(dp->timer); - if (dp->timer->realtime || !dp->timer->running) { + if (dp->timer->realtime || !dp->timer->running) snd_seq_oss_dispatch(dp, &event, 0, 0); - } else { - if (is_nonblock_mode(dp->file_mode)) - rc = snd_seq_kernel_client_enqueue(dp->cseq, &event, 0, 0); - else - rc = snd_seq_kernel_client_enqueue_blocking(dp->cseq, &event, opt, 0, 0); - } + else + rc = snd_seq_kernel_client_enqueue(dp->cseq, &event, opt, + !is_nonblock_mode(dp->file_mode)); return rc; } diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c index 5e04f4df10e4..b2f69617591f 100644 --- a/sound/core/seq/oss/seq_oss_writeq.c +++ b/sound/core/seq/oss/seq_oss_writeq.c @@ -116,7 +116,7 @@ snd_seq_oss_writeq_sync(struct seq_oss_writeq *q) rec->t.code = SEQ_SYNCTIMER; rec->t.time = time; q->sync_event_put = 1; - snd_seq_kernel_client_enqueue_blocking(dp->cseq, &ev, NULL, 0, 0); + snd_seq_kernel_client_enqueue(dp->cseq, &ev, NULL, true); } wait_event_interruptible_timeout(q->sync_sleep, ! q->sync_event_put, HZ); diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index de320b1b90de..0af5b1440b33 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2218,12 +2218,13 @@ int snd_seq_delete_kernel_client(int client) } EXPORT_SYMBOL(snd_seq_delete_kernel_client); -/* skeleton to enqueue event, called from snd_seq_kernel_client_enqueue - * and snd_seq_kernel_client_enqueue_blocking +/* + * exported, called by kernel clients to enqueue events (w/o blocking) + * + * RETURN VALUE: zero if succeed, negative if error */ -static int kernel_client_enqueue(int client, struct snd_seq_event *ev, - struct file *file, int blocking, - int atomic, int hop) +int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event *ev, + struct file *file, bool blocking) { struct snd_seq_client *cptr; int result; @@ -2250,37 +2251,13 @@ static int kernel_client_enqueue(int client, struct snd_seq_event *ev, result = -EPERM; else /* send it */ result = snd_seq_client_enqueue_event(cptr, ev, file, blocking, - atomic, hop, NULL); + false, 0, NULL); snd_seq_client_unlock(cptr); return result; } - -/* - * exported, called by kernel clients to enqueue events (w/o blocking) - * - * RETURN VALUE: zero if succeed, negative if error - */ -int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event * ev, - int atomic, int hop) -{ - return kernel_client_enqueue(client, ev, NULL, 0, atomic, hop); -} EXPORT_SYMBOL(snd_seq_kernel_client_enqueue); -/* - * exported, called by kernel clients to enqueue events (with blocking) - * - * RETURN VALUE: zero if succeed, negative if error - */ -int snd_seq_kernel_client_enqueue_blocking(int client, struct snd_seq_event * ev, - struct file *file, - int atomic, int hop) -{ - return kernel_client_enqueue(client, ev, file, 1, atomic, hop); -} -EXPORT_SYMBOL(snd_seq_kernel_client_enqueue_blocking); - /* * exported, called by kernel clients to dispatch events directly to other * clients, bypassing the queues. Event time-stamp will be updated. diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index 0611e1e0ed5b..66ad3d547916 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -93,10 +93,6 @@ struct snd_seq_client *snd_seq_client_use_ptr(int clientid); /* dispatch event to client(s) */ int snd_seq_dispatch_event(struct snd_seq_event_cell *cell, int atomic, int hop); -/* exported to other modules */ -int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event *ev, int atomic, int hop); -int snd_seq_kernel_client_enqueue_blocking(int client, struct snd_seq_event * ev, - struct file *file, int atomic, int hop); int snd_seq_kernel_client_write_poll(int clientid, struct file *file, poll_table *wait); int snd_seq_client_notify_subscription(int client, int port, struct snd_seq_port_subscribe *info, int evtype); -- cgit v1.2.3-59-g8ed1b From 6b580f523172f2c738b661069a57c23c74a75f88 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Apr 2019 12:44:39 +0200 Subject: ALSA: seq: Protect racy pool manipulation from OSS sequencer OSS sequencer emulation still allows to queue and issue the events that manipulate the client pool concurrently in a racy way. This patch serializes the access like the normal sequencer write / ioctl via taking the client ioctl_mutex. Since the access to the sequencer client is done indirectly via a client id number, a new helper to take/release the mutex is introduced. Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_device.h | 10 ++++++++-- sound/core/seq/seq_clientmgr.c | 40 ++++++++++++++++++++++++++++++++++--- sound/core/seq/seq_clientmgr.h | 4 ++++ 3 files changed, 49 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h index 2d0e9eaf13aa..77eb1fe1155c 100644 --- a/sound/core/seq/oss/seq_oss_device.h +++ b/sound/core/seq/oss/seq_oss_device.h @@ -30,6 +30,7 @@ #include #include #include +#include "../seq_clientmgr.h" /* max. applications */ #define SNDRV_SEQ_OSS_MAX_CLIENTS 16 @@ -150,11 +151,16 @@ snd_seq_oss_dispatch(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, int a return snd_seq_kernel_client_dispatch(dp->cseq, ev, atomic, hop); } -/* ioctl */ +/* ioctl for writeq */ static inline int snd_seq_oss_control(struct seq_oss_devinfo *dp, unsigned int type, void *arg) { - return snd_seq_kernel_client_ctl(dp->cseq, type, arg); + int err; + + snd_seq_client_ioctl_lock(dp->cseq); + err = snd_seq_kernel_client_ctl(dp->cseq, type, arg); + snd_seq_client_ioctl_unlock(dp->cseq); + return err; } /* fill the addresses in header */ diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 0af5b1440b33..a5c9d59eb5b8 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -179,6 +179,36 @@ struct snd_seq_client *snd_seq_client_use_ptr(int clientid) return client; } +/* Take refcount and perform ioctl_mutex lock on the given client; + * used only for OSS sequencer + * Unlock via snd_seq_client_ioctl_unlock() below + */ +bool snd_seq_client_ioctl_lock(int clientid) +{ + struct snd_seq_client *client; + + client = snd_seq_client_use_ptr(clientid); + if (!client) + return false; + mutex_lock(&client->ioctl_mutex); + return true; +} +EXPORT_SYMBOL_GPL(snd_seq_client_ioctl_lock); + +/* Unlock and unref the given client; for OSS sequencer use only */ +void snd_seq_client_ioctl_unlock(int clientid) +{ + struct snd_seq_client *client; + + client = snd_seq_client_use_ptr(clientid); + if (WARN_ON(!client)) + return; + mutex_unlock(&client->ioctl_mutex); + snd_use_lock_free(&client->use_lock); + snd_seq_client_unlock(client); +} +EXPORT_SYMBOL_GPL(snd_seq_client_ioctl_unlock); + static void usage_alloc(struct snd_seq_usage *res, int num) { res->cur += num; @@ -2247,11 +2277,15 @@ int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event *ev, if (cptr == NULL) return -EINVAL; - if (! cptr->accept_output) + if (!cptr->accept_output) { result = -EPERM; - else /* send it */ + } else { /* send it */ + mutex_lock(&cptr->ioctl_mutex); result = snd_seq_client_enqueue_event(cptr, ev, file, blocking, - false, 0, NULL); + false, 0, + &cptr->ioctl_mutex); + mutex_unlock(&cptr->ioctl_mutex); + } snd_seq_client_unlock(cptr); return result; diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index 66ad3d547916..28a51dcc0190 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -97,6 +97,10 @@ int snd_seq_kernel_client_write_poll(int clientid, struct file *file, poll_table int snd_seq_client_notify_subscription(int client, int port, struct snd_seq_port_subscribe *info, int evtype); +/* only for OSS sequencer */ +bool snd_seq_client_ioctl_lock(int clientid); +void snd_seq_client_ioctl_unlock(int clientid); + extern int seq_client_load[15]; #endif -- cgit v1.2.3-59-g8ed1b From e61ab9f017493ef42f0374a578b2758c30f41f74 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 10 Apr 2019 16:00:54 +0200 Subject: ALSA: hda: Initialize ext-bus-specific fields in snd_hdac_bus_init(), too Some fields in snd_hdac_bus are ext-bus specific, but they still should be initialized in snd_hdac_bus_init() for consistency, at least, for the ones that do need the explicit initialization like the list head. Also move the lock field to the more appropriate place and correct the comment to reflect the recent change where it serves for both the display power and the link management. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 7 +++---- sound/hda/ext/hdac_ext_bus.c | 2 -- sound/hda/hdac_bus.c | 1 + 3 files changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 45f944d57982..ca1d9a70d026 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -297,7 +297,7 @@ struct hdac_rb { * @num_streams: streams supported * @idx: HDA link index * @hlink_list: link list of HDA links - * @lock: lock for link mgmt + * @lock: lock for link and display power mgmt * @cmd_dma_state: state of cmd DMAs: CORB and RIRB */ struct hdac_bus { @@ -363,6 +363,7 @@ struct hdac_bus { /* locks */ spinlock_t reg_lock; struct mutex cmd_mutex; + struct mutex lock; /* DRM component interface */ struct drm_audio_component *audio_component; @@ -373,11 +374,9 @@ struct hdac_bus { int num_streams; int idx; + /* link management */ struct list_head hlink_list; - - struct mutex lock; bool cmd_dma_state; - }; int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index ec7715c6b0c0..c203af71a099 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -104,9 +104,7 @@ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, return ret; bus->ext_ops = ext_ops; - INIT_LIST_HEAD(&bus->hlink_list); bus->idx = idx++; - bus->cmd_dma_state = true; return 0; diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index ad8eee08013f..10e5d261fde1 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -39,6 +39,7 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, spin_lock_init(&bus->reg_lock); mutex_init(&bus->cmd_mutex); mutex_init(&bus->lock); + INIT_LIST_HEAD(&bus->hlink_list); bus->irq = -1; return 0; } -- cgit v1.2.3-59-g8ed1b From 46f5710f0b8829882faea735149d86250db5d0e5 Mon Sep 17 00:00:00 2001 From: Roope Salmi Date: Sun, 14 Apr 2019 14:13:06 +0300 Subject: ALSA: usb-audio: Add quirk for Focusrite Scarlett Solo The device reports Synch: Synchronous on the playback interface. This causes regular audible napping on sample rates that are not multiples of 1 kHz. Fix to Synch: Asynchronous. Specifically observed on Focusrite Scarlett Solo 2nd generation. I assume the first generation model has a different device ID. A first generation Scarlett 2i2 I was able to test advertised Synch: Asynchronous by default. For example, with a sample rate of 44100 Hz, a silent sample is played every 40.96 seconds (likely 44.0 samples instead of 44.1 transmitted per USB frame on average, 4096 being the size of some internal buffer). There may be some other bug at play here since this doesn't happen on other platforms. However, a feedback endpoint is listed and using it fixes the issue. That is the only change in the quirk, but I didn't find a way to declare only it. Tested on two units and on two different computers. Signed-off-by: Roope Salmi Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 84 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 84 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 86e80916a029..629b84532648 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2770,6 +2770,90 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + /* + * Focusrite Scarlett Solo 2nd generation + * Reports that playback should use Synch: Synchronous + * while still providing a feedback endpoint. Synchronous causes + * snapping on some sample rates. + * Force it to use Synch: Asynchronous. + */ + USB_DEVICE(0x1235, 0x8205), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .protocol = UAC_VERSION_2, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 44100, + .rate_max = 192000, + .nr_rates = 6, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, + 96000, 176400, 192000 + }, + .clock = 41 + } + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC | + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .protocol = UAC_VERSION_2, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 44100, + .rate_max = 192000, + .nr_rates = 6, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, + 96000, 176400, 192000 + }, + .clock = 41 + } + }, + { + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { -- cgit v1.2.3-59-g8ed1b From b5fd12d6c0fc64c2c2b5ae095e63824083d27151 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Apr 2019 09:03:01 +0200 Subject: ALSA: seq: Correct unlock sequence at snd_seq_client_ioctl_unlock() The doubly unlock sequence at snd_seq_client_ioctl_unlock() is tricky. I took a direct unref call since I thought it would avoid misunderstanding, but rather this seems more confusing. Let's use snd_seq_client_unlock() consistently even if they look strange to be called twice, and add more comments for avoiding reader's confusion. Fixes: 6b580f523172 ("ALSA: seq: Protect racy pool manipulation from OSS sequencer") Reviewed-by: Kai Vehmanen Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 3acd80e718f2..c0227a672442 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -191,6 +191,7 @@ bool snd_seq_client_ioctl_lock(int clientid) if (!client) return false; mutex_lock(&client->ioctl_mutex); + /* The client isn't unrefed here; see snd_seq_client_ioctl_unlock() */ return true; } EXPORT_SYMBOL_GPL(snd_seq_client_ioctl_lock); @@ -204,7 +205,11 @@ void snd_seq_client_ioctl_unlock(int clientid) if (WARN_ON(!client)) return; mutex_unlock(&client->ioctl_mutex); - snd_use_lock_free(&client->use_lock); + /* The doubly unrefs below are intentional; the first one releases the + * leftover from snd_seq_client_ioctl_lock() above, and the second one + * is for releasing snd_seq_client_use_ptr() in this function + */ + snd_seq_client_unlock(client); snd_seq_client_unlock(client); } EXPORT_SYMBOL_GPL(snd_seq_client_ioctl_unlock); -- cgit v1.2.3-59-g8ed1b From c989954938761a2939a21fcbc768af182de6be58 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Apr 2019 18:01:46 +0200 Subject: ALSA: emu10k1: Drop superfluous id-uniquification behavior The emu10k1 driver tries to create a unique id string by itself when it's copied from the card list, but it's rather superfluous, as the same thing will be done in ALSA core side at the card registration. Let's drop the code. This allows us removing snd_cards export. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 16 +--------------- 1 file changed, 1 insertion(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 61f85ff91cd9..0419c75bdf5a 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1882,22 +1882,8 @@ int snd_emu10k1_create(struct snd_card *card, c->name, pci->vendor, pci->device, emu->serial); - if (!*card->id && c->id) { - int i, n = 0; + if (!*card->id && c->id) strlcpy(card->id, c->id, sizeof(card->id)); - for (;;) { - for (i = 0; i < snd_ecards_limit; i++) { - if (snd_cards[i] && !strcmp(snd_cards[i]->id, card->id)) - break; - } - if (i >= snd_ecards_limit) - break; - n++; - if (n >= SNDRV_CARDS) - break; - snprintf(card->id, sizeof(card->id), "%s_%d", c->id, n); - } - } is_audigy = emu->audigy = c->emu10k2_chip; -- cgit v1.2.3-59-g8ed1b From f4fa968950aef7efb98a0899bb0132405e2edaf1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Apr 2019 18:18:47 +0200 Subject: ALSA: core: Don't refer to snd_cards array directly The snd_cards[] array holds the card pointers that have been currently registered, and it's exported for the external modules that may need to refer a card object. But accessing to this array can be racy against the driver probe or removal, as the card registration or free may happen concurrently. This patch gets rid of the direct access to snd_cards[] array and provides a helper function to give the card object from the index number with a refcount management. Then the caller can access to the given card object safely, and releases it via snd_card_unref(). While we're at it, add a proper comment to snd_card_unref() and make it an inlined function for type-safety, too. Signed-off-by: Takashi Iwai --- include/sound/core.h | 16 ++++++++++++++-- sound/core/init.c | 23 +++++++++++++++++++++-- sound/core/oss/mixer_oss.c | 16 ++++++++++++---- sound/core/sound.c | 5 ++++- sound/last.c | 10 +++++++--- 5 files changed, 58 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/include/sound/core.h b/include/sound/core.h index e923c23e05dd..c90ebbc8d9c4 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -226,7 +226,6 @@ int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size /* init.c */ -extern struct snd_card *snd_cards[SNDRV_CARDS]; int snd_card_locked(int card); #if IS_ENABLED(CONFIG_SND_MIXER_OSS) #define SND_MIXER_OSS_NOTIFY_REGISTER 0 @@ -251,7 +250,20 @@ int snd_card_add_dev_attr(struct snd_card *card, int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); -#define snd_card_unref(card) put_device(&(card)->card_dev) + +struct snd_card *snd_card_ref(int card); + +/** + * snd_card_unref - Unreference the card object + * @card: the card object to unreference + * + * Call this function for the card object that was obtained via snd_card_ref() + * or snd_lookup_minor_data(). + */ +static inline void snd_card_unref(struct snd_card *card) +{ + put_device(&card->card_dev); +} #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) diff --git a/sound/core/init.c b/sound/core/init.c index 0c4dc40376a7..356d41e4f3a4 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -49,8 +49,7 @@ static const struct file_operations snd_shutdown_f_ops; /* locked for registering/using */ static DECLARE_BITMAP(snd_cards_lock, SNDRV_CARDS); -struct snd_card *snd_cards[SNDRV_CARDS]; -EXPORT_SYMBOL(snd_cards); +static struct snd_card *snd_cards[SNDRV_CARDS]; static DEFINE_MUTEX(snd_card_mutex); @@ -268,6 +267,26 @@ int snd_card_new(struct device *parent, int idx, const char *xid, } EXPORT_SYMBOL(snd_card_new); +/** + * snd_card_ref - Get the card object from the index + * @idx: the card index + * + * Returns a card object corresponding to the given index or NULL if not found. + * Release the object via snd_card_unref(). + */ +struct snd_card *snd_card_ref(int idx) +{ + struct snd_card *card; + + mutex_lock(&snd_card_mutex); + card = snd_cards[idx]; + if (card) + get_device(&card->card_dev); + mutex_unlock(&snd_card_mutex); + return card; +} +EXPORT_SYMBOL_GPL(snd_card_ref); + /* return non-zero if a card is already locked */ int snd_card_locked(int card) { diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 64d904bee8bb..c8618678649c 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1403,24 +1403,32 @@ static int snd_mixer_oss_notify_handler(struct snd_card *card, int cmd) static int __init alsa_mixer_oss_init(void) { + struct snd_card *card; int idx; snd_mixer_oss_notify_callback = snd_mixer_oss_notify_handler; for (idx = 0; idx < SNDRV_CARDS; idx++) { - if (snd_cards[idx]) - snd_mixer_oss_notify_handler(snd_cards[idx], SND_MIXER_OSS_NOTIFY_REGISTER); + card = snd_card_ref(idx); + if (card) { + snd_mixer_oss_notify_handler(card, SND_MIXER_OSS_NOTIFY_REGISTER); + snd_card_unref(card); + } } return 0; } static void __exit alsa_mixer_oss_exit(void) { + struct snd_card *card; int idx; snd_mixer_oss_notify_callback = NULL; for (idx = 0; idx < SNDRV_CARDS; idx++) { - if (snd_cards[idx]) - snd_mixer_oss_notify_handler(snd_cards[idx], SND_MIXER_OSS_NOTIFY_FREE); + card = snd_card_ref(idx); + if (card) { + snd_mixer_oss_notify_handler(card, SND_MIXER_OSS_NOTIFY_FREE); + snd_card_unref(card); + } } } diff --git a/sound/core/sound.c b/sound/core/sound.c index b30f027eb0fe..a9ad4379523b 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -134,8 +134,11 @@ static struct snd_minor *autoload_device(unsigned int minor) if (dev == SNDRV_MINOR_CONTROL) { /* /dev/aloadC? */ int card = SNDRV_MINOR_CARD(minor); - if (snd_cards[card] == NULL) + struct snd_card *ref = snd_card_ref(card); + if (!ref) snd_request_card(card); + else + snd_card_unref(ref); } else if (dev == SNDRV_MINOR_GLOBAL) { /* /dev/aloadSEQ */ snd_request_other(minor); diff --git a/sound/last.c b/sound/last.c index 43f222825038..4f5a624ab438 100644 --- a/sound/last.c +++ b/sound/last.c @@ -24,14 +24,18 @@ static int __init alsa_sound_last_init(void) { + struct snd_card *card; int idx, ok = 0; printk(KERN_INFO "ALSA device list:\n"); - for (idx = 0; idx < SNDRV_CARDS; idx++) - if (snd_cards[idx] != NULL) { - printk(KERN_INFO " #%i: %s\n", idx, snd_cards[idx]->longname); + for (idx = 0; idx < SNDRV_CARDS; idx++) { + card = snd_card_ref(idx); + if (card) { + printk(KERN_INFO " #%i: %s\n", idx, card->longname); + snd_card_unref(card); ok++; } + } if (ok == 0) printk(KERN_INFO " No soundcards found.\n"); return 0; -- cgit v1.2.3-59-g8ed1b From 442e321ed223d9c01a444c6f874e32fb283c17d2 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Wed, 17 Apr 2019 22:57:22 +0800 Subject: ALSA: ps3: Remove set but not used variables 'start_vaddr' and 'pcm_index' Fixes gcc '-Wunused-but-set-variable' warnings: sound/ppc/snd_ps3.c: In function 'snd_ps3_program_dma': sound/ppc/snd_ps3.c:236:8: warning: variable 'start_vaddr' set but not used [-Wunused-but-set-variable] sound/ppc/snd_ps3.c: In function 'snd_ps3_pcm_open': sound/ppc/snd_ps3.c:529:6: warning: variable 'pcm_index' set but not used [-Wunused-but-set-variable] They are never used and can be removed. Reported-by: Hulk Robot Signed-off-by: YueHaibing Acked-by: Geoff Levand Signed-off-by: Takashi Iwai --- sound/ppc/snd_ps3.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 521236efcc4d..f77a0d5c0385 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -233,7 +233,6 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card, int fill_stages, dma_ch, stage; enum snd_ps3_ch ch; uint32_t ch0_kick_event = 0; /* initialize to mute gcc */ - void *start_vaddr; unsigned long irqsave; int silent = 0; @@ -257,7 +256,6 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card, fill_stages = 4; spin_lock_irqsave(&card->dma_lock, irqsave); for (ch = 0; ch < 2; ch++) { - start_vaddr = card->dma_next_transfer_vaddr[0]; for (stage = 0; stage < fill_stages; stage++) { dma_ch = stage * 2 + ch; if (silent) @@ -526,9 +524,7 @@ static int snd_ps3_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); - int pcm_index; - pcm_index = substream->pcm->device; /* to retrieve substream/runtime in interrupt handler */ card->substream = substream; -- cgit v1.2.3-59-g8ed1b From 4c61608402f5408855d54c91e32e61780644dd1d Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:36 +0200 Subject: ASoC: samsung: arndale: Convert to SPDX License Identifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/arndale_rt5631.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index ee1fda92f2f4..cc334e1866f6 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -1,15 +1,8 @@ -/* - * arndale_rt5631.c - * - * Copyright (c) 2014, Insignal Co., Ltd. - * - * Author: Claude - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright (c) 2014, Insignal Co., Ltd. +// +// Author: Claude #include #include -- cgit v1.2.3-59-g8ed1b From 41ce2ac0417c207b4b908108de1f0c72661cc558 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:37 +0200 Subject: ASoC: samsung: bells: Convert to SPDX License Identifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/bells.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 0e66cd8ef2f9..770845e2507a 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -1,13 +1,8 @@ -/* - * Bells audio support - * - * Copyright 2012 Wolfson Microelectronics - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Bells audio support +// +// Copyright 2012 Wolfson Microelectronics #include #include -- cgit v1.2.3-59-g8ed1b From df504eda6a7242d3ed5e5d2c9fbc8c718ca140c4 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:38 +0200 Subject: ASoC: samsung: dmaengine: Convert to SPDX License Indentifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/dma.h | 8 ++------ sound/soc/samsung/dmaengine.c | 22 ++++++---------------- 2 files changed, 8 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0ae15d01a3f6..7b5d4556e0fd 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -1,10 +1,6 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ /* - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * ALSA PCM interface for the Samsung SoC + * ALSA PCM interface for the Samsung SoC */ #ifndef _SAMSUNG_DMA_H diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index 302871974cb3..2802789a323e 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -1,19 +1,9 @@ -/* - * dmaengine.c - Samsung dmaengine wrapper - * - * Author: Mark Brown - * Copyright 2013 Linaro - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - */ +// SPDX-License-Identifier: GPL-2.0 +// +// dmaengine.c - Samsung dmaengine wrapper +// +// Author: Mark Brown +// Copyright 2013 Linaro #include #include -- cgit v1.2.3-59-g8ed1b From 392d20f60d000dd36ee69c993d13e5586ca54e7f Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:39 +0200 Subject: ASoC: samsung: h1940_uda1380: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/h1940_uda1380.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 051935162d7b..95925c4a5964 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -1,17 +1,11 @@ -/* - * h1940-uda1380.c -- ALSA Soc Audio Layer - * - * Copyright (c) 2010 Arnaud Patard - * Copyright (c) 2010 Vasily Khoruzhick - * - * Based on version from Arnaud Patard - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// h1940_uda1380.c - ALSA SoC Audio Layer +// +// Copyright (c) 2010 Arnaud Patard +// Copyright (c) 2010 Vasily Khoruzhick +// +// Based on version from Arnaud Patard #include #include -- cgit v1.2.3-59-g8ed1b From 7d5b1b8aaa659ff2a26e5a9910f2b6132ea78e45 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:40 +0200 Subject: ASoC: samsung: i2s: Convert to SPDX License Identifier Replace remaining GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/i2s-regs.h | 8 +------- sound/soc/samsung/i2s.h | 8 ++------ 2 files changed, 3 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h index 964985ea2e80..b4b5d6053503 100644 --- a/sound/soc/samsung/i2s-regs.h +++ b/sound/soc/samsung/i2s-regs.h @@ -1,15 +1,9 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ /* - * linux/sound/soc/samsung/i2s-regs.h - * * Copyright (c) 2011 Samsung Electronics Co., Ltd. * http://www.samsung.com * * Samsung I2S driver's register header - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. */ #ifndef __SND_SOC_SAMSUNG_I2S_REGS_H diff --git a/sound/soc/samsung/i2s.h b/sound/soc/samsung/i2s.h index a9832a9555cb..78b475ef98d9 100644 --- a/sound/soc/samsung/i2s.h +++ b/sound/soc/samsung/i2s.h @@ -1,13 +1,9 @@ -/* sound/soc/samsung/i2s.h - * +/* SPDX-License-Identifier: GPL-2.0 */ +/* * ALSA SoC Audio Layer - Samsung I2S Controller driver * * Copyright (c) 2010 Samsung Electronics Co. Ltd. * Jaswinder Singh - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SND_SOC_SAMSUNG_I2S_H -- cgit v1.2.3-59-g8ed1b From 1d3279c4ab5738631fa6f775991fcc65803dccff Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:41 +0200 Subject: ASoC: samsung: idma: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/idma.c | 20 +++++++------------- sound/soc/samsung/idma.h | 9 +-------- 2 files changed, 8 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index b1f09b942410..65497cd477a5 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -1,16 +1,10 @@ -/* - * sound/soc/samsung/idma.c - * - * Copyright (c) 2011 Samsung Electronics Co., Ltd. - * http://www.samsung.com - * - * I2S0's Internal DMA driver - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// idma.c - I2S0 internal DMA driver +// +// Copyright (c) 2011 Samsung Electronics Co., Ltd. +// http://www.samsung.com + #include #include #include diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h index 8644946973e5..8a46a918ed2a 100644 --- a/sound/soc/samsung/idma.h +++ b/sound/soc/samsung/idma.h @@ -1,14 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* - * sound/soc/samsung/idma.h - * * Copyright (c) 2011 Samsung Electronics Co., Ltd * http://www.samsung.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * */ #ifndef __SND_SOC_SAMSUNG_IDMA_H_ -- cgit v1.2.3-59-g8ed1b From 9c8df4cd99643759b69e1b8dca7f2065ccf63ed2 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:42 +0200 Subject: ASoC: samsung: jive_wm8750: Convert to SPDX License Identifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/jive_wm8750.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 529b10dc532b..f05f9e03f07d 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -1,15 +1,10 @@ -/* sound/soc/samsung/jive_wm8750.c - * - * Copyright 2007,2008 Simtec Electronics - * - * Based on sound/soc/pxa/spitz.c - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright 2007,2008 Simtec Electronics +// +// Based on sound/soc/pxa/spitz.c +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. #include #include -- cgit v1.2.3-59-g8ed1b From 1643b9fb8c77e1cf31918beea3856754e1d5f834 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:43 +0200 Subject: ASoC: samsung: littlemill: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/littlemill.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 087f8d738dfb..cd70b06cc99d 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -1,13 +1,8 @@ -/* - * Littlemill audio support - * - * Copyright 2011 Wolfson Microelectronics - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Littlemill audio support +// +// Copyright 2011 Wolfson Microelectronics #include #include -- cgit v1.2.3-59-g8ed1b From d5943db34af3825b73690724f10fe2bc95cc8dc2 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:44 +0200 Subject: ASoC: samsung: lowland: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/lowland.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index c9081f42f373..2fdab2ac8e8c 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -1,13 +1,8 @@ -/* - * Lowland audio support - * - * Copyright 2011 Wolfson Microelectronics - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Lowland audio support +// +// Copyright 2011 Wolfson Microelectronics #include #include -- cgit v1.2.3-59-g8ed1b From 7cab920a891ee635c26d85b04d10c04591333047 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:45 +0200 Subject: ASoC: samsung: neo1973_wm8753: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/neo1973_wm8753.c | 25 ++++++++++--------------- 1 file changed, 10 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 65602b935377..7e625066ddcd 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -1,18 +1,13 @@ -/* - * neo1973_wm8753.c -- SoC audio for Openmoko Neo1973 and Freerunner devices - * - * Copyright 2007 Openmoko Inc - * Author: Graeme Gregory - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * Copyright 2009 Wolfson Microelectronics - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// neo1973_wm8753.c - SoC audio for Openmoko Neo1973 and Freerunner devices +// +// Copyright 2007 Openmoko Inc +// Author: Graeme Gregory +// Copyright 2007 Wolfson Microelectronics PLC. +// Author: Graeme Gregory +// graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com +// Copyright 2009 Wolfson Microelectronics #include #include -- cgit v1.2.3-59-g8ed1b From 1896b485d3ec849585d60db2dc861a744ba73f1a Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:46 +0200 Subject: ASoC: samsung: odroid: Convert to SPDX License Identifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/odroid.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 1dc54c4206f0..e688169ff12a 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -1,10 +1,6 @@ -/* - * Copyright (C) 2017 Samsung Electronics Co., Ltd. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (C) 2017 Samsung Electronics Co., Ltd. #include #include -- cgit v1.2.3-59-g8ed1b From 65511dc7488eee1d5597e427726297ea49a59998 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:47 +0200 Subject: ASoC: samsung: pcm: Convert to SPDX License Identifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/pcm.c | 19 +++++++------------ sound/soc/samsung/pcm.h | 8 +------- 2 files changed, 8 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 3c7baa561084..f6e67d0e7882 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -1,15 +1,10 @@ -/* sound/soc/samsung/pcm.c - * - * ALSA SoC Audio Layer - S3C PCM-Controller driver - * - * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh - * based upon I2S drivers by Ben Dooks. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Audio Layer - S3C PCM-Controller driver +// +// Copyright (c) 2009 Samsung Electronics Co. Ltd +// Author: Jaswinder Singh +// based upon I2S drivers by Ben Dooks. #include #include diff --git a/sound/soc/samsung/pcm.h b/sound/soc/samsung/pcm.h index 726baf814613..208d8da27de1 100644 --- a/sound/soc/samsung/pcm.h +++ b/sound/soc/samsung/pcm.h @@ -1,10 +1,4 @@ -/* sound/soc/samsung/pcm.h - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - */ +/* SPDX-License-Identifier: GPL-2.0 */ #ifndef __S3C_PCM_H #define __S3C_PCM_H __FILE__ -- cgit v1.2.3-59-g8ed1b From 08ac76d01beaf32e8ea4e20c20ccea42206ddc90 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:48 +0200 Subject: ASoC: samsung: s3c-i2s-v2: Convert to SPDX License Identifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/regs-i2s-v2.h | 10 +++------- sound/soc/samsung/s3c-i2s-v2.c | 26 +++++++++++--------------- sound/soc/samsung/s3c-i2s-v2.h | 11 +++-------- 3 files changed, 17 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/regs-i2s-v2.h b/sound/soc/samsung/regs-i2s-v2.h index 5e5e5680580b..867984e75709 100644 --- a/sound/soc/samsung/regs-i2s-v2.h +++ b/sound/soc/samsung/regs-i2s-v2.h @@ -1,14 +1,10 @@ -/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h - * +/* SPDX-License-Identifier: GPL-2.0 */ +/* * Copyright 2007 Simtec Electronics * http://armlinux.simtec.co.uk/ * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * * S3C2412 IIS register definition -*/ + */ #ifndef __ASM_ARCH_REGS_S3C2412_IIS_H #define __ASM_ARCH_REGS_S3C2412_IIS_H diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 58c3e9bfc6b7..7e196b599be1 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -1,18 +1,14 @@ -/* ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. - * - * Copyright (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory@wolfsonmicro.com - * linux@wolfsonmicro.com - * - * Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. +// +// Copyright (c) 2006 Wolfson Microelectronics PLC. +// Graeme Gregory graeme.gregory@wolfsonmicro.com +// linux@wolfsonmicro.com +// +// Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics +// http://armlinux.simtec.co.uk/ +// Ben Dooks #include #include diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h index 3fca20f7a853..fe42b77999fd 100644 --- a/sound/soc/samsung/s3c-i2s-v2.h +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -1,16 +1,11 @@ -/* sound/soc/samsung/s3c-i2s-v2.h - * +/* SPDX-License-Identifier: GPL-2.0+ */ +/* * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver * * Copyright (c) 2007 Simtec Electronics * http://armlinux.simtec.co.uk/ * Ben Dooks - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. -*/ + */ /* This code is the core support for the I2S block found in a number of * Samsung SoC devices which is unofficially named I2S-V2. Currently the -- cgit v1.2.3-59-g8ed1b From 521cc68ea5888a6439a43a35bf84bbf7b5b79363 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:49 +0200 Subject: ASoC: samsung: s3c24xx-i2s: Convert to SPDX License Identifier Replace GPL v2.0(+) license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/regs-iis.h | 7 ++----- sound/soc/samsung/s3c24xx-i2s.c | 25 ++++++++++--------------- sound/soc/samsung/s3c24xx-i2s.h | 6 +----- 3 files changed, 13 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/regs-iis.h b/sound/soc/samsung/regs-iis.h index dc6cbbe9c4f0..253e172ad3b6 100644 --- a/sound/soc/samsung/regs-iis.h +++ b/sound/soc/samsung/regs-iis.h @@ -1,13 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * Copyright (c) 2003 Simtec Electronics * http://www.simtec.co.uk/products/SWLINUX/ * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * * S3C2410 IIS register definition -*/ + */ #ifndef __SAMSUNG_REGS_IIS_H__ #define __SAMSUNG_REGS_IIS_H__ diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index a8026b640c95..92bdaf0878f8 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -1,18 +1,13 @@ -/* - * s3c24xx-i2s.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// s3c24xx-i2s.c -- ALSA Soc Audio Layer +// +// (c) 2006 Wolfson Microelectronics PLC. +// Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com +// +// Copyright 2004-2005 Simtec Electronics +// http://armlinux.simtec.co.uk/ +// Ben Dooks #include #include diff --git a/sound/soc/samsung/s3c24xx-i2s.h b/sound/soc/samsung/s3c24xx-i2s.h index f9ca04edacb7..e073e31855d0 100644 --- a/sound/soc/samsung/s3c24xx-i2s.h +++ b/sound/soc/samsung/s3c24xx-i2s.h @@ -1,3 +1,4 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ /* * s3c24xx-i2s.c -- ALSA Soc Audio Layer * @@ -5,11 +6,6 @@ * Author: Graeme Gregory * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * * Revision history * 10th Nov 2006 Initial version. */ -- cgit v1.2.3-59-g8ed1b From 2dab7021aa52fb9b31c2ecd93d9b0399808c740b Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:50 +0200 Subject: ASoC: samsung: rx1950_uda1380: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/rx1950_uda1380.c | 30 ++++++++++++------------------ 1 file changed, 12 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index a064ca7d78c3..1dcc1b252ad1 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -1,21 +1,15 @@ -/* - * rx1950.c -- ALSA Soc Audio Layer - * - * Copyright (c) 2010 Vasily Khoruzhick - * - * Based on smdk2440.c and magician.c - * - * Authors: Graeme Gregory graeme.gregory@wolfsonmicro.com - * Philipp Zabel - * Denis Grigoriev - * Vasily Khoruzhick - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// rx1950.c - ALSA SoC Audio Layer +// +// Copyright (c) 2010 Vasily Khoruzhick +// +// Based on smdk2440.c and magician.c +// +// Authors: Graeme Gregory graeme.gregory@wolfsonmicro.com +// Philipp Zabel +// Denis Grigoriev +// Vasily Khoruzhick #include #include -- cgit v1.2.3-59-g8ed1b From 25bc05fbc464457a2187cb240b26a6dbd609399a Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:51 +0200 Subject: ASoC: samsung: s3c2412-i2s: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/s3c2412-i2s.c | 28 +++++++++++----------------- sound/soc/samsung/s3c2412-i2s.h | 11 +++-------- 2 files changed, 14 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index c08638b0e458..787a3f6e9f24 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -1,20 +1,14 @@ -/* sound/soc/samsung/s3c2412-i2s.c - * - * ALSA Soc Audio Layer - S3C2412 I2S driver - * - * Copyright (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory@wolfsonmicro.com - * linux@wolfsonmicro.com - * - * Copyright (c) 2007, 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// ALSA Soc Audio Layer - S3C2412 I2S driver +// +// Copyright (c) 2006 Wolfson Microelectronics PLC. +// Graeme Gregory graeme.gregory@wolfsonmicro.com +// linux@wolfsonmicro.com +// +// Copyright (c) 2007, 2004-2005 Simtec Electronics +// http://armlinux.simtec.co.uk/ +// Ben Dooks #include #include diff --git a/sound/soc/samsung/s3c2412-i2s.h b/sound/soc/samsung/s3c2412-i2s.h index 02ad5794c0a9..bff2a797cb08 100644 --- a/sound/soc/samsung/s3c2412-i2s.h +++ b/sound/soc/samsung/s3c2412-i2s.h @@ -1,16 +1,11 @@ -/* sound/soc/samsung/s3c2412-i2s.c - * +/* SPDX-License-Identifier: GPL-2.0+ */ +/* * ALSA Soc Audio Layer - S3C2412 I2S driver * * Copyright (c) 2007 Simtec Electronics * http://armlinux.simtec.co.uk/ * Ben Dooks - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. -*/ + */ #ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H #define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ -- cgit v1.2.3-59-g8ed1b From ac64529df8dfc127f3e212d51ff16f3ceb092da4 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:52 +0200 Subject: ASoC: samsung: s3c24xx_simtec*: Convert to SPDX License Identifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx_simtec.c | 11 +++-------- sound/soc/samsung/s3c24xx_simtec.h | 10 +++------- sound/soc/samsung/s3c24xx_simtec_hermes.c | 11 +++-------- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 11 +++-------- 4 files changed, 12 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index 6de63f3e37d5..4543705b8d87 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -1,11 +1,6 @@ -/* sound/soc/samsung/s3c24xx_simtec.c - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright 2009 Simtec Electronics #include #include diff --git a/sound/soc/samsung/s3c24xx_simtec.h b/sound/soc/samsung/s3c24xx_simtec.h index 8270748a2c41..38d8384755cd 100644 --- a/sound/soc/samsung/s3c24xx_simtec.h +++ b/sound/soc/samsung/s3c24xx_simtec.h @@ -1,11 +1,7 @@ -/* sound/soc/samsung/s3c24xx_simtec.h - * +/* SPDX-License-Identifier: GPL-2.0 */ +/* * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ + */ extern void simtec_audio_init(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index 7ac924c595bf..e3528e74a338 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -1,11 +1,6 @@ -/* sound/soc/samsung/s3c24xx_simtec_hermes.c - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright 2009 Simtec Electronics #include #include diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index b4ed2fc1a65c..1360b881400d 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -1,11 +1,6 @@ -/* sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright 2009 Simtec Electronics #include #include -- cgit v1.2.3-59-g8ed1b From 4d19f2cf5f2d292b72e848f8ac6ffa019a5e9c75 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:53 +0200 Subject: ASoC: samsung: s3c24xx_uda134x: Convert to SPDX License Identifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx_uda134x.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 5fb3bab6bbfe..9d68f8ca1fcc 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -1,15 +1,11 @@ -/* - * Modifications by Christian Pellegrin - * - * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver - * - * Copyright 2007 Dension Audio Systems Ltd. - * Author: Zoltan Devai - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Modifications by Christian Pellegrin +// +// s3c24xx_uda134x.c - S3C24XX_UDA134X ALSA SoC Audio board driver +// +// Copyright 2007 Dension Audio Systems Ltd. +// Author: Zoltan Devai #include #include -- cgit v1.2.3-59-g8ed1b From 8aa0ef201cdc43ea91bfc78f88c313b32c4b1ad3 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:54 +0200 Subject: ASoC: samsung: smartq_wm8987: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/smartq_wm8987.c | 21 +++++++-------------- 1 file changed, 7 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index cf0f54e652c1..b9e887ea60b2 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -1,17 +1,10 @@ -/* sound/soc/samsung/smartq_wm8987.c - * - * Copyright 2010 Maurus Cuelenaere - * - * Based on smdk6410_wm8987.c - * Copyright 2007 Wolfson Microelectronics PLC. - linux@wolfsonmicro.com - * Graeme Gregory - graeme.gregory@wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright 2010 Maurus Cuelenaere +// +// Based on smdk6410_wm8987.c +// Copyright 2007 Wolfson Microelectronics PLC. - linux@wolfsonmicro.com +// Graeme Gregory - graeme.gregory@wolfsonmicro.com #include #include -- cgit v1.2.3-59-g8ed1b From 64301240f86a3bc23adce5ac35c98f6632ad63f1 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:55 +0200 Subject: ASoC: samsung: smdk_spdif: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_spdif.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index 7fc7cc6d1530..87a70d872c00 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -1,14 +1,8 @@ -/* - * smdk_spdif.c -- S/PDIF audio for SMDK - * - * Copyright 2010 Samsung Electronics Co. Ltd. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License as - * published by the Free Software Foundation; either version 2 of the - * License, or (at your option) any later version. - * - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// smdk_spdif.c - S/PDIF audio for SMDK +// +// Copyright (C) 2010 Samsung Electronics Co., Ltd. #include #include -- cgit v1.2.3-59-g8ed1b From fe20039c8d09d41f9c6773ff96067b10ed9c87a4 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:56 +0200 Subject: ASoC: samsung: smdk_wm8580: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8580.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index 6e4dfa7e2c89..987807e6f8c3 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -1,14 +1,7 @@ -/* - * smdk_wm8580.c - * - * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright (c) 2009 Samsung Electronics Co. Ltd +// Author: Jaswinder Singh #include #include -- cgit v1.2.3-59-g8ed1b From f9ff38071cdc2c1c2bbc0e2b81b025476c152982 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:57 +0200 Subject: ASoC: samsung: smdk_wm8994*: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994.c | 9 +-------- sound/soc/samsung/smdk_wm8994pcm.c | 16 +++++----------- 2 files changed, 6 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index ff57b192d37d..135d8c2745be 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -1,11 +1,4 @@ -/* - * smdk_wm8994.c - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ #include "../codecs/wm8994.h" #include diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 2e621496be8b..43171d6457fa 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -1,14 +1,8 @@ -/* - * sound/soc/samsung/smdk_wm8994pcm.c - * - * Copyright (c) 2011 Samsung Electronics Co., Ltd - * http://www.samsung.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright (c) 2011 Samsung Electronics Co., Ltd +// http://www.samsung.com + #include #include #include -- cgit v1.2.3-59-g8ed1b From ebf4c423dd2a53f3a65f2a1645cc50b4db03de9e Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:58 +0200 Subject: ASoC: samsung: snow: Convert to SPDX License Identifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/snow.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 5d8efc2d5c38..57ce90fe5004 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -1,15 +1,6 @@ -/* - * ASoC machine driver for Snow boards - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ASoC machine driver for Snow boards #include #include -- cgit v1.2.3-59-g8ed1b From dbf0344a4b6ad38732fa246c1b8ea93435af58e0 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:21:59 +0200 Subject: ASoC: samsung: spdif: Convert to SPDX License Identifier Replace GPL v2.0 license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/spdif.c | 17 ++++++----------- sound/soc/samsung/spdif.h | 8 ++------ 2 files changed, 8 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 5e4afb330416..805c57986e0b 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -1,14 +1,9 @@ -/* sound/soc/samsung/spdif.c - * - * ALSA SoC Audio Layer - Samsung S/PDIF Controller driver - * - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * http://www.samsung.com/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Audio Layer - Samsung S/PDIF Controller driver +// +// Copyright (c) 2010 Samsung Electronics Co. Ltd +// http://www.samsung.com/ #include #include diff --git a/sound/soc/samsung/spdif.h b/sound/soc/samsung/spdif.h index 4f72cb446dbf..461da60ab040 100644 --- a/sound/soc/samsung/spdif.h +++ b/sound/soc/samsung/spdif.h @@ -1,13 +1,9 @@ -/* sound/soc/samsung/spdif.h - * +/* SPDX-License-Identifier: GPL-2.0 */ +/* * ALSA SoC Audio Layer - Samsung S/PDIF Controller driver * * Copyright (c) 2010 Samsung Electronics Co. Ltd * http://www.samsung.com/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SND_SOC_SAMSUNG_SPDIF_H -- cgit v1.2.3-59-g8ed1b From 3f8b21fd7960371bfd9f40955e5eb6be8db80bf9 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:22:00 +0200 Subject: ASoC: samsung: speyside: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/speyside.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 4b4147d07804..15465c84daa3 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -1,13 +1,8 @@ -/* - * Speyside audio support - * - * Copyright 2011 Wolfson Microelectronics - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Speyside audio support +// +// Copyright 2011 Wolfson Microelectronics #include #include -- cgit v1.2.3-59-g8ed1b From 686f47bb72dd77d372bffdc72e257be752752c43 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:22:01 +0200 Subject: ASoC: samsung: tm2_wm5110: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/tm2_wm5110.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index dc93941e01c3..31f4256c6c65 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -1,14 +1,9 @@ -/* - * Copyright (C) 2015 - 2016 Samsung Electronics Co., Ltd. - * - * Authors: Inha Song - * Sylwester Nawrocki - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Copyright (C) 2015 - 2016 Samsung Electronics Co., Ltd. +// +// Authors: Inha Song +// Sylwester Nawrocki #include #include -- cgit v1.2.3-59-g8ed1b From d18360e07fb73c61b5bd850daa9744d986e44d50 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 19 Apr 2019 12:22:02 +0200 Subject: ASoC: samsung: tobermory: Convert to SPDX License Identifier Replace GPL v2.0+ license statements with SPDX license identifier. Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/tobermory.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 998727cb4c31..14b11acb12a4 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -1,13 +1,8 @@ -/* - * Tobermory audio support - * - * Copyright 2011 Wolfson Microelectronics - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// Tobermory audio support +// +// Copyright 2011 Wolfson Microelectronics #include #include -- cgit v1.2.3-59-g8ed1b From ae061d2a857f84ad47b77b04c3ad37478651ab6c Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 19 Apr 2019 09:53:12 +0800 Subject: ASoC: pcm: save fixed-up hw_params of BE Some drivers mandate setting up hw params after resuming from system sleep. Since, the hw_params ioctl is not invoked upon resuming, the fixed-up BE dai hw params should be saved so the driver can use it in its resume sequence. Signed-off-by: Libin Yang Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2d5d5cac4ba6..035dedee78f1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2156,6 +2156,10 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) } } + /* copy the fixed-up hw params for BE dai */ + memcpy(&be->dpcm[stream].hw_params, &dpcm->hw_params, + sizeof(struct snd_pcm_hw_params)); + /* only allow hw_params() if no connected FEs are running */ if (!snd_soc_dpcm_can_be_params(fe, be, stream)) continue; -- cgit v1.2.3-59-g8ed1b From 479879701758206a2cc75176119fcc9dacc40846 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Thu, 18 Apr 2019 10:48:48 +0800 Subject: ASoC: Intel: Haswell: Remove set but not used variable 'stage_type' Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/intel/haswell/sst-haswell-ipc.c: In function 'hsw_stream_message': sound/soc/intel/haswell/sst-haswell-ipc.c:669:29: warning: variable 'stage_type' set but not used [-Wunused-but-set-variable] It is never used since introduction in commit ba57f68235cf ("ASoC: Intel: create haswell folder and move haswell platform files in") Signed-off-by: YueHaibing Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-ipc.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 31fcdf12c67d..74acf9c65161 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -345,11 +345,6 @@ static inline u32 msg_get_stream_type(u32 msg) return (msg & IPC_STR_TYPE_MASK) >> IPC_STR_TYPE_SHIFT; } -static inline u32 msg_get_stage_type(u32 msg) -{ - return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; -} - static inline u32 msg_get_stream_id(u32 msg) { return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; @@ -666,13 +661,12 @@ static int hsw_module_message(struct sst_hsw *hsw, u32 header) static int hsw_stream_message(struct sst_hsw *hsw, u32 header) { - u32 stream_msg, stream_id, stage_type; + u32 stream_msg, stream_id; struct sst_hsw_stream *stream; int handled = 0; stream_msg = msg_get_stream_type(header); stream_id = msg_get_stream_id(header); - stage_type = msg_get_stage_type(header); stream = get_stream_by_id(hsw, stream_id); if (stream == NULL) -- cgit v1.2.3-59-g8ed1b From 01c8327667c249818d3712c3e25c7ad2aca7f389 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Sat, 13 Apr 2019 21:18:12 +0800 Subject: ASoC: codec: hdac_hdmi add device_link to card device In resume from S3, HDAC HDMI codec driver dapm event callback may be operated before HDMI codec driver turns on the display audio power domain because of the contest between display driver and hdmi codec driver. This patch adds the device_link between soc card device (consumer) and hdmi codec device (supplier) to make sure the sequence is always correct. Signed-off-by: Libin Yang Reviewed-by: Takashi Iwai Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/hdac_hdmi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 5eeb0fe836a9..4de1fbfa8827 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1854,6 +1854,17 @@ static int hdmi_codec_probe(struct snd_soc_component *component) /* Imp: Store the card pointer in hda_codec */ hdmi->card = dapm->card->snd_card; + /* + * Setup a device_link between card device and HDMI codec device. + * The card device is the consumer and the HDMI codec device is + * the supplier. With this setting, we can make sure that the audio + * domain in display power will be always turned on before operating + * on the HDMI audio codec registers. + * Let's use the flag DL_FLAG_AUTOREMOVE_CONSUMER. This can make + * sure the device link is freed when the machine driver is removed. + */ + device_link_add(component->card->dev, &hdev->dev, DL_FLAG_RPM_ACTIVE | + DL_FLAG_AUTOREMOVE_CONSUMER); /* * hdac_device core already sets the state to active and calls * get_noresume. So enable runtime and set the device to suspend. -- cgit v1.2.3-59-g8ed1b From 328e9f6973be2ee67862cb17bf6c0c5c5918cd72 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 Apr 2019 13:00:03 +0200 Subject: ALSA: usb-audio: Handle the error from snd_usb_mixer_apply_create_quirk() The error from snd_usb_mixer_apply_create_quirk() is ignored in the current usb-audio driver code, which will continue the probing even after the error. Let's take it more serious. Fixes: 7b1eda223deb ("ALSA: usb-mixer: factor out quirks") Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 73d7dff425c1..c095d9751924 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3490,7 +3490,9 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, if (err < 0) goto _error; - snd_usb_mixer_apply_create_quirk(mixer); + err = snd_usb_mixer_apply_create_quirk(mixer); + if (err < 0) + goto _error; err = snd_device_new(chip->card, SNDRV_DEV_CODEC, mixer, &dev_ops); if (err < 0) -- cgit v1.2.3-59-g8ed1b From 66287def435315d9d8de740da4c543e37630b897 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 25 Apr 2019 15:16:58 +0900 Subject: ASoC: rsnd: fixup 6ch settings to 8ch rsnd need to use 8ch clock settings for 6ch for TDM. Otherwise, it can't work correctly. This patch fixup it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 18 +++++++++++++----- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 2 ++ 3 files changed, 16 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4fe83e611c01..37cb61553d5f 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -300,6 +300,18 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, return chan; } +int rsnd_channel_normalization(int chan) +{ + if ((chan > 8) || (chan < 0)) + return 0; + + /* TDM Extend Mode needs 8ch */ + if (chan == 6) + chan = 8; + + return chan; +} + int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, struct snd_pcm_hw_params *params) { @@ -312,11 +324,7 @@ int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, if (rsnd_runtime_is_multi_ssi(io)) chan /= rsnd_rdai_ssi_lane_get(rdai); - /* TDM Extend Mode needs 8ch */ - if (chan == 6) - chan = 8; - - return chan; + return rsnd_channel_normalization(chan); } int rsnd_runtime_is_multi_ssi(struct rsnd_dai_stream *io) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0e6ef4e18400..7727add3eb1a 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -446,6 +446,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); +int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ rsnd_runtime_channel_original_with_params(io, NULL) int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f5afab631abb..44bda210256e 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -303,6 +303,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_runtime_is_tdm_split(io)) chan = rsnd_io_converted_chan(io); + chan = rsnd_channel_normalization(chan); + main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- cgit v1.2.3-59-g8ed1b From 0bbf4084e0cfbbbff8673f3b9867778d5d4b1d6f Mon Sep 17 00:00:00 2001 From: Fabien Dessenne Date: Wed, 24 Apr 2019 17:28:44 +0200 Subject: ASoC: stm32: i2s: return the get_irq error During probe, return the "get_irq" error value instead of -ENOENT. This allows the driver to be deferred probed if needed. Signed-off-by: Fabien Dessenne Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 8968458eec62..e707d28cb203 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -845,8 +845,9 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, /* Get irqs */ irq = platform_get_irq(pdev, 0); if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); - return -ENOENT; + if (irq != -EPROBE_DEFER) + dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); + return irq; } ret = devm_request_irq(&pdev->dev, irq, stm32_i2s_isr, IRQF_ONESHOT, -- cgit v1.2.3-59-g8ed1b From c46ab1510de1d3dd9871ae36fbec0da3e0d2d6c5 Mon Sep 17 00:00:00 2001 From: Xun Zhang Date: Fri, 19 Apr 2019 15:53:49 -0500 Subject: ASoC: codecs: rt5682: initialize mutex before using In rt5682 codec driver, a mutex called "calibrate_mutex" is used in rt5682_calibrate() before initialization, which causes warning in lock debug. Move the initialization before the usage of mutex. Signed-off-by: Xun Zhang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 86a7fa31c294..505fb3d7b1c5 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2588,6 +2588,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, rt5682_reset(rt5682->regmap); + mutex_init(&rt5682->calibrate_mutex); rt5682_calibrate(rt5682); ret = regmap_multi_reg_write(rt5682->regmap, patch_list, @@ -2654,7 +2655,6 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, INIT_DELAYED_WORK(&rt5682->jd_check_work, rt5682_jd_check_handler); - mutex_init(&rt5682->calibrate_mutex); if (i2c->irq) { ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, -- cgit v1.2.3-59-g8ed1b From d7bff893e04f4e7b4a0789486da1435df5920f5a Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Tue, 23 Apr 2019 10:26:22 +0800 Subject: ASoC: sprd: Add Spreadtrum multi-channel data transfer support On Spreadtrum platform, the audio subsystem will use the multi-channel data transfer controller to transfer sound stream between audio subsystem and other AP/CP subsystem. It can support 10 DAC channel and 10 ADC channel, and each channel has 512 bytes depth data fifo. Moreover each channel can be used DMA mode or interrupt mode to transfer data. Signed-off-by: Baolin Wang Signed-off-by: Mark Brown --- sound/soc/sprd/Kconfig | 8 + sound/soc/sprd/Makefile | 2 + sound/soc/sprd/sprd-mcdt.c | 1011 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sprd/sprd-mcdt.h | 107 +++++ 4 files changed, 1128 insertions(+) create mode 100644 sound/soc/sprd/sprd-mcdt.c create mode 100644 sound/soc/sprd/sprd-mcdt.h (limited to 'sound') diff --git a/sound/soc/sprd/Kconfig b/sound/soc/sprd/Kconfig index 3b1eb320dab4..21f9cc34f8bd 100644 --- a/sound/soc/sprd/Kconfig +++ b/sound/soc/sprd/Kconfig @@ -5,3 +5,11 @@ config SND_SOC_SPRD help Say Y or M if you want to add support for codecs attached to the Spreadtrum SoCs' Audio interfaces. + +config SND_SOC_SPRD_MCDT + bool "Spreadtrum multi-channel data transfer support" + depends on SND_SOC_SPRD + help + Say y here to enable multi-channel data transfer support. It + is used for sound stream transmission between audio subsystem + and other AP/CP subsystem. diff --git a/sound/soc/sprd/Makefile b/sound/soc/sprd/Makefile index e6c2606da8e2..a95fa56cd000 100644 --- a/sound/soc/sprd/Makefile +++ b/sound/soc/sprd/Makefile @@ -4,3 +4,5 @@ snd-soc-sprd-platform-objs := sprd-pcm-dma.o sprd-pcm-compress.o obj-$(CONFIG_SND_SOC_SPRD) += snd-soc-sprd-platform.o + +obj-$(CONFIG_SND_SOC_SPRD_MCDT) += sprd-mcdt.o diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c new file mode 100644 index 000000000000..28f5e649733d --- /dev/null +++ b/sound/soc/sprd/sprd-mcdt.c @@ -0,0 +1,1011 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (C) 2019 Spreadtrum Communications Inc. + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sprd-mcdt.h" + +/* MCDT registers definition */ +#define MCDT_CH0_TXD 0x0 +#define MCDT_CH0_RXD 0x28 +#define MCDT_DAC0_WTMK 0x60 +#define MCDT_ADC0_WTMK 0x88 +#define MCDT_DMA_EN 0xb0 + +#define MCDT_INT_EN0 0xb4 +#define MCDT_INT_EN1 0xb8 +#define MCDT_INT_EN2 0xbc + +#define MCDT_INT_CLR0 0xc0 +#define MCDT_INT_CLR1 0xc4 +#define MCDT_INT_CLR2 0xc8 + +#define MCDT_INT_RAW1 0xcc +#define MCDT_INT_RAW2 0xd0 +#define MCDT_INT_RAW3 0xd4 + +#define MCDT_INT_MSK1 0xd8 +#define MCDT_INT_MSK2 0xdc +#define MCDT_INT_MSK3 0xe0 + +#define MCDT_DAC0_FIFO_ADDR_ST 0xe4 +#define MCDT_ADC0_FIFO_ADDR_ST 0xe8 + +#define MCDT_CH_FIFO_ST0 0x134 +#define MCDT_CH_FIFO_ST1 0x138 +#define MCDT_CH_FIFO_ST2 0x13c + +#define MCDT_INT_MSK_CFG0 0x140 +#define MCDT_INT_MSK_CFG1 0x144 + +#define MCDT_DMA_CFG0 0x148 +#define MCDT_FIFO_CLR 0x14c +#define MCDT_DMA_CFG1 0x150 +#define MCDT_DMA_CFG2 0x154 +#define MCDT_DMA_CFG3 0x158 +#define MCDT_DMA_CFG4 0x15c +#define MCDT_DMA_CFG5 0x160 + +/* Channel water mark definition */ +#define MCDT_CH_FIFO_AE_SHIFT 16 +#define MCDT_CH_FIFO_AE_MASK GENMASK(24, 16) +#define MCDT_CH_FIFO_AF_MASK GENMASK(8, 0) + +/* DMA channel select definition */ +#define MCDT_DMA_CH0_SEL_MASK GENMASK(3, 0) +#define MCDT_DMA_CH0_SEL_SHIFT 0 +#define MCDT_DMA_CH1_SEL_MASK GENMASK(7, 4) +#define MCDT_DMA_CH1_SEL_SHIFT 4 +#define MCDT_DMA_CH2_SEL_MASK GENMASK(11, 8) +#define MCDT_DMA_CH2_SEL_SHIFT 8 +#define MCDT_DMA_CH3_SEL_MASK GENMASK(15, 12) +#define MCDT_DMA_CH3_SEL_SHIFT 12 +#define MCDT_DMA_CH4_SEL_MASK GENMASK(19, 16) +#define MCDT_DMA_CH4_SEL_SHIFT 16 +#define MCDT_DAC_DMA_SHIFT 16 + +/* DMA channel ACK select definition */ +#define MCDT_DMA_ACK_SEL_MASK GENMASK(3, 0) + +/* Channel FIFO definition */ +#define MCDT_CH_FIFO_ADDR_SHIFT 16 +#define MCDT_CH_FIFO_ADDR_MASK GENMASK(9, 0) +#define MCDT_ADC_FIFO_SHIFT 16 +#define MCDT_FIFO_LENGTH 512 + +#define MCDT_ADC_CHANNEL_NUM 10 +#define MCDT_DAC_CHANNEL_NUM 10 +#define MCDT_CHANNEL_NUM (MCDT_ADC_CHANNEL_NUM + MCDT_DAC_CHANNEL_NUM) + +enum sprd_mcdt_fifo_int { + MCDT_ADC_FIFO_AE_INT, + MCDT_ADC_FIFO_AF_INT, + MCDT_DAC_FIFO_AE_INT, + MCDT_DAC_FIFO_AF_INT, + MCDT_ADC_FIFO_OV_INT, + MCDT_DAC_FIFO_OV_INT +}; + +enum sprd_mcdt_fifo_sts { + MCDT_ADC_FIFO_REAL_FULL, + MCDT_ADC_FIFO_REAL_EMPTY, + MCDT_ADC_FIFO_AF, + MCDT_ADC_FIFO_AE, + MCDT_DAC_FIFO_REAL_FULL, + MCDT_DAC_FIFO_REAL_EMPTY, + MCDT_DAC_FIFO_AF, + MCDT_DAC_FIFO_AE +}; + +struct sprd_mcdt_dev { + struct device *dev; + void __iomem *base; + spinlock_t lock; + struct sprd_mcdt_chan chan[MCDT_CHANNEL_NUM]; +}; + +static LIST_HEAD(sprd_mcdt_chan_list); +static DEFINE_MUTEX(sprd_mcdt_list_mutex); + +static void sprd_mcdt_update(struct sprd_mcdt_dev *mcdt, u32 reg, u32 val, + u32 mask) +{ + u32 orig = readl_relaxed(mcdt->base + reg); + u32 tmp; + + tmp = (orig & ~mask) | val; + writel_relaxed(tmp, mcdt->base + reg); +} + +static void sprd_mcdt_dac_set_watermark(struct sprd_mcdt_dev *mcdt, u8 channel, + u32 full, u32 empty) +{ + u32 reg = MCDT_DAC0_WTMK + channel * 4; + u32 water_mark = + (empty << MCDT_CH_FIFO_AE_SHIFT) & MCDT_CH_FIFO_AE_MASK; + + water_mark |= full & MCDT_CH_FIFO_AF_MASK; + sprd_mcdt_update(mcdt, reg, water_mark, + MCDT_CH_FIFO_AE_MASK | MCDT_CH_FIFO_AF_MASK); +} + +static void sprd_mcdt_adc_set_watermark(struct sprd_mcdt_dev *mcdt, u8 channel, + u32 full, u32 empty) +{ + u32 reg = MCDT_ADC0_WTMK + channel * 4; + u32 water_mark = + (empty << MCDT_CH_FIFO_AE_SHIFT) & MCDT_CH_FIFO_AE_MASK; + + water_mark |= full & MCDT_CH_FIFO_AF_MASK; + sprd_mcdt_update(mcdt, reg, water_mark, + MCDT_CH_FIFO_AE_MASK | MCDT_CH_FIFO_AF_MASK); +} + +static void sprd_mcdt_dac_dma_enable(struct sprd_mcdt_dev *mcdt, u8 channel, + bool enable) +{ + u32 shift = MCDT_DAC_DMA_SHIFT + channel; + + if (enable) + sprd_mcdt_update(mcdt, MCDT_DMA_EN, BIT(shift), BIT(shift)); + else + sprd_mcdt_update(mcdt, MCDT_DMA_EN, 0, BIT(shift)); +} + +static void sprd_mcdt_adc_dma_enable(struct sprd_mcdt_dev *mcdt, u8 channel, + bool enable) +{ + if (enable) + sprd_mcdt_update(mcdt, MCDT_DMA_EN, BIT(channel), BIT(channel)); + else + sprd_mcdt_update(mcdt, MCDT_DMA_EN, 0, BIT(channel)); +} + +static void sprd_mcdt_ap_int_enable(struct sprd_mcdt_dev *mcdt, u8 channel, + bool enable) +{ + if (enable) + sprd_mcdt_update(mcdt, MCDT_INT_MSK_CFG0, BIT(channel), + BIT(channel)); + else + sprd_mcdt_update(mcdt, MCDT_INT_MSK_CFG0, 0, BIT(channel)); +} + +static void sprd_mcdt_dac_write_fifo(struct sprd_mcdt_dev *mcdt, u8 channel, + u32 val) +{ + u32 reg = MCDT_CH0_TXD + channel * 4; + + writel_relaxed(val, mcdt->base + reg); +} + +static void sprd_mcdt_adc_read_fifo(struct sprd_mcdt_dev *mcdt, u8 channel, + u32 *val) +{ + u32 reg = MCDT_CH0_RXD + channel * 4; + + *val = readl_relaxed(mcdt->base + reg); +} + +static void sprd_mcdt_dac_dma_chn_select(struct sprd_mcdt_dev *mcdt, u8 channel, + enum sprd_mcdt_dma_chan dma_chan) +{ + switch (dma_chan) { + case SPRD_MCDT_DMA_CH0: + sprd_mcdt_update(mcdt, MCDT_DMA_CFG0, + channel << MCDT_DMA_CH0_SEL_SHIFT, + MCDT_DMA_CH0_SEL_MASK); + break; + + case SPRD_MCDT_DMA_CH1: + sprd_mcdt_update(mcdt, MCDT_DMA_CFG0, + channel << MCDT_DMA_CH1_SEL_SHIFT, + MCDT_DMA_CH1_SEL_MASK); + break; + + case SPRD_MCDT_DMA_CH2: + sprd_mcdt_update(mcdt, MCDT_DMA_CFG0, + channel << MCDT_DMA_CH2_SEL_SHIFT, + MCDT_DMA_CH2_SEL_MASK); + break; + + case SPRD_MCDT_DMA_CH3: + sprd_mcdt_update(mcdt, MCDT_DMA_CFG0, + channel << MCDT_DMA_CH3_SEL_SHIFT, + MCDT_DMA_CH3_SEL_MASK); + break; + + case SPRD_MCDT_DMA_CH4: + sprd_mcdt_update(mcdt, MCDT_DMA_CFG0, + channel << MCDT_DMA_CH4_SEL_SHIFT, + MCDT_DMA_CH4_SEL_MASK); + break; + } +} + +static void sprd_mcdt_adc_dma_chn_select(struct sprd_mcdt_dev *mcdt, u8 channel, + enum sprd_mcdt_dma_chan dma_chan) +{ + switch (dma_chan) { + case SPRD_MCDT_DMA_CH0: + sprd_mcdt_update(mcdt, MCDT_DMA_CFG1, + channel << MCDT_DMA_CH0_SEL_SHIFT, + MCDT_DMA_CH0_SEL_MASK); + break; + + case SPRD_MCDT_DMA_CH1: + sprd_mcdt_update(mcdt, MCDT_DMA_CFG1, + channel << MCDT_DMA_CH1_SEL_SHIFT, + MCDT_DMA_CH1_SEL_MASK); + break; + + case SPRD_MCDT_DMA_CH2: + sprd_mcdt_update(mcdt, MCDT_DMA_CFG1, + channel << MCDT_DMA_CH2_SEL_SHIFT, + MCDT_DMA_CH2_SEL_MASK); + break; + + case SPRD_MCDT_DMA_CH3: + sprd_mcdt_update(mcdt, MCDT_DMA_CFG1, + channel << MCDT_DMA_CH3_SEL_SHIFT, + MCDT_DMA_CH3_SEL_MASK); + break; + + case SPRD_MCDT_DMA_CH4: + sprd_mcdt_update(mcdt, MCDT_DMA_CFG1, + channel << MCDT_DMA_CH4_SEL_SHIFT, + MCDT_DMA_CH4_SEL_MASK); + break; + } +} + +static u32 sprd_mcdt_dma_ack_shift(u8 channel) +{ + switch (channel) { + default: + case 0: + case 8: + return 0; + case 1: + case 9: + return 4; + case 2: + return 8; + case 3: + return 12; + case 4: + return 16; + case 5: + return 20; + case 6: + return 24; + case 7: + return 28; + } +} + +static void sprd_mcdt_dac_dma_ack_select(struct sprd_mcdt_dev *mcdt, u8 channel, + enum sprd_mcdt_dma_chan dma_chan) +{ + u32 reg, shift = sprd_mcdt_dma_ack_shift(channel), ack = dma_chan; + + switch (channel) { + case 0 ... 7: + reg = MCDT_DMA_CFG2; + break; + + case 8 ... 9: + reg = MCDT_DMA_CFG3; + break; + + default: + return; + } + + sprd_mcdt_update(mcdt, reg, ack << shift, + MCDT_DMA_ACK_SEL_MASK << shift); +} + +static void sprd_mcdt_adc_dma_ack_select(struct sprd_mcdt_dev *mcdt, u8 channel, + enum sprd_mcdt_dma_chan dma_chan) +{ + u32 reg, shift = sprd_mcdt_dma_ack_shift(channel), ack = dma_chan; + + switch (channel) { + case 0 ... 7: + reg = MCDT_DMA_CFG4; + break; + + case 8 ... 9: + reg = MCDT_DMA_CFG5; + break; + + default: + return; + } + + sprd_mcdt_update(mcdt, reg, ack << shift, + MCDT_DMA_ACK_SEL_MASK << shift); +} + +static bool sprd_mcdt_chan_fifo_sts(struct sprd_mcdt_dev *mcdt, u8 channel, + enum sprd_mcdt_fifo_sts fifo_sts) +{ + u32 reg, shift; + + switch (channel) { + case 0 ... 3: + reg = MCDT_CH_FIFO_ST0; + break; + case 4 ... 7: + reg = MCDT_CH_FIFO_ST1; + break; + case 8 ... 9: + reg = MCDT_CH_FIFO_ST2; + break; + default: + return false; + } + + switch (channel) { + case 0: + case 4: + case 8: + shift = fifo_sts; + break; + + case 1: + case 5: + case 9: + shift = 8 + fifo_sts; + break; + + case 2: + case 6: + shift = 16 + fifo_sts; + break; + + case 3: + case 7: + shift = 24 + fifo_sts; + break; + + default: + return false; + } + + return !!(readl_relaxed(mcdt->base + reg) & BIT(shift)); +} + +static void sprd_mcdt_dac_fifo_clear(struct sprd_mcdt_dev *mcdt, u8 channel) +{ + sprd_mcdt_update(mcdt, MCDT_FIFO_CLR, BIT(channel), BIT(channel)); +} + +static void sprd_mcdt_adc_fifo_clear(struct sprd_mcdt_dev *mcdt, u8 channel) +{ + u32 shift = MCDT_ADC_FIFO_SHIFT + channel; + + sprd_mcdt_update(mcdt, MCDT_FIFO_CLR, BIT(shift), BIT(shift)); +} + +static u32 sprd_mcdt_dac_fifo_avail(struct sprd_mcdt_dev *mcdt, u8 channel) +{ + u32 reg = MCDT_DAC0_FIFO_ADDR_ST + channel * 8; + u32 r_addr = (readl_relaxed(mcdt->base + reg) >> + MCDT_CH_FIFO_ADDR_SHIFT) & MCDT_CH_FIFO_ADDR_MASK; + u32 w_addr = readl_relaxed(mcdt->base + reg) & MCDT_CH_FIFO_ADDR_MASK; + + if (w_addr >= r_addr) + return 4 * (MCDT_FIFO_LENGTH - w_addr + r_addr); + else + return 4 * (r_addr - w_addr); +} + +static u32 sprd_mcdt_adc_fifo_avail(struct sprd_mcdt_dev *mcdt, u8 channel) +{ + u32 reg = MCDT_ADC0_FIFO_ADDR_ST + channel * 8; + u32 r_addr = (readl_relaxed(mcdt->base + reg) >> + MCDT_CH_FIFO_ADDR_SHIFT) & MCDT_CH_FIFO_ADDR_MASK; + u32 w_addr = readl_relaxed(mcdt->base + reg) & MCDT_CH_FIFO_ADDR_MASK; + + if (w_addr >= r_addr) + return 4 * (w_addr - r_addr); + else + return 4 * (MCDT_FIFO_LENGTH - r_addr + w_addr); +} + +static u32 sprd_mcdt_int_type_shift(u8 channel, + enum sprd_mcdt_fifo_int int_type) +{ + switch (channel) { + case 0: + case 4: + case 8: + return int_type; + + case 1: + case 5: + case 9: + return 8 + int_type; + + case 2: + case 6: + return 16 + int_type; + + case 3: + case 7: + return 24 + int_type; + + default: + return 0; + } +} + +static void sprd_mcdt_chan_int_en(struct sprd_mcdt_dev *mcdt, u8 channel, + enum sprd_mcdt_fifo_int int_type, bool enable) +{ + u32 reg, shift = sprd_mcdt_int_type_shift(channel, int_type); + + switch (channel) { + case 0 ... 3: + reg = MCDT_INT_EN0; + break; + case 4 ... 7: + reg = MCDT_INT_EN1; + break; + case 8 ... 9: + reg = MCDT_INT_EN2; + break; + default: + return; + } + + if (enable) + sprd_mcdt_update(mcdt, reg, BIT(shift), BIT(shift)); + else + sprd_mcdt_update(mcdt, reg, 0, BIT(shift)); +} + +static void sprd_mcdt_chan_int_clear(struct sprd_mcdt_dev *mcdt, u8 channel, + enum sprd_mcdt_fifo_int int_type) +{ + u32 reg, shift = sprd_mcdt_int_type_shift(channel, int_type); + + switch (channel) { + case 0 ... 3: + reg = MCDT_INT_CLR0; + break; + case 4 ... 7: + reg = MCDT_INT_CLR1; + break; + case 8 ... 9: + reg = MCDT_INT_CLR2; + break; + default: + return; + } + + sprd_mcdt_update(mcdt, reg, BIT(shift), BIT(shift)); +} + +static bool sprd_mcdt_chan_int_sts(struct sprd_mcdt_dev *mcdt, u8 channel, + enum sprd_mcdt_fifo_int int_type) +{ + u32 reg, shift = sprd_mcdt_int_type_shift(channel, int_type); + + switch (channel) { + case 0 ... 3: + reg = MCDT_INT_MSK1; + break; + case 4 ... 7: + reg = MCDT_INT_MSK2; + break; + case 8 ... 9: + reg = MCDT_INT_MSK3; + break; + default: + return false; + } + + return !!(readl_relaxed(mcdt->base + reg) & BIT(shift)); +} + +static irqreturn_t sprd_mcdt_irq_handler(int irq, void *dev_id) +{ + struct sprd_mcdt_dev *mcdt = (struct sprd_mcdt_dev *)dev_id; + int i; + + spin_lock(&mcdt->lock); + + for (i = 0; i < MCDT_ADC_CHANNEL_NUM; i++) { + if (sprd_mcdt_chan_int_sts(mcdt, i, MCDT_ADC_FIFO_AF_INT)) { + struct sprd_mcdt_chan *chan = &mcdt->chan[i]; + + sprd_mcdt_chan_int_clear(mcdt, i, MCDT_ADC_FIFO_AF_INT); + if (chan->cb) + chan->cb->notify(chan->cb->data); + } + } + + for (i = 0; i < MCDT_DAC_CHANNEL_NUM; i++) { + if (sprd_mcdt_chan_int_sts(mcdt, i, MCDT_DAC_FIFO_AE_INT)) { + struct sprd_mcdt_chan *chan = + &mcdt->chan[i + MCDT_ADC_CHANNEL_NUM]; + + sprd_mcdt_chan_int_clear(mcdt, i, MCDT_DAC_FIFO_AE_INT); + if (chan->cb) + chan->cb->notify(chan->cb->data); + } + } + + spin_unlock(&mcdt->lock); + + return IRQ_HANDLED; +} + +/** + * sprd_mcdt_chan_write - write data to the MCDT channel's fifo + * @chan: the MCDT channel + * @tx_buf: send buffer + * @size: data size + * + * Note: We can not write data to the channel fifo when enabling the DMA mode, + * otherwise the channel fifo data will be invalid. + * + * If there are not enough space of the channel fifo, it will return errors + * to users. + * + * Returns 0 on success, or an appropriate error code on failure. + */ +int sprd_mcdt_chan_write(struct sprd_mcdt_chan *chan, char *tx_buf, u32 size) +{ + struct sprd_mcdt_dev *mcdt = chan->mcdt; + unsigned long flags; + int avail, i = 0, words = size / 4; + u32 *buf = (u32 *)tx_buf; + + spin_lock_irqsave(&mcdt->lock, flags); + + if (chan->dma_enable) { + dev_err(mcdt->dev, + "Can not write data when DMA mode enabled\n"); + spin_unlock_irqrestore(&mcdt->lock, flags); + return -EINVAL; + } + + if (sprd_mcdt_chan_fifo_sts(mcdt, chan->id, MCDT_DAC_FIFO_REAL_FULL)) { + dev_err(mcdt->dev, "Channel fifo is full now\n"); + spin_unlock_irqrestore(&mcdt->lock, flags); + return -EBUSY; + } + + avail = sprd_mcdt_dac_fifo_avail(mcdt, chan->id); + if (size > avail) { + dev_err(mcdt->dev, + "Data size is larger than the available fifo size\n"); + spin_unlock_irqrestore(&mcdt->lock, flags); + return -EBUSY; + } + + while (i++ < words) + sprd_mcdt_dac_write_fifo(mcdt, chan->id, *buf++); + + spin_unlock_irqrestore(&mcdt->lock, flags); + return 0; +} +EXPORT_SYMBOL_GPL(sprd_mcdt_chan_write); + +/** + * sprd_mcdt_chan_read - read data from the MCDT channel's fifo + * @chan: the MCDT channel + * @rx_buf: receive buffer + * @size: data size + * + * Note: We can not read data from the channel fifo when enabling the DMA mode, + * otherwise the reading data will be invalid. + * + * Usually user need start to read data once receiving the fifo full interrupt. + * + * Returns data size of reading successfully, or an error code on failure. + */ +int sprd_mcdt_chan_read(struct sprd_mcdt_chan *chan, char *rx_buf, u32 size) +{ + struct sprd_mcdt_dev *mcdt = chan->mcdt; + unsigned long flags; + int i = 0, avail, words = size / 4; + u32 *buf = (u32 *)rx_buf; + + spin_lock_irqsave(&mcdt->lock, flags); + + if (chan->dma_enable) { + dev_err(mcdt->dev, "Can not read data when DMA mode enabled\n"); + spin_unlock_irqrestore(&mcdt->lock, flags); + return -EINVAL; + } + + if (sprd_mcdt_chan_fifo_sts(mcdt, chan->id, MCDT_ADC_FIFO_REAL_EMPTY)) { + dev_err(mcdt->dev, "Channel fifo is empty\n"); + spin_unlock_irqrestore(&mcdt->lock, flags); + return -EBUSY; + } + + avail = sprd_mcdt_adc_fifo_avail(mcdt, chan->id); + if (size > avail) + words = avail / 4; + + while (i++ < words) + sprd_mcdt_adc_read_fifo(mcdt, chan->id, buf++); + + spin_unlock_irqrestore(&mcdt->lock, flags); + return words * 4; +} +EXPORT_SYMBOL_GPL(sprd_mcdt_chan_read); + +/** + * sprd_mcdt_chan_int_enable - enable the interrupt mode for the MCDT channel + * @chan: the MCDT channel + * @water_mark: water mark to trigger a interrupt + * @cb: callback when a interrupt happened + * + * Now it only can enable fifo almost full interrupt for ADC channel and fifo + * almost empty interrupt for DAC channel. Morevoer for interrupt mode, user + * should use sprd_mcdt_chan_read() or sprd_mcdt_chan_write() to read or write + * data manually. + * + * For ADC channel, user can start to read data once receiving one fifo full + * interrupt. For DAC channel, user can start to write data once receiving one + * fifo empty interrupt or just call sprd_mcdt_chan_write() to write data + * directly. + * + * Returns 0 on success, or an error code on failure. + */ +int sprd_mcdt_chan_int_enable(struct sprd_mcdt_chan *chan, u32 water_mark, + struct sprd_mcdt_chan_callback *cb) +{ + struct sprd_mcdt_dev *mcdt = chan->mcdt; + unsigned long flags; + int ret = 0; + + spin_lock_irqsave(&mcdt->lock, flags); + + if (chan->dma_enable || chan->int_enable) { + dev_err(mcdt->dev, "Failed to set interrupt mode.\n"); + spin_unlock_irqrestore(&mcdt->lock, flags); + return -EINVAL; + } + + switch (chan->type) { + case SPRD_MCDT_ADC_CHAN: + sprd_mcdt_adc_fifo_clear(mcdt, chan->id); + sprd_mcdt_adc_set_watermark(mcdt, chan->id, water_mark, + MCDT_FIFO_LENGTH - 1); + sprd_mcdt_chan_int_en(mcdt, chan->id, + MCDT_ADC_FIFO_AF_INT, true); + sprd_mcdt_ap_int_enable(mcdt, chan->id, true); + break; + + case SPRD_MCDT_DAC_CHAN: + sprd_mcdt_dac_fifo_clear(mcdt, chan->id); + sprd_mcdt_dac_set_watermark(mcdt, chan->id, + MCDT_FIFO_LENGTH - 1, water_mark); + sprd_mcdt_chan_int_en(mcdt, chan->id, + MCDT_DAC_FIFO_AE_INT, true); + sprd_mcdt_ap_int_enable(mcdt, chan->id, true); + break; + + default: + dev_err(mcdt->dev, "Unsupported channel type\n"); + ret = -EINVAL; + } + + if (!ret) { + chan->cb = cb; + chan->int_enable = true; + } + + spin_unlock_irqrestore(&mcdt->lock, flags); + + return ret; +} +EXPORT_SYMBOL_GPL(sprd_mcdt_chan_int_enable); + +/** + * sprd_mcdt_chan_int_disable - disable the interrupt mode for the MCDT channel + * @chan: the MCDT channel + */ +void sprd_mcdt_chan_int_disable(struct sprd_mcdt_chan *chan) +{ + struct sprd_mcdt_dev *mcdt = chan->mcdt; + unsigned long flags; + + spin_lock_irqsave(&mcdt->lock, flags); + + if (!chan->int_enable) { + spin_unlock_irqrestore(&mcdt->lock, flags); + return; + } + + switch (chan->type) { + case SPRD_MCDT_ADC_CHAN: + sprd_mcdt_chan_int_en(mcdt, chan->id, + MCDT_ADC_FIFO_AF_INT, false); + sprd_mcdt_chan_int_clear(mcdt, chan->id, MCDT_ADC_FIFO_AF_INT); + sprd_mcdt_ap_int_enable(mcdt, chan->id, false); + break; + + case SPRD_MCDT_DAC_CHAN: + sprd_mcdt_chan_int_en(mcdt, chan->id, + MCDT_DAC_FIFO_AE_INT, false); + sprd_mcdt_chan_int_clear(mcdt, chan->id, MCDT_DAC_FIFO_AE_INT); + sprd_mcdt_ap_int_enable(mcdt, chan->id, false); + break; + + default: + break; + } + + chan->int_enable = false; + spin_unlock_irqrestore(&mcdt->lock, flags); +} +EXPORT_SYMBOL_GPL(sprd_mcdt_chan_int_disable); + +/** + * sprd_mcdt_chan_dma_enable - enable the DMA mode for the MCDT channel + * @chan: the MCDT channel + * @dma_chan: specify which DMA channel will be used for this MCDT channel + * @water_mark: water mark to trigger a DMA request + * + * Enable the DMA mode for the MCDT channel, that means we can use DMA to + * transfer data to the channel fifo and do not need reading/writing data + * manually. + * + * Returns 0 on success, or an error code on failure. + */ +int sprd_mcdt_chan_dma_enable(struct sprd_mcdt_chan *chan, + enum sprd_mcdt_dma_chan dma_chan, + u32 water_mark) +{ + struct sprd_mcdt_dev *mcdt = chan->mcdt; + unsigned long flags; + int ret = 0; + + spin_lock_irqsave(&mcdt->lock, flags); + + if (chan->dma_enable || chan->int_enable || + dma_chan > SPRD_MCDT_DMA_CH4) { + dev_err(mcdt->dev, "Failed to set DMA mode\n"); + spin_unlock_irqrestore(&mcdt->lock, flags); + return -EINVAL; + } + + switch (chan->type) { + case SPRD_MCDT_ADC_CHAN: + sprd_mcdt_adc_fifo_clear(mcdt, chan->id); + sprd_mcdt_adc_set_watermark(mcdt, chan->id, + water_mark, MCDT_FIFO_LENGTH - 1); + sprd_mcdt_adc_dma_enable(mcdt, chan->id, true); + sprd_mcdt_adc_dma_chn_select(mcdt, chan->id, dma_chan); + sprd_mcdt_adc_dma_ack_select(mcdt, chan->id, dma_chan); + break; + + case SPRD_MCDT_DAC_CHAN: + sprd_mcdt_dac_fifo_clear(mcdt, chan->id); + sprd_mcdt_dac_set_watermark(mcdt, chan->id, + MCDT_FIFO_LENGTH - 1, water_mark); + sprd_mcdt_dac_dma_enable(mcdt, chan->id, true); + sprd_mcdt_dac_dma_chn_select(mcdt, chan->id, dma_chan); + sprd_mcdt_dac_dma_ack_select(mcdt, chan->id, dma_chan); + break; + + default: + dev_err(mcdt->dev, "Unsupported channel type\n"); + ret = -EINVAL; + } + + if (!ret) + chan->dma_enable = true; + + spin_unlock_irqrestore(&mcdt->lock, flags); + + return ret; +} +EXPORT_SYMBOL_GPL(sprd_mcdt_chan_dma_enable); + +/** + * sprd_mcdt_chan_dma_disable - disable the DMA mode for the MCDT channel + * @chan: the MCDT channel + */ +void sprd_mcdt_chan_dma_disable(struct sprd_mcdt_chan *chan) +{ + struct sprd_mcdt_dev *mcdt = chan->mcdt; + unsigned long flags; + + spin_lock_irqsave(&mcdt->lock, flags); + + if (!chan->dma_enable) { + spin_unlock_irqrestore(&mcdt->lock, flags); + return; + } + + switch (chan->type) { + case SPRD_MCDT_ADC_CHAN: + sprd_mcdt_adc_dma_enable(mcdt, chan->id, false); + sprd_mcdt_adc_fifo_clear(mcdt, chan->id); + break; + + case SPRD_MCDT_DAC_CHAN: + sprd_mcdt_dac_dma_enable(mcdt, chan->id, false); + sprd_mcdt_dac_fifo_clear(mcdt, chan->id); + break; + + default: + break; + } + + chan->dma_enable = false; + spin_unlock_irqrestore(&mcdt->lock, flags); +} +EXPORT_SYMBOL_GPL(sprd_mcdt_chan_dma_disable); + +/** + * sprd_mcdt_request_chan - request one MCDT channel + * @channel: channel id + * @type: channel type, it can be one ADC channel or DAC channel + * + * Rreturn NULL if no available channel. + */ +struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel, + enum sprd_mcdt_channel_type type) +{ + struct sprd_mcdt_chan *temp, *chan = NULL; + + mutex_lock(&sprd_mcdt_list_mutex); + + list_for_each_entry(temp, &sprd_mcdt_chan_list, list) { + if (temp->type == type && temp->id == channel) { + chan = temp; + break; + } + } + + if (chan) + list_del(&chan->list); + + mutex_unlock(&sprd_mcdt_list_mutex); + + return chan; +} +EXPORT_SYMBOL_GPL(sprd_mcdt_request_chan); + +/** + * sprd_mcdt_free_chan - free one MCDT channel + * @chan: the channel to be freed + */ +void sprd_mcdt_free_chan(struct sprd_mcdt_chan *chan) +{ + struct sprd_mcdt_chan *temp; + + sprd_mcdt_chan_dma_disable(chan); + sprd_mcdt_chan_int_disable(chan); + + mutex_lock(&sprd_mcdt_list_mutex); + + list_for_each_entry(temp, &sprd_mcdt_chan_list, list) { + if (temp == chan) { + mutex_unlock(&sprd_mcdt_list_mutex); + return; + } + } + + list_add_tail(&chan->list, &sprd_mcdt_chan_list); + mutex_unlock(&sprd_mcdt_list_mutex); +} +EXPORT_SYMBOL_GPL(sprd_mcdt_free_chan); + +static void sprd_mcdt_init_chans(struct sprd_mcdt_dev *mcdt, + struct resource *res) +{ + int i; + + for (i = 0; i < MCDT_CHANNEL_NUM; i++) { + struct sprd_mcdt_chan *chan = &mcdt->chan[i]; + + if (i < MCDT_ADC_CHANNEL_NUM) { + chan->id = i; + chan->type = SPRD_MCDT_ADC_CHAN; + chan->fifo_phys = res->start + MCDT_CH0_RXD + i * 4; + } else { + chan->id = i - MCDT_ADC_CHANNEL_NUM; + chan->type = SPRD_MCDT_DAC_CHAN; + chan->fifo_phys = res->start + MCDT_CH0_TXD + + (i - MCDT_ADC_CHANNEL_NUM) * 4; + } + + chan->mcdt = mcdt; + INIT_LIST_HEAD(&chan->list); + + mutex_lock(&sprd_mcdt_list_mutex); + list_add_tail(&chan->list, &sprd_mcdt_chan_list); + mutex_unlock(&sprd_mcdt_list_mutex); + } +} + +static int sprd_mcdt_probe(struct platform_device *pdev) +{ + struct sprd_mcdt_dev *mcdt; + struct resource *res; + int ret, irq; + + mcdt = devm_kzalloc(&pdev->dev, sizeof(*mcdt), GFP_KERNEL); + if (!mcdt) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + mcdt->base = devm_ioremap_resource(&pdev->dev, res); + if (!mcdt->base) + return -ENOMEM; + + mcdt->dev = &pdev->dev; + spin_lock_init(&mcdt->lock); + platform_set_drvdata(pdev, mcdt); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "Failed to get MCDT interrupt\n"); + return irq; + } + + ret = devm_request_irq(&pdev->dev, irq, sprd_mcdt_irq_handler, + 0, "sprd-mcdt", mcdt); + if (ret) { + dev_err(&pdev->dev, "Failed to request MCDT IRQ\n"); + return ret; + } + + sprd_mcdt_init_chans(mcdt, res); + + return 0; +} + +static int sprd_mcdt_remove(struct platform_device *pdev) +{ + struct sprd_mcdt_chan *temp; + + mutex_lock(&sprd_mcdt_list_mutex); + + list_for_each_entry(temp, &sprd_mcdt_chan_list, list) + list_del(&temp->list); + + mutex_unlock(&sprd_mcdt_list_mutex); + + return 0; +} + +static const struct of_device_id sprd_mcdt_of_match[] = { + { .compatible = "sprd,sc9860-mcdt", }, + { } +}; +MODULE_DEVICE_TABLE(of, sprd_mcdt_of_match); + +static struct platform_driver sprd_mcdt_driver = { + .probe = sprd_mcdt_probe, + .remove = sprd_mcdt_remove, + .driver = { + .name = "sprd-mcdt", + .of_match_table = sprd_mcdt_of_match, + }, +}; + +module_platform_driver(sprd_mcdt_driver); + +MODULE_DESCRIPTION("Spreadtrum Multi-Channel Data Transfer Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/sprd/sprd-mcdt.h b/sound/soc/sprd/sprd-mcdt.h new file mode 100644 index 000000000000..9cc7e207ac76 --- /dev/null +++ b/sound/soc/sprd/sprd-mcdt.h @@ -0,0 +1,107 @@ +// SPDX-License-Identifier: GPL-2.0 + +#ifndef __SPRD_MCDT_H +#define __SPRD_MCDT_H + +enum sprd_mcdt_channel_type { + SPRD_MCDT_DAC_CHAN, + SPRD_MCDT_ADC_CHAN, + SPRD_MCDT_UNKNOWN_CHAN, +}; + +enum sprd_mcdt_dma_chan { + SPRD_MCDT_DMA_CH0, + SPRD_MCDT_DMA_CH1, + SPRD_MCDT_DMA_CH2, + SPRD_MCDT_DMA_CH3, + SPRD_MCDT_DMA_CH4, +}; + +struct sprd_mcdt_chan_callback { + void (*notify)(void *data); + void *data; +}; + +/** + * struct sprd_mcdt_chan - this struct represents a single channel instance + * @mcdt: the mcdt controller + * @id: channel id + * @fifo_phys: channel fifo physical address which is used for DMA transfer + * @type: channel type + * @cb: channel fifo interrupt's callback interface to notify the fifo events + * @dma_enable: indicate if use DMA mode to transfer data + * @int_enable: indicate if use interrupt mode to notify users to read or + * write data manually + * @list: used to link into the global list + * + * Note: users should not modify any members of this structure. + */ +struct sprd_mcdt_chan { + struct sprd_mcdt_dev *mcdt; + u8 id; + unsigned long fifo_phys; + enum sprd_mcdt_channel_type type; + enum sprd_mcdt_dma_chan dma_chan; + struct sprd_mcdt_chan_callback *cb; + bool dma_enable; + bool int_enable; + struct list_head list; +}; + +#ifdef CONFIG_SND_SOC_SPRD_MCDT +struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel, + enum sprd_mcdt_channel_type type); +void sprd_mcdt_free_chan(struct sprd_mcdt_chan *chan); + +int sprd_mcdt_chan_write(struct sprd_mcdt_chan *chan, char *tx_buf, u32 size); +int sprd_mcdt_chan_read(struct sprd_mcdt_chan *chan, char *rx_buf, u32 size); +int sprd_mcdt_chan_int_enable(struct sprd_mcdt_chan *chan, u32 water_mark, + struct sprd_mcdt_chan_callback *cb); +void sprd_mcdt_chan_int_disable(struct sprd_mcdt_chan *chan); + +int sprd_mcdt_chan_dma_enable(struct sprd_mcdt_chan *chan, + enum sprd_mcdt_dma_chan dma_chan, u32 water_mark); +void sprd_mcdt_chan_dma_disable(struct sprd_mcdt_chan *chan); + +#else + +struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel, + enum sprd_mcdt_channel_type type) +{ + return NULL; +} + +void sprd_mcdt_free_chan(struct sprd_mcdt_chan *chan) +{ } + +int sprd_mcdt_chan_write(struct sprd_mcdt_chan *chan, char *tx_buf, u32 size) +{ + return -EINVAL; +} + +int sprd_mcdt_chan_read(struct sprd_mcdt_chan *chan, char *rx_buf, u32 size) +{ + return 0; +} + +int sprd_mcdt_chan_int_enable(struct sprd_mcdt_chan *chan, u32 water_mark, + struct sprd_mcdt_chan_callback *cb) +{ + return -EINVAL; +} + +void sprd_mcdt_chan_int_disable(struct sprd_mcdt_chan *chan) +{ } + +int sprd_mcdt_chan_dma_enable(struct sprd_mcdt_chan *chan, + enum sprd_mcdt_dma_chan dma_chan, u32 water_mark) +{ + return -EINVAL; +} + +void sprd_mcdt_chan_dma_disable(struct sprd_mcdt_chan *chan) +{ } + +#endif + +#endif /* __SPRD_MCDT_H */ -- cgit v1.2.3-59-g8ed1b From ddb351145a967ee791a0fb0156852ec2fcb746ba Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Sun, 21 Apr 2019 19:39:08 +0000 Subject: ASoC: fsl_sai: Update is_slave_mode with correct value is_slave_mode defaults to false because sai structure that contains it is kzalloc'ed. Anyhow, if we decide to set the following configuration SAI slave -> SAI master, is_slave_mode will remain set on true although SAI being master it should be set to false. Fix this by updating is_slave_mode for each call of fsl_sai_set_dai_fmt. Signed-off-by: Daniel Baluta Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9e0872f73d..7549b74e464e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -268,12 +268,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_CBS_CFS: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + sai->is_slave_mode = false; break; case SND_SOC_DAIFMT_CBM_CFM: sai->is_slave_mode = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; + sai->is_slave_mode = false; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; -- cgit v1.2.3-59-g8ed1b From 812ad463e0894d78e93ab8017e0edbe69038c55c Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Sun, 21 Apr 2019 19:39:09 +0000 Subject: ASoC: fsl_sai: Add support for runtime pm Basically the same actions as for system PM, so make use of pm_runtime_force_suspend/pm_runtime_force_resume. Signed-off-by: Shengjiu Wang Signed-off-by: Daniel Baluta Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7549b74e464e..26c27dc0781f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -9,6 +9,7 @@ #include #include #include +#include #include #include #include @@ -901,6 +902,8 @@ static int fsl_sai_probe(struct platform_device *pdev) platform_set_drvdata(pdev, sai); + pm_runtime_enable(&pdev->dev); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, &fsl_sai_dai, 1); if (ret) @@ -912,6 +915,11 @@ static int fsl_sai_probe(struct platform_device *pdev) return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); } +static int fsl_sai_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); +} + static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, @@ -920,8 +928,8 @@ static const struct of_device_id fsl_sai_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -#ifdef CONFIG_PM_SLEEP -static int fsl_sai_suspend(struct device *dev) +#ifdef CONFIG_PM +static int fsl_sai_runtime_suspend(struct device *dev) { struct fsl_sai *sai = dev_get_drvdata(dev); @@ -931,7 +939,7 @@ static int fsl_sai_suspend(struct device *dev) return 0; } -static int fsl_sai_resume(struct device *dev) +static int fsl_sai_runtime_resume(struct device *dev) { struct fsl_sai *sai = dev_get_drvdata(dev); @@ -943,14 +951,18 @@ static int fsl_sai_resume(struct device *dev) regmap_write(sai->regmap, FSL_SAI_RCSR, 0); return regcache_sync(sai->regmap); } -#endif /* CONFIG_PM_SLEEP */ +#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_sai_pm_ops = { - SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) + SET_RUNTIME_PM_OPS(fsl_sai_runtime_suspend, + fsl_sai_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, + .remove = fsl_sai_remove, .driver = { .name = "fsl-sai", .pm = &fsl_sai_pm_ops, -- cgit v1.2.3-59-g8ed1b From 1fb1e93a1dc7c9673917a8081e144f48bde46495 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 19 Apr 2019 15:12:16 -0500 Subject: ASoC: Intel: bytcht_es8316: fix compilation warning Remove warning below, align with other machine drivers. bytcht_es8316.c:508:11: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] quirk = (int)dmi_id->driver_data; ^ Fixes: a8d218f4fe811 ('ASoC: Intel: bytcht_es8316: Add quirk for the Teclast X98+ II') Signed-off-by: Pierre-Louis Bossart Reviewed-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 38975827e276..e8c585ffd04d 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -61,7 +61,7 @@ enum { #define BYT_CHT_ES8316_MONO_SPEAKER BIT(17) #define BYT_CHT_ES8316_JD_INVERTED BIT(18) -static int quirk; +static unsigned long quirk; static int quirk_override = -1; module_param_named(quirk, quirk_override, int, 0444); @@ -505,7 +505,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* Check for BYTCR or other platform and setup quirks */ dmi_id = dmi_first_match(byt_cht_es8316_quirk_table); if (dmi_id) { - quirk = (int)dmi_id->driver_data; + quirk = (unsigned long)dmi_id->driver_data; } else if (x86_match_cpu(baytrail_cpu_ids) && mach->mach_params.acpi_ipc_irq_index == 0) { /* On BYTCR default to SSP0, internal-mic-in2-map, mono-spk */ @@ -517,7 +517,8 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) BYT_CHT_ES8316_MONO_SPEAKER; } if (quirk_override != -1) { - dev_info(dev, "Overriding quirk 0x%x => 0x%x\n", quirk, + dev_info(dev, "Overriding quirk 0x%x => 0x%x\n", + (unsigned int)quirk, quirk_override); quirk = quirk_override; } -- cgit v1.2.3-59-g8ed1b From 2fb2a19af17ac6da3c9b5c6ea4c51b7353290587 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 19 Apr 2019 15:12:17 -0500 Subject: ASoC: Intel: bytcr_rt5640: align quirk override handling As discussed on alsa-devel, a zero value is useful to get rid of all quirks. Set default to -1 and align types as done in other machine drivers. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index f9175cf6747e..dc22df9a99fb 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -98,8 +98,8 @@ struct byt_rt5640_private { static bool is_bytcr; static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN; -static unsigned int quirk_override; -module_param_named(quirk, quirk_override, uint, 0444); +static int quirk_override = -1; +module_param_named(quirk, quirk_override, int, 0444); MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) @@ -1254,7 +1254,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) dmi_id = dmi_first_match(byt_rt5640_quirk_table); if (dmi_id) byt_rt5640_quirk = (unsigned long)dmi_id->driver_data; - if (quirk_override) { + if (quirk_override != -1) { dev_info(&pdev->dev, "Overriding quirk 0x%x => 0x%x\n", (unsigned int)byt_rt5640_quirk, quirk_override); byt_rt5640_quirk = quirk_override; -- cgit v1.2.3-59-g8ed1b From fb45befa7ea07399372266f09e562b26db5693c9 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 19 Apr 2019 15:12:18 -0500 Subject: ASoC: Intel: bytcr_rt5651: align quirk override handling As discussed on alsa-devel, a zero value is useful to get rid of all quirks. Set default to -1 and align types as done in other machine drivers. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Hans de Goede Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5651.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 21c6675abd19..ca657c3e5726 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -102,8 +102,8 @@ static const struct acpi_gpio_mapping *byt_rt5651_gpios; static unsigned long byt_rt5651_quirk = BYT_RT5651_DEFAULT_QUIRKS | BYT_RT5651_IN2_MAP; -static unsigned int quirk_override; -module_param_named(quirk, quirk_override, uint, 0444); +static int quirk_override = -1; +module_param_named(quirk, quirk_override, int, 0444); MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) @@ -987,7 +987,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* check quirks before creating card */ dmi_check_system(byt_rt5651_quirk_table); - if (quirk_override) { + if (quirk_override != -1) { dev_info(&pdev->dev, "Overriding quirk 0x%x => 0x%x\n", (unsigned int)byt_rt5651_quirk, quirk_override); byt_rt5651_quirk = quirk_override; -- cgit v1.2.3-59-g8ed1b From 780f202f2bf48aaa09282efc4dd6c39a146cc1f3 Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang Date: Fri, 26 Apr 2019 10:09:43 +0800 Subject: ASoC: mediatek: btcvsd fix irq mask disable = 1 enable = 0 prevent AP wake up, when not AP BTSCO scenario Signed-off-by: KaiChieh Chuang Signed-off-by: Mark Brown --- sound/soc/mediatek/common/mtk-btcvsd.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index 9a163d7064d1..bd55c546e790 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -193,13 +193,13 @@ static const u8 table_msbc_silence[SCO_PACKET_180] = { static void mtk_btcvsd_snd_irq_enable(struct mtk_btcvsd_snd *bt) { regmap_update_bits(bt->infra, bt->infra_misc_offset, - bt->conn_bt_cvsd_mask, bt->conn_bt_cvsd_mask); + bt->conn_bt_cvsd_mask, 0); } static void mtk_btcvsd_snd_irq_disable(struct mtk_btcvsd_snd *bt) { regmap_update_bits(bt->infra, bt->infra_misc_offset, - bt->conn_bt_cvsd_mask, 0); + bt->conn_bt_cvsd_mask, bt->conn_bt_cvsd_mask); } static void mtk_btcvsd_snd_set_state(struct mtk_btcvsd_snd *bt, -- cgit v1.2.3-59-g8ed1b From 90194281ee8b01fc5a223a12ac90006ae98862c4 Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 26 Apr 2019 04:25:49 +0200 Subject: ASoC: simple-card-utils: add asoc_simple_parse_pin_switches() This function is a helper that permits to create pin switch controls for a list of widgets whose names are listed in the PREFIX "pin-switches" devicetree property. Signed-off-by: Paul Cercueil Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 57 +++++++++++++++++++++++++++++++++++ 2 files changed, 59 insertions(+) (limited to 'sound') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 67dc3ee6ed01..3429888347e7 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -133,6 +133,8 @@ int asoc_simple_parse_routing(struct snd_soc_card *card, char *prefix); int asoc_simple_parse_widgets(struct snd_soc_card *card, char *prefix); +int asoc_simple_parse_pin_switches(struct snd_soc_card *card, + char *prefix); int asoc_simple_init_jack(struct snd_soc_card *card, struct asoc_simple_jack *sjack, diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index db1458a19985..f4c6375d11c7 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -430,6 +430,63 @@ int asoc_simple_parse_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_parse_widgets); +int asoc_simple_parse_pin_switches(struct snd_soc_card *card, + char *prefix) +{ + const unsigned int nb_controls_max = 16; + const char **strings, *control_name; + struct snd_kcontrol_new *controls; + struct device *dev = card->dev; + unsigned int i, nb_controls; + char prop[128]; + int ret; + + if (!prefix) + prefix = ""; + + snprintf(prop, sizeof(prop), "%s%s", prefix, "pin-switches"); + + if (!of_property_read_bool(dev->of_node, prop)) + return 0; + + strings = devm_kcalloc(dev, nb_controls_max, + sizeof(*strings), GFP_KERNEL); + if (!strings) + return -ENOMEM; + + ret = of_property_read_string_array(dev->of_node, prop, + strings, nb_controls_max); + if (ret < 0) + return ret; + + nb_controls = (unsigned int)ret; + + controls = devm_kcalloc(dev, nb_controls, + sizeof(*controls), GFP_KERNEL); + if (!controls) + return -ENOMEM; + + for (i = 0; i < nb_controls; i++) { + control_name = devm_kasprintf(dev, GFP_KERNEL, + "%s Switch", strings[i]); + if (!control_name) + return -ENOMEM; + + controls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + controls[i].name = control_name; + controls[i].info = snd_soc_dapm_info_pin_switch; + controls[i].get = snd_soc_dapm_get_pin_switch; + controls[i].put = snd_soc_dapm_put_pin_switch; + controls[i].private_value = (unsigned long)strings[i]; + } + + card->controls = controls; + card->num_controls = nb_controls; + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_parse_pin_switches); + int asoc_simple_init_jack(struct snd_soc_card *card, struct asoc_simple_jack *sjack, int is_hp, char *prefix) -- cgit v1.2.3-59-g8ed1b From b46ea8f4d23807e41be86bf214618e4fd4682b9b Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 25 Apr 2019 20:20:50 -0700 Subject: ASoC: fsl_sai: Add missing return 0 in remove() Build warning being reported: sound/soc/fsl/fsl_sai.c: In function 'fsl_sai_remove': sound/soc/fsl/fsl_sai.c:921:1: warning: no return statement in function returning non-void [-Wreturn-type] So this patch just adds a "return 0" to fix it. Fixes: 812ad463e089 ("ASoC: fsl_sai: Add support for runtime pm") Reported-by: Stephen Rothwell Signed-off-by: Nicolin Chen Reviewed-by: Daniel Baluta Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 26c27dc0781f..8593269156bd 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -918,6 +918,8 @@ static int fsl_sai_probe(struct platform_device *pdev) static int fsl_sai_remove(struct platform_device *pdev) { pm_runtime_disable(&pdev->dev); + + return 0; } static const struct of_device_id fsl_sai_ids[] = { -- cgit v1.2.3-59-g8ed1b From 552f2d7edd38e29073f8036191320f09b534c480 Mon Sep 17 00:00:00 2001 From: Paul Cercueil Date: Fri, 26 Apr 2019 04:25:50 +0200 Subject: ASoC: simple-card: Read pin switches conf from devicetree When the routing path between a widget (e.g. "Speaker") and the codec goes through an external amplifier, having a pin switch for this widget allows the amplifier to be disabled when the widget is not to be used (e.g. when using headphones). Signed-off-by: Paul Cercueil Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 335ead0cbb8a..9b568f578bcd 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -464,6 +464,10 @@ static int simple_parse_of(struct asoc_simple_priv *priv) if (ret < 0) return ret; + ret = asoc_simple_parse_pin_switches(card, PREFIX); + if (ret < 0) + return ret; + /* Single/Muti DAI link(s) & New style of DT node */ memset(&li, 0, sizeof(li)); for (li.cpu = 1; li.cpu >= 0; li.cpu--) { -- cgit v1.2.3-59-g8ed1b From 8f8a54884dfd3d756693b7f174a06397bf19d8a5 Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen Date: Tue, 23 Apr 2019 17:58:08 +0200 Subject: ASoC: stm32: sai: simplify dai driver initialisation Suppress the useless dynamic allocation of the dai driver structure. Signed-off-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 43 +++++++++---------------------------------- 1 file changed, 9 insertions(+), 34 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3dd54bc54fa1..e3b021c9e8d0 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -109,7 +109,7 @@ struct stm32_sai_sub_data { struct regmap *regmap; const struct regmap_config *regmap_config; struct snd_dmaengine_dai_dma_data dma_params; - struct snd_soc_dai_driver *cpu_dai_drv; + struct snd_soc_dai_driver cpu_dai_drv; struct snd_soc_dai *cpu_dai; struct snd_pcm_substream *substream; struct stm32_sai_data *pdata; @@ -1204,8 +1204,7 @@ static const struct snd_pcm_hardware stm32_sai_pcm_hw = { .periods_max = 8, }; -static struct snd_soc_dai_driver stm32_sai_playback_dai[] = { -{ +static struct snd_soc_dai_driver stm32_sai_playback_dai = { .probe = stm32_sai_dai_probe, .pcm_new = stm32_sai_pcm_new, .id = 1, /* avoid call to fmt_single_name() */ @@ -1222,11 +1221,9 @@ static struct snd_soc_dai_driver stm32_sai_playback_dai[] = { SNDRV_PCM_FMTBIT_S32_LE, }, .ops = &stm32_sai_pcm_dai_ops, - } }; -static struct snd_soc_dai_driver stm32_sai_capture_dai[] = { -{ +static struct snd_soc_dai_driver stm32_sai_capture_dai = { .probe = stm32_sai_dai_probe, .id = 1, /* avoid call to fmt_single_name() */ .capture = { @@ -1242,7 +1239,6 @@ static struct snd_soc_dai_driver stm32_sai_capture_dai[] = { SNDRV_PCM_FMTBIT_S32_LE, }, .ops = &stm32_sai_pcm_dai_ops, - } }; static const struct snd_dmaengine_pcm_config stm32_sai_pcm_config = { @@ -1411,29 +1407,6 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, return 0; } -static int stm32_sai_sub_dais_init(struct platform_device *pdev, - struct stm32_sai_sub_data *sai) -{ - sai->cpu_dai_drv = devm_kzalloc(&pdev->dev, - sizeof(struct snd_soc_dai_driver), - GFP_KERNEL); - if (!sai->cpu_dai_drv) - return -ENOMEM; - - if (STM_SAI_IS_PLAYBACK(sai)) { - memcpy(sai->cpu_dai_drv, &stm32_sai_playback_dai, - sizeof(stm32_sai_playback_dai)); - sai->cpu_dai_drv->playback.stream_name = sai->cpu_dai_drv->name; - } else { - memcpy(sai->cpu_dai_drv, &stm32_sai_capture_dai, - sizeof(stm32_sai_capture_dai)); - sai->cpu_dai_drv->capture.stream_name = sai->cpu_dai_drv->name; - } - sai->cpu_dai_drv->name = dev_name(&pdev->dev); - - return 0; -} - static int stm32_sai_sub_probe(struct platform_device *pdev) { struct stm32_sai_sub_data *sai; @@ -1465,9 +1438,11 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) if (ret) return ret; - ret = stm32_sai_sub_dais_init(pdev, sai); - if (ret) - return ret; + if (STM_SAI_IS_PLAYBACK(sai)) + sai->cpu_dai_drv = stm32_sai_playback_dai; + else + sai->cpu_dai_drv = stm32_sai_capture_dai; + sai->cpu_dai_drv.name = dev_name(&pdev->dev); ret = devm_request_irq(&pdev->dev, sai->pdata->irq, stm32_sai_isr, IRQF_SHARED, dev_name(&pdev->dev), sai); @@ -1477,7 +1452,7 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) } ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - sai->cpu_dai_drv, 1); + &sai->cpu_dai_drv, 1); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From c16211d6226dbde8819b84da07508083a1138a06 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:06 -0500 Subject: ASoC: SOF: Add Sound Open Firmware driver core The Sound Open Firmware driver core is a generic architecture independent layer that allows SOF to be used on many different architectures and platforms. It abstracts DSP operations and IO methods so that the target DSP can be an internal memory mapped or external SPI or I2C based device. This abstraction also allows SOF to be run on many different VMs on the same physical HW. SOF also requires some data in ASoC PCM runtime data for looking up SOF data during ASoC PCM operations. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/sof.h | 94 +++++++ sound/soc/sof/core.c | 506 +++++++++++++++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 632 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1232 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/sof-priv.h (limited to 'sound') diff --git a/include/sound/sof.h b/include/sound/sof.h new file mode 100644 index 000000000000..54f65ec33a6c --- /dev/null +++ b/include/sound/sof.h @@ -0,0 +1,94 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + * + * Author: Liam Girdwood + */ + +#ifndef __INCLUDE_SOUND_SOF_H +#define __INCLUDE_SOUND_SOF_H + +#include +#include + +struct snd_sof_dsp_ops; + +/* + * SOF Platform data. + */ +struct snd_sof_pdata { + const struct firmware *fw; + const char *drv_name; + const char *name; + const char *platform; + + struct device *dev; + + /* + * notification callback used if the hardware initialization + * can take time or is handled in a workqueue. This callback + * can be used by the caller to e.g. enable runtime_pm + * or limit functionality until all low-level inits are + * complete. + */ + void (*sof_probe_complete)(struct device *dev); + + /* descriptor */ + const struct sof_dev_desc *desc; + + /* firmware and topology filenames */ + const char *fw_filename_prefix; + const char *fw_filename; + const char *tplg_filename_prefix; + const char *tplg_filename; + + /* machine */ + struct platform_device *pdev_mach; + const struct snd_soc_acpi_mach *machine; + + void *hw_pdata; +}; + +/* + * Descriptor used for setting up SOF platform data. This is used when + * ACPI/PCI data is missing or mapped differently. + */ +struct sof_dev_desc { + /* list of machines using this configuration */ + struct snd_soc_acpi_mach *machines; + + /* Platform resource indexes in BAR / ACPI resources. */ + /* Must set to -1 if not used - add new items to end */ + int resindex_lpe_base; + int resindex_pcicfg_base; + int resindex_imr_base; + int irqindex_host_ipc; + int resindex_dma_base; + + /* DMA only valid when resindex_dma_base != -1*/ + int dma_engine; + int dma_size; + + /* IPC timeouts in ms */ + int ipc_timeout; + int boot_timeout; + + /* chip information for dsp */ + const void *chip_info; + + /* defaults for no codec mode */ + const char *nocodec_fw_filename; + const char *nocodec_tplg_filename; + + /* defaults paths for firmware and topology files */ + const char *default_fw_path; + const char *default_tplg_path; + + const struct snd_sof_dsp_ops *ops; + const struct sof_arch_ops *arch_ops; +}; + +#endif diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c new file mode 100644 index 000000000000..39cbd84ff9c8 --- /dev/null +++ b/sound/soc/sof/core.c @@ -0,0 +1,506 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +#include +#include +#include +#include +#include +#include "sof-priv.h" +#include "ops.h" + +/* SOF defaults if not provided by the platform in ms */ +#define TIMEOUT_DEFAULT_IPC_MS 5 +#define TIMEOUT_DEFAULT_BOOT_MS 100 + +/* + * Generic object lookup APIs. + */ + +struct snd_sof_pcm *snd_sof_find_spcm_name(struct snd_sof_dev *sdev, + const char *name) +{ + struct snd_sof_pcm *spcm; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + /* match with PCM dai name */ + if (strcmp(spcm->pcm.dai_name, name) == 0) + return spcm; + + /* match with playback caps name if set */ + if (*spcm->pcm.caps[0].name && + !strcmp(spcm->pcm.caps[0].name, name)) + return spcm; + + /* match with capture caps name if set */ + if (*spcm->pcm.caps[1].name && + !strcmp(spcm->pcm.caps[1].name, name)) + return spcm; + } + + return NULL; +} + +struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_sof_dev *sdev, + unsigned int comp_id, + int *direction) +{ + struct snd_sof_pcm *spcm; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + if (spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].comp_id == comp_id) { + *direction = SNDRV_PCM_STREAM_PLAYBACK; + return spcm; + } + if (spcm->stream[SNDRV_PCM_STREAM_CAPTURE].comp_id == comp_id) { + *direction = SNDRV_PCM_STREAM_CAPTURE; + return spcm; + } + } + + return NULL; +} + +struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_sof_dev *sdev, + unsigned int pcm_id) +{ + struct snd_sof_pcm *spcm; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + if (le32_to_cpu(spcm->pcm.pcm_id) == pcm_id) + return spcm; + } + + return NULL; +} + +struct snd_sof_widget *snd_sof_find_swidget(struct snd_sof_dev *sdev, + const char *name) +{ + struct snd_sof_widget *swidget; + + list_for_each_entry(swidget, &sdev->widget_list, list) { + if (strcmp(name, swidget->widget->name) == 0) + return swidget; + } + + return NULL; +} + +/* find widget by stream name and direction */ +struct snd_sof_widget *snd_sof_find_swidget_sname(struct snd_sof_dev *sdev, + const char *pcm_name, int dir) +{ + struct snd_sof_widget *swidget; + enum snd_soc_dapm_type type; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) + type = snd_soc_dapm_aif_in; + else + type = snd_soc_dapm_aif_out; + + list_for_each_entry(swidget, &sdev->widget_list, list) { + if (!strcmp(pcm_name, swidget->widget->sname) && swidget->id == type) + return swidget; + } + + return NULL; +} + +struct snd_sof_dai *snd_sof_find_dai(struct snd_sof_dev *sdev, + const char *name) +{ + struct snd_sof_dai *dai; + + list_for_each_entry(dai, &sdev->dai_list, list) { + if (dai->name && (strcmp(name, dai->name) == 0)) + return dai; + } + + return NULL; +} + +/* + * FW Panic/fault handling. + */ + +struct sof_panic_msg { + u32 id; + const char *msg; +}; + +/* standard FW panic types */ +static const struct sof_panic_msg panic_msg[] = { + {SOF_IPC_PANIC_MEM, "out of memory"}, + {SOF_IPC_PANIC_WORK, "work subsystem init failed"}, + {SOF_IPC_PANIC_IPC, "IPC subsystem init failed"}, + {SOF_IPC_PANIC_ARCH, "arch init failed"}, + {SOF_IPC_PANIC_PLATFORM, "platform init failed"}, + {SOF_IPC_PANIC_TASK, "scheduler init failed"}, + {SOF_IPC_PANIC_EXCEPTION, "runtime exception"}, + {SOF_IPC_PANIC_DEADLOCK, "deadlock"}, + {SOF_IPC_PANIC_STACK, "stack overflow"}, + {SOF_IPC_PANIC_IDLE, "can't enter idle"}, + {SOF_IPC_PANIC_WFI, "invalid wait state"}, + {SOF_IPC_PANIC_ASSERT, "assertion failed"}, +}; + +/* + * helper to be called from .dbg_dump callbacks. No error code is + * provided, it's left as an exercise for the caller of .dbg_dump + * (typically IPC or loader) + */ +void snd_sof_get_status(struct snd_sof_dev *sdev, u32 panic_code, + u32 tracep_code, void *oops, + struct sof_ipc_panic_info *panic_info, + void *stack, size_t stack_words) +{ + u32 code; + int i; + + /* is firmware dead ? */ + if ((panic_code & SOF_IPC_PANIC_MAGIC_MASK) != SOF_IPC_PANIC_MAGIC) { + dev_err(sdev->dev, "error: unexpected fault 0x%8.8x trace 0x%8.8x\n", + panic_code, tracep_code); + return; /* no fault ? */ + } + + code = panic_code & (SOF_IPC_PANIC_MAGIC_MASK | SOF_IPC_PANIC_CODE_MASK); + + for (i = 0; i < ARRAY_SIZE(panic_msg); i++) { + if (panic_msg[i].id == code) { + dev_err(sdev->dev, "error: %s\n", panic_msg[i].msg); + dev_err(sdev->dev, "error: trace point %8.8x\n", + tracep_code); + goto out; + } + } + + /* unknown error */ + dev_err(sdev->dev, "error: unknown reason %8.8x\n", panic_code); + dev_err(sdev->dev, "error: trace point %8.8x\n", tracep_code); + +out: + dev_err(sdev->dev, "error: panic at %s:%d\n", + panic_info->filename, panic_info->linenum); + sof_oops(sdev, oops); + sof_stack(sdev, oops, stack, stack_words); +} +EXPORT_SYMBOL(snd_sof_get_status); + +/* + * Generic buffer page table creation. + * Take the each physical page address and drop the least significant unused + * bits from each (based on PAGE_SIZE). Then pack valid page address bits + * into compressed page table. + */ + +int snd_sof_create_page_table(struct snd_sof_dev *sdev, + struct snd_dma_buffer *dmab, + unsigned char *page_table, size_t size) +{ + int i, pages; + + pages = snd_sgbuf_aligned_pages(size); + + dev_dbg(sdev->dev, "generating page table for %p size 0x%zx pages %d\n", + dmab->area, size, pages); + + for (i = 0; i < pages; i++) { + /* + * The number of valid address bits for each page is 20. + * idx determines the byte position within page_table + * where the current page's address is stored + * in the compressed page_table. + * This can be calculated by multiplying the page number by 2.5. + */ + u32 idx = (5 * i) >> 1; + u32 pfn = snd_sgbuf_get_addr(dmab, i * PAGE_SIZE) >> PAGE_SHIFT; + u8 *pg_table; + + dev_vdbg(sdev->dev, "pfn i %i idx %d pfn %x\n", i, idx, pfn); + + pg_table = (u8 *)(page_table + idx); + + /* + * pagetable compression: + * byte 0 byte 1 byte 2 byte 3 byte 4 byte 5 + * ___________pfn 0__________ __________pfn 1___________ _pfn 2... + * .... .... .... .... .... .... .... .... .... .... .... + * It is created by: + * 1. set current location to 0, PFN index i to 0 + * 2. put pfn[i] at current location in Little Endian byte order + * 3. calculate an intermediate value as + * x = (pfn[i+1] << 4) | (pfn[i] & 0xf) + * 4. put x at offset (current location + 2) in LE byte order + * 5. increment current location by 5 bytes, increment i by 2 + * 6. continue to (2) + */ + if (i & 1) + put_unaligned_le32((pg_table[0] & 0xf) | pfn << 4, + pg_table); + else + put_unaligned_le32(pfn, pg_table); + } + + return pages; +} + +/* + * SOF Driver enumeration. + */ +static int sof_machine_check(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *plat_data = sdev->pdata; + struct snd_soc_acpi_mach *machine; + int ret; + + if (plat_data->machine) + return 0; + + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC)) { + dev_err(sdev->dev, "error: no matching ASoC machine driver found - aborting probe\n"); + return -ENODEV; + } + + /* fallback to nocodec mode */ + dev_warn(sdev->dev, "No ASoC machine driver found - using nocodec\n"); + machine = devm_kzalloc(sdev->dev, sizeof(*machine), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + ret = sof_nocodec_setup(sdev->dev, plat_data, machine, + plat_data->desc, plat_data->desc->ops); + if (ret < 0) + return ret; + + plat_data->machine = machine; + + return 0; +} + +static int sof_probe_continue(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *plat_data = sdev->pdata; + const char *drv_name; + const void *mach; + int size; + int ret; + + /* probe the DSP hardware */ + ret = snd_sof_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to probe DSP %d\n", ret); + return ret; + } + + /* check machine info */ + ret = sof_machine_check(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to get machine info %d\n", + ret); + goto dbg_err; + } + + /* set up platform component driver */ + snd_sof_new_platform_drv(sdev); + + /* register any debug/trace capabilities */ + ret = snd_sof_dbg_init(sdev); + if (ret < 0) { + /* + * debugfs issues are suppressed in snd_sof_dbg_init() since + * we cannot rely on debugfs + * here we trap errors due to memory allocation only. + */ + dev_err(sdev->dev, "error: failed to init DSP trace/debug %d\n", + ret); + goto dbg_err; + } + + /* init the IPC */ + sdev->ipc = snd_sof_ipc_init(sdev); + if (!sdev->ipc) { + dev_err(sdev->dev, "error: failed to init DSP IPC %d\n", ret); + goto ipc_err; + } + + /* load the firmware */ + ret = snd_sof_load_firmware(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to load DSP firmware %d\n", + ret); + goto fw_load_err; + } + + /* boot the firmware */ + ret = snd_sof_run_firmware(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to boot DSP firmware %d\n", + ret); + goto fw_run_err; + } + + /* init DMA trace */ + ret = snd_sof_init_trace(sdev); + if (ret < 0) { + /* non fatal */ + dev_warn(sdev->dev, + "warning: failed to initialize trace %d\n", ret); + } + + /* hereafter all FW boot flows are for PM reasons */ + sdev->first_boot = false; + + /* now register audio DSP platform driver and dai */ + ret = devm_snd_soc_register_component(sdev->dev, &sdev->plat_drv, + sof_ops(sdev)->drv, + sof_ops(sdev)->num_drv); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to register DSP DAI driver %d\n", ret); + goto fw_run_err; + } + + drv_name = plat_data->machine->drv_name; + mach = (const void *)plat_data->machine; + size = sizeof(*plat_data->machine); + + /* register machine driver, pass machine info as pdata */ + plat_data->pdev_mach = + platform_device_register_data(sdev->dev, drv_name, + PLATFORM_DEVID_NONE, mach, size); + + if (IS_ERR(plat_data->pdev_mach)) { + ret = PTR_ERR(plat_data->pdev_mach); + goto comp_err; + } + + dev_dbg(sdev->dev, "created machine %s\n", + dev_name(&plat_data->pdev_mach->dev)); + + if (plat_data->sof_probe_complete) + plat_data->sof_probe_complete(sdev->dev); + + return 0; + +comp_err: + snd_soc_unregister_component(sdev->dev); +fw_run_err: + snd_sof_fw_unload(sdev); +fw_load_err: + snd_sof_ipc_free(sdev); +ipc_err: + snd_sof_free_debug(sdev); +dbg_err: + snd_sof_remove(sdev); + + return ret; +} + +static void sof_probe_work(struct work_struct *work) +{ + struct snd_sof_dev *sdev = + container_of(work, struct snd_sof_dev, probe_work); + int ret; + + ret = sof_probe_continue(sdev); + if (ret < 0) { + /* errors cannot be propagated, log */ + dev_err(sdev->dev, "error: %s failed err: %d\n", __func__, ret); + } +} + +int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) +{ + struct snd_sof_dev *sdev; + + sdev = devm_kzalloc(dev, sizeof(*sdev), GFP_KERNEL); + if (!sdev) + return -ENOMEM; + + /* initialize sof device */ + sdev->dev = dev; + + sdev->pdata = plat_data; + sdev->first_boot = true; + dev_set_drvdata(dev, sdev); + + /* check all mandatory ops */ + if (!sof_ops(sdev) || !sof_ops(sdev)->probe || !sof_ops(sdev)->run || + !sof_ops(sdev)->block_read || !sof_ops(sdev)->block_write || + !sof_ops(sdev)->send_msg || !sof_ops(sdev)->load_firmware || + !sof_ops(sdev)->ipc_msg_data || !sof_ops(sdev)->ipc_pcm_params) + return -EINVAL; + + INIT_LIST_HEAD(&sdev->pcm_list); + INIT_LIST_HEAD(&sdev->kcontrol_list); + INIT_LIST_HEAD(&sdev->widget_list); + INIT_LIST_HEAD(&sdev->dai_list); + INIT_LIST_HEAD(&sdev->route_list); + spin_lock_init(&sdev->ipc_lock); + spin_lock_init(&sdev->hw_lock); + + if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) + INIT_WORK(&sdev->probe_work, sof_probe_work); + + /* set default timeouts if none provided */ + if (plat_data->desc->ipc_timeout == 0) + sdev->ipc_timeout = TIMEOUT_DEFAULT_IPC_MS; + else + sdev->ipc_timeout = plat_data->desc->ipc_timeout; + if (plat_data->desc->boot_timeout == 0) + sdev->boot_timeout = TIMEOUT_DEFAULT_BOOT_MS; + else + sdev->boot_timeout = plat_data->desc->boot_timeout; + + if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) { + schedule_work(&sdev->probe_work); + return 0; + } + + return sof_probe_continue(sdev); +} +EXPORT_SYMBOL(snd_sof_device_probe); + +int snd_sof_device_remove(struct device *dev) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_sof_pdata *pdata = sdev->pdata; + + if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) + cancel_work_sync(&sdev->probe_work); + + snd_sof_fw_unload(sdev); + snd_sof_ipc_free(sdev); + snd_sof_free_debug(sdev); + snd_sof_free_trace(sdev); + snd_sof_remove(sdev); + + /* + * Unregister machine driver. This will unbind the snd_card which + * will remove the component driver and unload the topology + * before freeing the snd_card. + */ + if (!IS_ERR_OR_NULL(pdata->pdev_mach)) + platform_device_unregister(pdata->pdev_mach); + + /* release firmware */ + release_firmware(pdata->fw); + pdata->fw = NULL; + + return 0; +} +EXPORT_SYMBOL(snd_sof_device_remove); + +MODULE_AUTHOR("Liam Girdwood"); +MODULE_DESCRIPTION("Sound Open Firmware (SOF) Core"); +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_ALIAS("platform:sof-audio"); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h new file mode 100644 index 000000000000..35e78ffecce2 --- /dev/null +++ b/sound/soc/sof/sof-priv.h @@ -0,0 +1,632 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + * + * Author: Liam Girdwood + */ + +#ifndef __SOUND_SOC_SOF_PRIV_H +#define __SOUND_SOC_SOF_PRIV_H + +#include + +#include +#include + +#include +#include /* needs to be included before control.h */ +#include +#include +#include +#include +#include +#include + +#include + +/* debug flags */ +#define SOF_DBG_REGS BIT(1) +#define SOF_DBG_MBOX BIT(2) +#define SOF_DBG_TEXT BIT(3) +#define SOF_DBG_PCI BIT(4) + +/* max BARs mmaped devices can use */ +#define SND_SOF_BARS 8 + +/* time in ms for runtime suspend delay */ +#define SND_SOF_SUSPEND_DELAY_MS 2000 + +/* DMA buffer size for trace */ +#define DMA_BUF_SIZE_FOR_TRACE (PAGE_SIZE * 16) + +/* max number of FE PCMs before BEs */ +#define SOF_BE_PCM_BASE 16 + +#define SOF_IPC_DSP_REPLY 0 +#define SOF_IPC_HOST_REPLY 1 + +/* convenience constructor for DAI driver streams */ +#define SOF_DAI_STREAM(sname, scmin, scmax, srates, sfmt) \ + {.stream_name = sname, .channels_min = scmin, .channels_max = scmax, \ + .rates = srates, .formats = sfmt} + +#define SOF_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_FLOAT) + +struct snd_sof_dev; +struct snd_sof_ipc_msg; +struct snd_sof_ipc; +struct snd_sof_debugfs_map; +struct snd_soc_tplg_ops; +struct snd_soc_component; +struct snd_sof_pdata; + +/* + * SOF DSP HW abstraction operations. + * Used to abstract DSP HW architecture and any IO busses between host CPU + * and DSP device(s). + */ +struct snd_sof_dsp_ops { + + /* probe and remove */ + int (*probe)(struct snd_sof_dev *sof_dev); /* mandatory */ + int (*remove)(struct snd_sof_dev *sof_dev); /* optional */ + + /* DSP core boot / reset */ + int (*run)(struct snd_sof_dev *sof_dev); /* mandatory */ + int (*stall)(struct snd_sof_dev *sof_dev); /* optional */ + int (*reset)(struct snd_sof_dev *sof_dev); /* optional */ + int (*core_power_up)(struct snd_sof_dev *sof_dev, + unsigned int core_mask); /* optional */ + int (*core_power_down)(struct snd_sof_dev *sof_dev, + unsigned int core_mask); /* optional */ + + /* + * Register IO: only used by respective drivers themselves, + * TODO: consider removing these operations and calling respective + * implementations directly + */ + void (*write)(struct snd_sof_dev *sof_dev, void __iomem *addr, + u32 value); /* optional */ + u32 (*read)(struct snd_sof_dev *sof_dev, + void __iomem *addr); /* optional */ + void (*write64)(struct snd_sof_dev *sof_dev, void __iomem *addr, + u64 value); /* optional */ + u64 (*read64)(struct snd_sof_dev *sof_dev, + void __iomem *addr); /* optional */ + + /* memcpy IO */ + void (*block_read)(struct snd_sof_dev *sof_dev, u32 bar, + u32 offset, void *dest, + size_t size); /* mandatory */ + void (*block_write)(struct snd_sof_dev *sof_dev, u32 bar, + u32 offset, void *src, + size_t size); /* mandatory */ + + /* doorbell */ + irqreturn_t (*irq_handler)(int irq, void *context); /* optional */ + irqreturn_t (*irq_thread)(int irq, void *context); /* optional */ + + /* ipc */ + int (*send_msg)(struct snd_sof_dev *sof_dev, + struct snd_sof_ipc_msg *msg); /* mandatory */ + + /* FW loading */ + int (*load_firmware)(struct snd_sof_dev *sof_dev); /* mandatory */ + int (*load_module)(struct snd_sof_dev *sof_dev, + struct snd_sof_mod_hdr *hdr); /* optional */ + /* + * FW ready checks for ABI compatibility and creates + * memory windows at first boot + */ + int (*fw_ready)(struct snd_sof_dev *sdev, u32 msg_id); /* optional */ + + /* connect pcm substream to a host stream */ + int (*pcm_open)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream); /* optional */ + /* disconnect pcm substream to a host stream */ + int (*pcm_close)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream); /* optional */ + + /* host stream hw params */ + int (*pcm_hw_params)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct sof_ipc_stream_params *ipc_params); /* optional */ + + /* host stream trigger */ + int (*pcm_trigger)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + int cmd); /* optional */ + + /* host stream pointer */ + snd_pcm_uframes_t (*pcm_pointer)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream); /* optional */ + + /* host read DSP stream data */ + void (*ipc_msg_data)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz); /* mandatory */ + + /* host configure DSP HW parameters */ + int (*ipc_pcm_params)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + const struct sof_ipc_pcm_params_reply *reply); /* mandatory */ + + /* pre/post firmware run */ + int (*pre_fw_run)(struct snd_sof_dev *sof_dev); /* optional */ + int (*post_fw_run)(struct snd_sof_dev *sof_dev); /* optional */ + + /* DSP PM */ + int (*suspend)(struct snd_sof_dev *sof_dev, int state); /* optional */ + int (*resume)(struct snd_sof_dev *sof_dev); /* optional */ + int (*runtime_suspend)(struct snd_sof_dev *sof_dev, + int state); /* optional */ + int (*runtime_resume)(struct snd_sof_dev *sof_dev); /* optional */ + + /* DSP clocking */ + int (*set_clk)(struct snd_sof_dev *sof_dev, u32 freq); /* optional */ + + /* debug */ + const struct snd_sof_debugfs_map *debug_map; /* optional */ + int debug_map_count; /* optional */ + void (*dbg_dump)(struct snd_sof_dev *sof_dev, + u32 flags); /* optional */ + + /* host DMA trace initialization */ + int (*trace_init)(struct snd_sof_dev *sdev, + u32 *stream_tag); /* optional */ + int (*trace_release)(struct snd_sof_dev *sdev); /* optional */ + int (*trace_trigger)(struct snd_sof_dev *sdev, + int cmd); /* optional */ + + /* DAI ops */ + struct snd_soc_dai_driver *drv; + int num_drv; +}; + +/* DSP architecture specific callbacks for oops and stack dumps */ +struct sof_arch_ops { + void (*dsp_oops)(struct snd_sof_dev *sdev, void *oops); + void (*dsp_stack)(struct snd_sof_dev *sdev, void *oops, + u32 *stack, u32 stack_words); +}; + +#define sof_arch_ops(sdev) ((sdev)->pdata->desc->arch_ops) + +/* DSP device HW descriptor mapping between bus ID and ops */ +struct sof_ops_table { + const struct sof_dev_desc *desc; + const struct snd_sof_dsp_ops *ops; +}; + +enum sof_dfsentry_type { + SOF_DFSENTRY_TYPE_IOMEM = 0, + SOF_DFSENTRY_TYPE_BUF, +}; + +enum sof_debugfs_access_type { + SOF_DEBUGFS_ACCESS_ALWAYS = 0, + SOF_DEBUGFS_ACCESS_D0_ONLY, +}; + +/* FS entry for debug files that can expose DSP memories, registers */ +struct snd_sof_dfsentry { + struct dentry *dfsentry; + size_t size; + enum sof_dfsentry_type type; + /* + * access_type specifies if the + * memory -> DSP resource (memory, register etc) is always accessible + * or if it is accessible only when the DSP is in D0. + */ + enum sof_debugfs_access_type access_type; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) + char *cache_buf; /* buffer to cache the contents of debugfs memory */ +#endif + struct snd_sof_dev *sdev; + struct list_head list; /* list in sdev dfsentry list */ + union { + void __iomem *io_mem; + void *buf; + }; +}; + +/* Debug mapping for any DSP memory or registers that can used for debug */ +struct snd_sof_debugfs_map { + const char *name; + u32 bar; + u32 offset; + u32 size; + /* + * access_type specifies if the memory is always accessible + * or if it is accessible only when the DSP is in D0. + */ + enum sof_debugfs_access_type access_type; +}; + +/* mailbox descriptor, used for host <-> DSP IPC */ +struct snd_sof_mailbox { + u32 offset; + size_t size; +}; + +/* IPC message descriptor for host <-> DSP IO */ +struct snd_sof_ipc_msg { + /* message data */ + u32 header; + void *msg_data; + void *reply_data; + size_t msg_size; + size_t reply_size; + int reply_error; + + wait_queue_head_t waitq; + bool ipc_complete; +}; + +/* PCM stream, mapped to FW component */ +struct snd_sof_pcm_stream { + u32 comp_id; + struct snd_dma_buffer page_table; + struct sof_ipc_stream_posn posn; + struct snd_pcm_substream *substream; +}; + +/* ALSA SOF PCM device */ +struct snd_sof_pcm { + struct snd_sof_dev *sdev; + struct snd_soc_tplg_pcm pcm; + struct snd_sof_pcm_stream stream[2]; + struct list_head list; /* list in sdev pcm list */ + struct snd_pcm_hw_params params[2]; + int hw_params_upon_resume[2]; /* set up hw_params upon resume */ +}; + +/* ALSA SOF Kcontrol device */ +struct snd_sof_control { + struct snd_sof_dev *sdev; + int comp_id; + int num_channels; + u32 readback_offset; /* offset to mmaped data if used */ + struct sof_ipc_ctrl_data *control_data; + u32 size; /* cdata size */ + enum sof_ipc_ctrl_cmd cmd; + u32 *volume_table; /* volume table computed from tlv data*/ + + struct list_head list; /* list in sdev control list */ +}; + +/* ASoC SOF DAPM widget */ +struct snd_sof_widget { + struct snd_sof_dev *sdev; + int comp_id; + int pipeline_id; + int complete; + int id; + + struct snd_soc_dapm_widget *widget; + struct list_head list; /* list in sdev widget list */ + + void *private; /* core does not touch this */ +}; + +/* ASoC SOF DAPM route */ +struct snd_sof_route { + struct snd_sof_dev *sdev; + + struct snd_soc_dapm_route *route; + struct list_head list; /* list in sdev route list */ + + void *private; +}; + +/* ASoC DAI device */ +struct snd_sof_dai { + struct snd_sof_dev *sdev; + const char *name; + + struct sof_ipc_comp_dai comp_dai; + struct sof_ipc_dai_config *dai_config; + struct list_head list; /* list in sdev dai list */ +}; + +/* + * SOF Device Level. + */ +struct snd_sof_dev { + struct device *dev; + spinlock_t ipc_lock; /* lock for IPC users */ + spinlock_t hw_lock; /* lock for HW IO access */ + + /* + * ASoC components. plat_drv fields are set dynamically so + * can't use const + */ + struct snd_soc_component_driver plat_drv; + + /* DSP firmware boot */ + wait_queue_head_t boot_wait; + u32 boot_complete; + u32 first_boot; + + /* work queue in case the probe is implemented in two steps */ + struct work_struct probe_work; + + /* DSP HW differentiation */ + struct snd_sof_pdata *pdata; + + /* IPC */ + struct snd_sof_ipc *ipc; + struct snd_sof_mailbox dsp_box; /* DSP initiated IPC */ + struct snd_sof_mailbox host_box; /* Host initiated IPC */ + struct snd_sof_mailbox stream_box; /* Stream position update */ + struct snd_sof_ipc_msg *msg; + u64 irq_status; + int ipc_irq; + u32 next_comp_id; /* monotonic - reset during S3 */ + + /* memory bases for mmaped DSPs - set by dsp_init() */ + void __iomem *bar[SND_SOF_BARS]; /* DSP base address */ + int mmio_bar; + int mailbox_bar; + size_t dsp_oops_offset; + + /* debug */ + struct dentry *debugfs_root; + struct list_head dfsentry_list; + + /* firmware loader */ + struct snd_dma_buffer dmab; + struct snd_dma_buffer dmab_bdl; + struct sof_ipc_fw_ready fw_ready; + struct sof_ipc_fw_version fw_version; + + /* topology */ + struct snd_soc_tplg_ops *tplg_ops; + struct list_head pcm_list; + struct list_head kcontrol_list; + struct list_head widget_list; + struct list_head dai_list; + struct list_head route_list; + struct snd_soc_component *component; + u32 enabled_cores_mask; /* keep track of enabled cores */ + + /* FW configuration */ + struct sof_ipc_dma_buffer_data *info_buffer; + struct sof_ipc_window *info_window; + + /* IPC timeouts in ms */ + int ipc_timeout; + int boot_timeout; + + /* Wait queue for code loading */ + wait_queue_head_t waitq; + int code_loading; + + /* DMA for Trace */ + struct snd_dma_buffer dmatb; + struct snd_dma_buffer dmatp; + int dma_trace_pages; + wait_queue_head_t trace_sleep; + u32 host_offset; + u32 dtrace_is_enabled; + u32 dtrace_error; + u32 msi_enabled; + + void *private; /* core does not touch this */ +}; + +/* + * Device Level. + */ + +int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data); +int snd_sof_device_remove(struct device *dev); + +int snd_sof_runtime_suspend(struct device *dev); +int snd_sof_runtime_resume(struct device *dev); +int snd_sof_resume(struct device *dev); +int snd_sof_suspend(struct device *dev); + +void snd_sof_new_platform_drv(struct snd_sof_dev *sdev); + +int snd_sof_create_page_table(struct snd_sof_dev *sdev, + struct snd_dma_buffer *dmab, + unsigned char *page_table, size_t size); + +/* + * Firmware loading. + */ +int snd_sof_load_firmware(struct snd_sof_dev *sdev); +int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev); +int snd_sof_load_firmware_memcpy(struct snd_sof_dev *sdev); +int snd_sof_run_firmware(struct snd_sof_dev *sdev); +int snd_sof_parse_module_memcpy(struct snd_sof_dev *sdev, + struct snd_sof_mod_hdr *module); +void snd_sof_fw_unload(struct snd_sof_dev *sdev); +int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset); + +/* + * IPC low level APIs. + */ +struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev); +void snd_sof_ipc_free(struct snd_sof_dev *sdev); +int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id); +void snd_sof_ipc_msgs_rx(struct snd_sof_dev *sdev); +int snd_sof_ipc_stream_pcm_params(struct snd_sof_dev *sdev, + struct sof_ipc_pcm_params *params); +int snd_sof_dsp_mailbox_init(struct snd_sof_dev *sdev, u32 dspbox, + size_t dspbox_size, u32 hostbox, + size_t hostbox_size); +int snd_sof_ipc_valid(struct snd_sof_dev *sdev); +int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header, + void *msg_data, size_t msg_bytes, void *reply_data, + size_t reply_bytes); +struct snd_sof_widget *snd_sof_find_swidget(struct snd_sof_dev *sdev, + const char *name); +struct snd_sof_widget *snd_sof_find_swidget_sname(struct snd_sof_dev *sdev, + const char *pcm_name, + int dir); +struct snd_sof_dai *snd_sof_find_dai(struct snd_sof_dev *sdev, + const char *name); + +static inline +struct snd_sof_pcm *snd_sof_find_spcm_dai(struct snd_sof_dev *sdev, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_sof_pcm *spcm = NULL; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + if (le32_to_cpu(spcm->pcm.dai_id) == rtd->dai_link->id) + return spcm; + } + + return NULL; +} + +struct snd_sof_pcm *snd_sof_find_spcm_name(struct snd_sof_dev *sdev, + const char *name); +struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_sof_dev *sdev, + unsigned int comp_id, + int *direction); +struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_sof_dev *sdev, + unsigned int pcm_id); + +/* + * Stream IPC + */ +int snd_sof_ipc_stream_posn(struct snd_sof_dev *sdev, + struct snd_sof_pcm *spcm, int direction, + struct sof_ipc_stream_posn *posn); + +/* + * Mixer IPC + */ +int snd_sof_ipc_set_get_comp_data(struct snd_sof_ipc *ipc, + struct snd_sof_control *scontrol, u32 ipc_cmd, + enum sof_ipc_ctrl_type ctrl_type, + enum sof_ipc_ctrl_cmd ctrl_cmd, + bool send); + +/* + * Topology. + * There is no snd_sof_free_topology since topology components will + * be freed by snd_soc_unregister_component, + */ +int snd_sof_init_topology(struct snd_sof_dev *sdev, + struct snd_soc_tplg_ops *ops); +int snd_sof_load_topology(struct snd_sof_dev *sdev, const char *file); +int snd_sof_complete_pipeline(struct snd_sof_dev *sdev, + struct snd_sof_widget *swidget); + +int sof_load_pipeline_ipc(struct snd_sof_dev *sdev, + struct sof_ipc_pipe_new *pipeline, + struct sof_ipc_comp_reply *r); + +/* + * Trace/debug + */ +int snd_sof_init_trace(struct snd_sof_dev *sdev); +void snd_sof_release_trace(struct snd_sof_dev *sdev); +void snd_sof_free_trace(struct snd_sof_dev *sdev); +int snd_sof_dbg_init(struct snd_sof_dev *sdev); +void snd_sof_free_debug(struct snd_sof_dev *sdev); +int snd_sof_debugfs_io_item(struct snd_sof_dev *sdev, + void __iomem *base, size_t size, + const char *name, + enum sof_debugfs_access_type access_type); +int snd_sof_debugfs_buf_item(struct snd_sof_dev *sdev, + void *base, size_t size, + const char *name); +int snd_sof_trace_update_pos(struct snd_sof_dev *sdev, + struct sof_ipc_dma_trace_posn *posn); +void snd_sof_trace_notify_for_error(struct snd_sof_dev *sdev); +void snd_sof_get_status(struct snd_sof_dev *sdev, u32 panic_code, + u32 tracep_code, void *oops, + struct sof_ipc_panic_info *panic_info, + void *stack, size_t stack_words); +int snd_sof_init_trace_ipc(struct snd_sof_dev *sdev); + +/* + * Platform specific ops. + */ +extern struct snd_compr_ops sof_compressed_ops; + +/* + * Kcontrols. + */ + +int snd_sof_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_bytes_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_bytes_ext_put(struct snd_kcontrol *kcontrol, + const unsigned int __user *binary_data, + unsigned int size); +int snd_sof_bytes_ext_get(struct snd_kcontrol *kcontrol, + unsigned int __user *binary_data, + unsigned int size); + +/* + * DSP Architectures. + */ +static inline void sof_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack, + u32 stack_words) +{ + if (sof_arch_ops(sdev)->dsp_stack) + sof_arch_ops(sdev)->dsp_stack(sdev, oops, stack, stack_words); +} + +static inline void sof_oops(struct snd_sof_dev *sdev, void *oops) +{ + if (sof_arch_ops(sdev)->dsp_oops) + sof_arch_ops(sdev)->dsp_oops(sdev, oops); +} + +extern const struct sof_arch_ops sof_xtensa_arch_ops; + +/* + * Utilities + */ +void sof_io_write(struct snd_sof_dev *sdev, void __iomem *addr, u32 value); +void sof_io_write64(struct snd_sof_dev *sdev, void __iomem *addr, u64 value); +u32 sof_io_read(struct snd_sof_dev *sdev, void __iomem *addr); +u64 sof_io_read64(struct snd_sof_dev *sdev, void __iomem *addr); +void sof_mailbox_write(struct snd_sof_dev *sdev, u32 offset, + void *message, size_t bytes); +void sof_mailbox_read(struct snd_sof_dev *sdev, u32 offset, + void *message, size_t bytes); +void sof_block_write(struct snd_sof_dev *sdev, u32 bar, u32 offset, void *src, + size_t size); +void sof_block_read(struct snd_sof_dev *sdev, u32 bar, u32 offset, void *dest, + size_t size); + +void intel_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz); +int intel_ipc_pcm_params(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + const struct sof_ipc_pcm_params_reply *reply); + +int intel_pcm_open(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream); +int intel_pcm_close(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream); + +#endif -- cgit v1.2.3-59-g8ed1b From c3078f5397046755ddcd0447d202124c398469c5 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:07 -0500 Subject: ASoC: SOF: Add Sound Open Firmware KControl support SOF exposes regular ALSA Kcontrols that are defined by topology. This patch converts the Kcontrol IO to DSP IPC. The current implementation is aligned with previous Intel solutions, but is not optimal and can be improved: a) for every get/put the host wakes up the DSP and generates an IPC. The kernel should cache the values and generate an IPC only when strictly necessary. b) the firmware can be implemented to only instantiate the pipelines and related control-related parts that are needed at a given time, and power-gate the relevant SRAM blocks. The development tasks for these two improvements has started, once validated they will be provided in an update. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/control.c | 552 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 552 insertions(+) create mode 100644 sound/soc/sof/control.c (limited to 'sound') diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c new file mode 100644 index 000000000000..11762c4580f1 --- /dev/null +++ b/sound/soc/sof/control.c @@ -0,0 +1,552 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +/* Mixer Controls */ + +#include +#include "sof-priv.h" + +static inline u32 mixer_to_ipc(unsigned int value, u32 *volume_map, int size) +{ + if (value >= size) + return volume_map[size - 1]; + + return volume_map[value]; +} + +static inline u32 ipc_to_mixer(u32 value, u32 *volume_map, int size) +{ + int i; + + for (i = 0; i < size; i++) { + if (volume_map[i] >= value) + return i; + } + + return i - 1; +} + +int snd_sof_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *sm = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_sof_control *scontrol = sm->dobj.private; + struct snd_sof_dev *sdev = scontrol->sdev; + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + unsigned int i, channels = scontrol->num_channels; + int err, ret; + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: volume get failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* get all the mixer data from DSP */ + snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + SOF_IPC_COMP_GET_VALUE, + SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_CMD_VOLUME, + false); + + /* read back each channel */ + for (i = 0; i < channels; i++) + ucontrol->value.integer.value[i] = + ipc_to_mixer(cdata->chanv[i].value, + scontrol->volume_table, sm->max + 1); + + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: volume get failed to idle %d\n", + err); + return 0; +} + +int snd_sof_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *sm = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_sof_control *scontrol = sm->dobj.private; + struct snd_sof_dev *sdev = scontrol->sdev; + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + unsigned int i, channels = scontrol->num_channels; + int ret, err; + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: volume put failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* update each channel */ + for (i = 0; i < channels; i++) { + cdata->chanv[i].value = + mixer_to_ipc(ucontrol->value.integer.value[i], + scontrol->volume_table, sm->max + 1); + cdata->chanv[i].channel = i; + } + + /* notify DSP of mixer updates */ + snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + SOF_IPC_COMP_SET_VALUE, + SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_CMD_VOLUME, + true); + + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: volume put failed to idle %d\n", + err); + return 0; +} + +int snd_sof_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *sm = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_sof_control *scontrol = sm->dobj.private; + struct snd_sof_dev *sdev = scontrol->sdev; + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + unsigned int i, channels = scontrol->num_channels; + int err, ret; + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: switch get failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* get all the mixer data from DSP */ + snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + SOF_IPC_COMP_GET_VALUE, + SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_CMD_SWITCH, + false); + + /* read back each channel */ + for (i = 0; i < channels; i++) + ucontrol->value.integer.value[i] = cdata->chanv[i].value; + + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: switch get failed to idle %d\n", + err); + return 0; +} + +int snd_sof_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *sm = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_sof_control *scontrol = sm->dobj.private; + struct snd_sof_dev *sdev = scontrol->sdev; + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + unsigned int i, channels = scontrol->num_channels; + int ret, err; + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: switch put failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* update each channel */ + for (i = 0; i < channels; i++) { + cdata->chanv[i].value = ucontrol->value.integer.value[i]; + cdata->chanv[i].channel = i; + } + + /* notify DSP of mixer updates */ + snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + SOF_IPC_COMP_SET_VALUE, + SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_CMD_SWITCH, + true); + + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: switch put failed to idle %d\n", + err); + return 0; +} + +int snd_sof_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *se = + (struct soc_enum *)kcontrol->private_value; + struct snd_sof_control *scontrol = se->dobj.private; + struct snd_sof_dev *sdev = scontrol->sdev; + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + unsigned int i, channels = scontrol->num_channels; + int err, ret; + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: enum get failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* get all the enum data from DSP */ + snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + SOF_IPC_COMP_GET_VALUE, + SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_CMD_ENUM, + false); + + /* read back each channel */ + for (i = 0; i < channels; i++) + ucontrol->value.enumerated.item[i] = cdata->chanv[i].value; + + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: enum get failed to idle %d\n", + err); + return 0; +} + +int snd_sof_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *se = + (struct soc_enum *)kcontrol->private_value; + struct snd_sof_control *scontrol = se->dobj.private; + struct snd_sof_dev *sdev = scontrol->sdev; + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + unsigned int i, channels = scontrol->num_channels; + int ret, err; + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: enum put failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* update each channel */ + for (i = 0; i < channels; i++) { + cdata->chanv[i].value = ucontrol->value.enumerated.item[i]; + cdata->chanv[i].channel = i; + } + + /* notify DSP of enum updates */ + snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + SOF_IPC_COMP_SET_VALUE, + SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_CMD_ENUM, + true); + + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: enum put failed to idle %d\n", + err); + return 0; +} + +int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_bytes_ext *be = + (struct soc_bytes_ext *)kcontrol->private_value; + struct snd_sof_control *scontrol = be->dobj.private; + struct snd_sof_dev *sdev = scontrol->sdev; + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + struct sof_abi_hdr *data = cdata->data; + size_t size; + int ret, err; + + if (be->max > sizeof(ucontrol->value.bytes.data)) { + dev_err_ratelimited(sdev->dev, + "error: data max %d exceeds ucontrol data array size\n", + be->max); + return -EINVAL; + } + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: bytes get failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* get all the binary data from DSP */ + snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + SOF_IPC_COMP_GET_DATA, + SOF_CTRL_TYPE_DATA_GET, + scontrol->cmd, + false); + + size = data->size + sizeof(*data); + if (size > be->max) { + dev_err_ratelimited(sdev->dev, + "error: DSP sent %zu bytes max is %d\n", + size, be->max); + ret = -EINVAL; + goto out; + } + + /* copy back to kcontrol */ + memcpy(ucontrol->value.bytes.data, data, size); + +out: + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: bytes get failed to idle %d\n", + err); + return ret; +} + +int snd_sof_bytes_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_bytes_ext *be = + (struct soc_bytes_ext *)kcontrol->private_value; + struct snd_sof_control *scontrol = be->dobj.private; + struct snd_sof_dev *sdev = scontrol->sdev; + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + struct sof_abi_hdr *data = cdata->data; + int ret, err; + + if (be->max > sizeof(ucontrol->value.bytes.data)) { + dev_err_ratelimited(sdev->dev, + "error: data max %d exceeds ucontrol data array size\n", + be->max); + return -EINVAL; + } + + if (data->size > be->max) { + dev_err_ratelimited(sdev->dev, + "error: size too big %d bytes max is %d\n", + data->size, be->max); + return -EINVAL; + } + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: bytes put failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* copy from kcontrol */ + memcpy(data, ucontrol->value.bytes.data, data->size); + + /* notify DSP of byte control updates */ + snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + SOF_IPC_COMP_SET_DATA, + SOF_CTRL_TYPE_DATA_SET, + scontrol->cmd, + true); + + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: bytes put failed to idle %d\n", + err); + return ret; +} + +int snd_sof_bytes_ext_put(struct snd_kcontrol *kcontrol, + const unsigned int __user *binary_data, + unsigned int size) +{ + struct soc_bytes_ext *be = + (struct soc_bytes_ext *)kcontrol->private_value; + struct snd_sof_control *scontrol = be->dobj.private; + struct snd_sof_dev *sdev = scontrol->sdev; + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + struct snd_ctl_tlv header; + const struct snd_ctl_tlv __user *tlvd = + (const struct snd_ctl_tlv __user *)binary_data; + int ret; + int err; + + /* + * The beginning of bytes data contains a header from where + * the length (as bytes) is needed to know the correct copy + * length of data from tlvd->tlv. + */ + if (copy_from_user(&header, tlvd, sizeof(const struct snd_ctl_tlv))) + return -EFAULT; + + /* be->max is coming from topology */ + if (header.length > be->max) { + dev_err_ratelimited(sdev->dev, "error: Bytes data size %d exceeds max %d.\n", + header.length, be->max); + return -EINVAL; + } + + /* Check that header id matches the command */ + if (header.numid != scontrol->cmd) { + dev_err_ratelimited(sdev->dev, + "error: incorrect numid %d\n", + header.numid); + return -EINVAL; + } + + if (copy_from_user(cdata->data, tlvd->tlv, header.length)) + return -EFAULT; + + if (cdata->data->magic != SOF_ABI_MAGIC) { + dev_err_ratelimited(sdev->dev, + "error: Wrong ABI magic 0x%08x.\n", + cdata->data->magic); + return -EINVAL; + } + + if (SOF_ABI_VERSION_INCOMPATIBLE(SOF_ABI_VERSION, cdata->data->abi)) { + dev_err_ratelimited(sdev->dev, "error: Incompatible ABI version 0x%08x.\n", + cdata->data->abi); + return -EINVAL; + } + + if (cdata->data->size + sizeof(const struct sof_abi_hdr) > be->max) { + dev_err_ratelimited(sdev->dev, "error: Mismatch in ABI data size (truncated?).\n"); + return -EINVAL; + } + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: bytes_ext put failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* notify DSP of byte control updates */ + snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + SOF_IPC_COMP_SET_DATA, + SOF_CTRL_TYPE_DATA_SET, + scontrol->cmd, + true); + + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: bytes_ext put failed to idle %d\n", + err); + + return ret; +} + +int snd_sof_bytes_ext_get(struct snd_kcontrol *kcontrol, + unsigned int __user *binary_data, + unsigned int size) +{ + struct soc_bytes_ext *be = + (struct soc_bytes_ext *)kcontrol->private_value; + struct snd_sof_control *scontrol = be->dobj.private; + struct snd_sof_dev *sdev = scontrol->sdev; + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + struct snd_ctl_tlv header; + struct snd_ctl_tlv __user *tlvd = + (struct snd_ctl_tlv __user *)binary_data; + int data_size; + int err; + int ret; + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: bytes_ext get failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* + * Decrement the limit by ext bytes header size to + * ensure the user space buffer is not exceeded. + */ + size -= sizeof(const struct snd_ctl_tlv); + + /* set the ABI header values */ + cdata->data->magic = SOF_ABI_MAGIC; + cdata->data->abi = SOF_ABI_VERSION; + + /* get all the component data from DSP */ + ret = snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + SOF_IPC_COMP_GET_DATA, + SOF_CTRL_TYPE_DATA_GET, + scontrol->cmd, + false); + + /* Prevent read of other kernel data or possibly corrupt response */ + data_size = cdata->data->size + sizeof(const struct sof_abi_hdr); + + /* check data size doesn't exceed max coming from topology */ + if (data_size > be->max) { + dev_err_ratelimited(sdev->dev, "error: user data size %d exceeds max size %d.\n", + data_size, be->max); + ret = -EINVAL; + goto out; + } + + header.numid = scontrol->cmd; + header.length = data_size; + if (copy_to_user(tlvd, &header, sizeof(const struct snd_ctl_tlv))) { + ret = -EFAULT; + goto out; + } + + if (copy_to_user(tlvd->tlv, cdata->data, data_size)) + ret = -EFAULT; + +out: + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: bytes_ext get failed to idle %d\n", + err); + return ret; +} -- cgit v1.2.3-59-g8ed1b From 86b02f71ccf320727473448733503e6f9378102d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:08 -0500 Subject: ASoC: SOF: Add driver debug support. Add debugFS files that can be used to expose DSP memories and and peripherals to userspace to assist with firmware debugging. Since we cannot rely on debugFS, errors are logged but don't stop execution. When a resource cannot be read in D3, it is optionally cached on suspend. Copying memories from IO will increase the suspend latency, this should only used in engineering builds w/ debug options. This part will have to be enhanced when support for D0ix states is provided, currently only D0 and D3 are supported. Signed-off-by: Pan Xiuli Signed-off-by: Ranjani Sridharan Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/debug.c | 232 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 232 insertions(+) create mode 100644 sound/soc/sof/debug.c (limited to 'sound') diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c new file mode 100644 index 000000000000..55f1d808dba0 --- /dev/null +++ b/sound/soc/sof/debug.c @@ -0,0 +1,232 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// +// Generic debug routines used to export DSP MMIO and memories to userspace +// for firmware debugging. +// + +#include +#include +#include +#include "sof-priv.h" +#include "ops.h" + +static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer, + size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + loff_t pos = *ppos; + size_t size_ret; + int skip = 0; + int size; + u8 *buf; + + size = dfse->size; + + /* validate position & count */ + if (pos < 0) + return -EINVAL; + if (pos >= size || !count) + return 0; + /* find the minimum. min() is not used since it adds sparse warnings */ + if (count > size - pos) + count = size - pos; + + /* align io read start to u32 multiple */ + pos = ALIGN_DOWN(pos, 4); + + /* intermediate buffer size must be u32 multiple */ + size = ALIGN(count, 4); + + /* if start position is unaligned, read extra u32 */ + if (unlikely(pos != *ppos)) { + skip = *ppos - pos; + if (pos + size + 4 < dfse->size) + size += 4; + } + + buf = kzalloc(size, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + if (dfse->type == SOF_DFSENTRY_TYPE_IOMEM) { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) + /* + * If the DSP is active: copy from IO. + * If the DSP is suspended: + * - Copy from IO if the memory is always accessible. + * - Otherwise, copy from cached buffer. + */ + if (pm_runtime_active(sdev->dev) || + dfse->access_type == SOF_DEBUGFS_ACCESS_ALWAYS) { + memcpy_fromio(buf, dfse->io_mem + pos, size); + } else { + dev_info(sdev->dev, + "Copying cached debugfs data\n"); + memcpy(buf, dfse->cache_buf + pos, size); + } +#else + /* if the DSP is in D3 */ + if (!pm_runtime_active(sdev->dev) && + dfse->access_type == SOF_DEBUGFS_ACCESS_D0_ONLY) { + dev_err(sdev->dev, + "error: debugfs entry %s cannot be read in DSP D3\n", + dfse->dfsentry->d_name.name); + kfree(buf); + return -EINVAL; + } + + memcpy_fromio(buf, dfse->io_mem + pos, size); +#endif + } else { + memcpy(buf, ((u8 *)(dfse->buf) + pos), size); + } + + /* copy to userspace */ + size_ret = copy_to_user(buffer, buf + skip, count); + + kfree(buf); + + /* update count & position if copy succeeded */ + if (size_ret) + return -EFAULT; + + *ppos = pos + count; + + return count; +} + +static const struct file_operations sof_dfs_fops = { + .open = simple_open, + .read = sof_dfsentry_read, + .llseek = default_llseek, +}; + +/* create FS entry for debug files that can expose DSP memories, registers */ +int snd_sof_debugfs_io_item(struct snd_sof_dev *sdev, + void __iomem *base, size_t size, + const char *name, + enum sof_debugfs_access_type access_type) +{ + struct snd_sof_dfsentry *dfse; + + if (!sdev) + return -EINVAL; + + dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL); + if (!dfse) + return -ENOMEM; + + dfse->type = SOF_DFSENTRY_TYPE_IOMEM; + dfse->io_mem = base; + dfse->size = size; + dfse->sdev = sdev; + dfse->access_type = access_type; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) + /* + * allocate cache buffer that will be used to save the mem window + * contents prior to suspend + */ + if (access_type == SOF_DEBUGFS_ACCESS_D0_ONLY) { + dfse->cache_buf = devm_kzalloc(sdev->dev, size, GFP_KERNEL); + if (!dfse->cache_buf) + return -ENOMEM; + } +#endif + + dfse->dfsentry = debugfs_create_file(name, 0444, sdev->debugfs_root, + dfse, &sof_dfs_fops); + if (!dfse->dfsentry) { + /* can't rely on debugfs, only log error and keep going */ + dev_err(sdev->dev, "error: cannot create debugfs entry %s\n", + name); + } else { + /* add to dfsentry list */ + list_add(&dfse->list, &sdev->dfsentry_list); + + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_sof_debugfs_io_item); + +/* create FS entry for debug files to expose kernel memory */ +int snd_sof_debugfs_buf_item(struct snd_sof_dev *sdev, + void *base, size_t size, + const char *name) +{ + struct snd_sof_dfsentry *dfse; + + if (!sdev) + return -EINVAL; + + dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL); + if (!dfse) + return -ENOMEM; + + dfse->type = SOF_DFSENTRY_TYPE_BUF; + dfse->buf = base; + dfse->size = size; + dfse->sdev = sdev; + + dfse->dfsentry = debugfs_create_file(name, 0444, sdev->debugfs_root, + dfse, &sof_dfs_fops); + if (!dfse->dfsentry) { + /* can't rely on debugfs, only log error and keep going */ + dev_err(sdev->dev, "error: cannot create debugfs entry %s\n", + name); + } else { + /* add to dfsentry list */ + list_add(&dfse->list, &sdev->dfsentry_list); + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_sof_debugfs_buf_item); + +int snd_sof_dbg_init(struct snd_sof_dev *sdev) +{ + const struct snd_sof_dsp_ops *ops = sof_ops(sdev); + const struct snd_sof_debugfs_map *map; + int i; + int err; + + /* use "sof" as top level debugFS dir */ + sdev->debugfs_root = debugfs_create_dir("sof", NULL); + if (IS_ERR_OR_NULL(sdev->debugfs_root)) { + dev_err(sdev->dev, "error: failed to create debugfs directory\n"); + return 0; + } + + /* init dfsentry list */ + INIT_LIST_HEAD(&sdev->dfsentry_list); + + /* create debugFS files for platform specific MMIO/DSP memories */ + for (i = 0; i < ops->debug_map_count; i++) { + map = &ops->debug_map[i]; + + err = snd_sof_debugfs_io_item(sdev, sdev->bar[map->bar] + + map->offset, map->size, + map->name, map->access_type); + /* errors are only due to memory allocation, not debugfs */ + if (err < 0) + return err; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_sof_dbg_init); + +void snd_sof_free_debug(struct snd_sof_dev *sdev) +{ + debugfs_remove_recursive(sdev->debugfs_root); +} +EXPORT_SYMBOL_GPL(snd_sof_free_debug); -- cgit v1.2.3-59-g8ed1b From 53e0c72d98ba3eae314c32476103eac47612aa58 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:09 -0500 Subject: ASoC: SOF: Add support for IPC IO between DSP and Host Define an IPC ABI for all host <--> DSP communication. This ABI should be transport agnostic. i.e. it should work on MMIO and SPI/I2C style interfaces. Signed-off-by: Guennadi Liakhovetski Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/sof/control.h | 158 ++++++++ include/sound/sof/dai-intel.h | 178 +++++++++ include/sound/sof/dai.h | 75 ++++ include/sound/sof/header.h | 158 ++++++++ include/sound/sof/info.h | 118 ++++++ include/sound/sof/pm.h | 48 +++ include/sound/sof/stream.h | 148 ++++++++ include/sound/sof/trace.h | 67 ++++ sound/soc/sof/ipc.c | 832 ++++++++++++++++++++++++++++++++++++++++++ 9 files changed, 1782 insertions(+) create mode 100644 include/sound/sof/control.h create mode 100644 include/sound/sof/dai-intel.h create mode 100644 include/sound/sof/dai.h create mode 100644 include/sound/sof/header.h create mode 100644 include/sound/sof/info.h create mode 100644 include/sound/sof/pm.h create mode 100644 include/sound/sof/stream.h create mode 100644 include/sound/sof/trace.h create mode 100644 sound/soc/sof/ipc.c (limited to 'sound') diff --git a/include/sound/sof/control.h b/include/sound/sof/control.h new file mode 100644 index 000000000000..bded69e696d4 --- /dev/null +++ b/include/sound/sof/control.h @@ -0,0 +1,158 @@ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_CONTROL_H__ +#define __INCLUDE_SOUND_SOF_CONTROL_H__ + +#include +#include + +/* + * Component Mixers and Controls + */ + +/* channel positions - uses same values as ALSA */ +enum sof_ipc_chmap { + SOF_CHMAP_UNKNOWN = 0, + SOF_CHMAP_NA, /**< N/A, silent */ + SOF_CHMAP_MONO, /**< mono stream */ + SOF_CHMAP_FL, /**< front left */ + SOF_CHMAP_FR, /**< front right */ + SOF_CHMAP_RL, /**< rear left */ + SOF_CHMAP_RR, /**< rear right */ + SOF_CHMAP_FC, /**< front centre */ + SOF_CHMAP_LFE, /**< LFE */ + SOF_CHMAP_SL, /**< side left */ + SOF_CHMAP_SR, /**< side right */ + SOF_CHMAP_RC, /**< rear centre */ + SOF_CHMAP_FLC, /**< front left centre */ + SOF_CHMAP_FRC, /**< front right centre */ + SOF_CHMAP_RLC, /**< rear left centre */ + SOF_CHMAP_RRC, /**< rear right centre */ + SOF_CHMAP_FLW, /**< front left wide */ + SOF_CHMAP_FRW, /**< front right wide */ + SOF_CHMAP_FLH, /**< front left high */ + SOF_CHMAP_FCH, /**< front centre high */ + SOF_CHMAP_FRH, /**< front right high */ + SOF_CHMAP_TC, /**< top centre */ + SOF_CHMAP_TFL, /**< top front left */ + SOF_CHMAP_TFR, /**< top front right */ + SOF_CHMAP_TFC, /**< top front centre */ + SOF_CHMAP_TRL, /**< top rear left */ + SOF_CHMAP_TRR, /**< top rear right */ + SOF_CHMAP_TRC, /**< top rear centre */ + SOF_CHMAP_TFLC, /**< top front left centre */ + SOF_CHMAP_TFRC, /**< top front right centre */ + SOF_CHMAP_TSL, /**< top side left */ + SOF_CHMAP_TSR, /**< top side right */ + SOF_CHMAP_LLFE, /**< left LFE */ + SOF_CHMAP_RLFE, /**< right LFE */ + SOF_CHMAP_BC, /**< bottom centre */ + SOF_CHMAP_BLC, /**< bottom left centre */ + SOF_CHMAP_BRC, /**< bottom right centre */ + SOF_CHMAP_LAST = SOF_CHMAP_BRC, +}; + +/* control data type and direction */ +enum sof_ipc_ctrl_type { + /* per channel data - uses struct sof_ipc_ctrl_value_chan */ + SOF_CTRL_TYPE_VALUE_CHAN_GET = 0, + SOF_CTRL_TYPE_VALUE_CHAN_SET, + /* component data - uses struct sof_ipc_ctrl_value_comp */ + SOF_CTRL_TYPE_VALUE_COMP_GET, + SOF_CTRL_TYPE_VALUE_COMP_SET, + /* bespoke data - uses struct sof_abi_hdr */ + SOF_CTRL_TYPE_DATA_GET, + SOF_CTRL_TYPE_DATA_SET, +}; + +/* control command type */ +enum sof_ipc_ctrl_cmd { + SOF_CTRL_CMD_VOLUME = 0, /**< maps to ALSA volume style controls */ + SOF_CTRL_CMD_ENUM, /**< maps to ALSA enum style controls */ + SOF_CTRL_CMD_SWITCH, /**< maps to ALSA switch style controls */ + SOF_CTRL_CMD_BINARY, /**< maps to ALSA binary style controls */ +}; + +/* generic channel mapped value data */ +struct sof_ipc_ctrl_value_chan { + uint32_t channel; /**< channel map - enum sof_ipc_chmap */ + uint32_t value; +} __packed; + +/* generic component mapped value data */ +struct sof_ipc_ctrl_value_comp { + uint32_t index; /**< component source/sink/control index in control */ + union { + uint32_t uvalue; + int32_t svalue; + }; +} __packed; + +/* generic control data */ +struct sof_ipc_ctrl_data { + struct sof_ipc_reply rhdr; + uint32_t comp_id; + + /* control access and data type */ + uint32_t type; /**< enum sof_ipc_ctrl_type */ + uint32_t cmd; /**< enum sof_ipc_ctrl_cmd */ + uint32_t index; /**< control index for comps > 1 control */ + + /* control data - can either be appended or DMAed from host */ + struct sof_ipc_host_buffer buffer; + uint32_t num_elems; /**< in array elems or bytes for data type */ + uint32_t elems_remaining; /**< elems remaining if sent in parts */ + + uint32_t msg_index; /**< for large messages sent in parts */ + + /* reserved for future use */ + uint32_t reserved[6]; + + /* control data - add new types if needed */ + union { + /* channel values can be used by volume type controls */ + struct sof_ipc_ctrl_value_chan chanv[0]; + /* component values used by routing controls like mux, mixer */ + struct sof_ipc_ctrl_value_comp compv[0]; + /* data can be used by binary controls */ + struct sof_abi_hdr data[0]; + }; +} __packed; + +/** Event type */ +enum sof_ipc_ctrl_event_type { + SOF_CTRL_EVENT_GENERIC = 0, /**< generic event */ + SOF_CTRL_EVENT_GENERIC_METADATA, /**< generic event with metadata */ + SOF_CTRL_EVENT_KD, /**< keyword detection event */ + SOF_CTRL_EVENT_VAD, /**< voice activity detection event */ +}; + +/** + * Generic notification data. + */ +struct sof_ipc_comp_event { + struct sof_ipc_reply rhdr; + uint16_t src_comp_type; /**< COMP_TYPE_ */ + uint32_t src_comp_id; /**< source component id */ + uint32_t event_type; /**< event type - SOF_CTRL_EVENT_* */ + uint32_t num_elems; /**< in array elems or bytes for data type */ + + /* reserved for future use */ + uint32_t reserved[8]; + + /* control data - add new types if needed */ + union { + /* data can be used by binary controls */ + struct sof_abi_hdr data[0]; + /* event specific values */ + uint32_t event_value; + }; +} __packed; + +#endif diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h new file mode 100644 index 000000000000..4bd83f7adddf --- /dev/null +++ b/include/sound/sof/dai-intel.h @@ -0,0 +1,178 @@ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_DAI_INTEL_H__ +#define __INCLUDE_SOUND_SOF_DAI_INTEL_H__ + +#include + + /* ssc1: TINTE */ +#define SOF_DAI_INTEL_SSP_QUIRK_TINTE (1 << 0) + /* ssc1: PINTE */ +#define SOF_DAI_INTEL_SSP_QUIRK_PINTE (1 << 1) + /* ssc2: SMTATF */ +#define SOF_DAI_INTEL_SSP_QUIRK_SMTATF (1 << 2) + /* ssc2: MMRATF */ +#define SOF_DAI_INTEL_SSP_QUIRK_MMRATF (1 << 3) + /* ssc2: PSPSTWFDFD */ +#define SOF_DAI_INTEL_SSP_QUIRK_PSPSTWFDFD (1 << 4) + /* ssc2: PSPSRWFDFD */ +#define SOF_DAI_INTEL_SSP_QUIRK_PSPSRWFDFD (1 << 5) +/* ssc1: LBM */ +#define SOF_DAI_INTEL_SSP_QUIRK_LBM (1 << 6) + + /* here is the possibility to define others aux macros */ + +#define SOF_DAI_INTEL_SSP_FRAME_PULSE_WIDTH_MAX 38 +#define SOF_DAI_INTEL_SSP_SLOT_PADDING_MAX 31 + +/* SSP clocks control settings + * + * Macros for clks_control field in sof_ipc_dai_ssp_params struct. + */ + +/* mclk 0 disable */ +#define SOF_DAI_INTEL_SSP_MCLK_0_DISABLE BIT(0) +/* mclk 1 disable */ +#define SOF_DAI_INTEL_SSP_MCLK_1_DISABLE BIT(1) +/* mclk keep active */ +#define SOF_DAI_INTEL_SSP_CLKCTRL_MCLK_KA BIT(2) +/* bclk keep active */ +#define SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_KA BIT(3) +/* fs keep active */ +#define SOF_DAI_INTEL_SSP_CLKCTRL_FS_KA BIT(4) +/* bclk idle */ +#define SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_IDLE_HIGH BIT(5) + +/* SSP Configuration Request - SOF_IPC_DAI_SSP_CONFIG */ +struct sof_ipc_dai_ssp_params { + struct sof_ipc_hdr hdr; + uint16_t reserved1; + uint16_t mclk_id; + + uint32_t mclk_rate; /* mclk frequency in Hz */ + uint32_t fsync_rate; /* fsync frequency in Hz */ + uint32_t bclk_rate; /* bclk frequency in Hz */ + + /* TDM */ + uint32_t tdm_slots; + uint32_t rx_slots; + uint32_t tx_slots; + + /* data */ + uint32_t sample_valid_bits; + uint16_t tdm_slot_width; + uint16_t reserved2; /* alignment */ + + /* MCLK */ + uint32_t mclk_direction; + + uint16_t frame_pulse_width; + uint16_t tdm_per_slot_padding_flag; + uint32_t clks_control; + uint32_t quirks; +} __packed; + +/* HDA Configuration Request - SOF_IPC_DAI_HDA_CONFIG */ +struct sof_ipc_dai_hda_params { + struct sof_ipc_hdr hdr; + uint32_t link_dma_ch; +} __packed; + +/* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */ + +/* This struct is defined per 2ch PDM controller available in the platform. + * Normally it is sufficient to set the used microphone specific enables to 1 + * and keep other parameters as zero. The customizations are: + * + * 1. If a device mixes different microphones types with different polarity + * and/or the absolute polarity matters the PCM signal from a microphone + * can be inverted with the controls. + * + * 2. If the microphones in a stereo pair do not appear in captured stream + * in desired order due to board schematics choises they can be swapped with + * the clk_edge parameter. + * + * 3. If PDM bit errors are seen in capture (poor quality) the skew parameter + * that delays the sampling time of data by half cycles of DMIC source clock + * can be tried for improvement. However there is no guarantee for this to fix + * data integrity problems. + */ +struct sof_ipc_dai_dmic_pdm_ctrl { + struct sof_ipc_hdr hdr; + uint16_t id; /**< PDM controller ID */ + + uint16_t enable_mic_a; /**< Use A (left) channel mic (0 or 1)*/ + uint16_t enable_mic_b; /**< Use B (right) channel mic (0 or 1)*/ + + uint16_t polarity_mic_a; /**< Optionally invert mic A signal (0 or 1) */ + uint16_t polarity_mic_b; /**< Optionally invert mic B signal (0 or 1) */ + + uint16_t clk_edge; /**< Optionally swap data clock edge (0 or 1) */ + uint16_t skew; /**< Adjust PDM data sampling vs. clock (0..15) */ + + uint16_t reserved[3]; /**< Make sure the total size is 4 bytes aligned */ +} __packed; + +/* This struct contains the global settings for all 2ch PDM controllers. The + * version number used in configuration data is checked vs. version used by + * device driver src/drivers/dmic.c need to match. It is incremented from + * initial value 1 if updates done for the to driver would alter the operation + * of the microhone. + * + * Note: The microphone clock (pdmclk_min, pdmclk_max, duty_min, duty_max) + * parameters need to be set as defined in microphone data sheet. E.g. clock + * range 1.0 - 3.2 MHz is usually supported microphones. Some microphones are + * multi-mode capable and there may be denied mic clock frequencies between + * the modes. In such case set the clock range limits of the desired mode to + * avoid the driver to set clock to an illegal rate. + * + * The duty cycle could be set to 48-52% if not known. Generally these + * parameters can be altered within data sheet specified limits to match + * required audio application performance power. + * + * The microphone clock needs to be usually about 50-80 times the used audio + * sample rate. With highest sample rates above 48 kHz this can relaxed + * somewhat. + * + * The parameter wake_up_time describes how long time the microphone needs + * for the data line to produce valid output from mic clock start. The driver + * will mute the captured audio for the given time. The min_clock_on_time + * parameter is used to prevent too short clock bursts to happen. The driver + * will keep the clock active after capture stop if this time is not yet + * met. The unit for both is microseconds (us). Exceed of 100 ms will be + * treated as an error. + */ +struct sof_ipc_dai_dmic_params { + struct sof_ipc_hdr hdr; + uint32_t driver_ipc_version; /**< Version (1..N) */ + + uint32_t pdmclk_min; /**< Minimum microphone clock in Hz (100000..N) */ + uint32_t pdmclk_max; /**< Maximum microphone clock in Hz (min...N) */ + + uint32_t fifo_fs; /**< FIFO sample rate in Hz (8000..96000) */ + uint32_t reserved_1; /**< Reserved */ + uint16_t fifo_bits; /**< FIFO word length (16 or 32) */ + uint16_t reserved_2; /**< Reserved */ + + uint16_t duty_min; /**< Min. mic clock duty cycle in % (20..80) */ + uint16_t duty_max; /**< Max. mic clock duty cycle in % (min..80) */ + + uint32_t num_pdm_active; /**< Number of active pdm controllers */ + + uint32_t wake_up_time; /**< Time from clock start to data (us) */ + uint32_t min_clock_on_time; /**< Min. time that clk is kept on (us) */ + + /* reserved for future use */ + uint32_t reserved[6]; + + /**< variable number of pdm controller config */ + struct sof_ipc_dai_dmic_pdm_ctrl pdm[0]; +} __packed; + +#endif diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h new file mode 100644 index 000000000000..3b67c93ff101 --- /dev/null +++ b/include/sound/sof/dai.h @@ -0,0 +1,75 @@ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_DAI_H__ +#define __INCLUDE_SOUND_SOF_DAI_H__ + +#include +#include + +/* + * DAI Configuration. + * + * Each different DAI type will have it's own structure and IPC cmd. + */ + +#define SOF_DAI_FMT_I2S 1 /**< I2S mode */ +#define SOF_DAI_FMT_RIGHT_J 2 /**< Right Justified mode */ +#define SOF_DAI_FMT_LEFT_J 3 /**< Left Justified mode */ +#define SOF_DAI_FMT_DSP_A 4 /**< L data MSB after FRM LRC */ +#define SOF_DAI_FMT_DSP_B 5 /**< L data MSB during FRM LRC */ +#define SOF_DAI_FMT_PDM 6 /**< Pulse density modulation */ + +#define SOF_DAI_FMT_CONT (1 << 4) /**< continuous clock */ +#define SOF_DAI_FMT_GATED (0 << 4) /**< clock is gated */ + +#define SOF_DAI_FMT_NB_NF (0 << 8) /**< normal bit clock + frame */ +#define SOF_DAI_FMT_NB_IF (2 << 8) /**< normal BCLK + inv FRM */ +#define SOF_DAI_FMT_IB_NF (3 << 8) /**< invert BCLK + nor FRM */ +#define SOF_DAI_FMT_IB_IF (4 << 8) /**< invert BCLK + FRM */ + +#define SOF_DAI_FMT_CBM_CFM (0 << 12) /**< codec clk & FRM master */ +#define SOF_DAI_FMT_CBS_CFM (2 << 12) /**< codec clk slave & FRM master */ +#define SOF_DAI_FMT_CBM_CFS (3 << 12) /**< codec clk master & frame slave */ +#define SOF_DAI_FMT_CBS_CFS (4 << 12) /**< codec clk & FRM slave */ + +#define SOF_DAI_FMT_FORMAT_MASK 0x000f +#define SOF_DAI_FMT_CLOCK_MASK 0x00f0 +#define SOF_DAI_FMT_INV_MASK 0x0f00 +#define SOF_DAI_FMT_MASTER_MASK 0xf000 + +/** \brief Types of DAI */ +enum sof_ipc_dai_type { + SOF_DAI_INTEL_NONE = 0, /**< None */ + SOF_DAI_INTEL_SSP, /**< Intel SSP */ + SOF_DAI_INTEL_DMIC, /**< Intel DMIC */ + SOF_DAI_INTEL_HDA, /**< Intel HD/A */ +}; + +/* general purpose DAI configuration */ +struct sof_ipc_dai_config { + struct sof_ipc_cmd_hdr hdr; + uint32_t type; /**< DAI type - enum sof_ipc_dai_type */ + uint32_t dai_index; /**< index of this type dai */ + + /* physical protocol and clocking */ + uint16_t format; /**< SOF_DAI_FMT_ */ + uint16_t reserved16; /**< alignment */ + + /* reserved for future use */ + uint32_t reserved[8]; + + /* HW specific data */ + union { + struct sof_ipc_dai_ssp_params ssp; + struct sof_ipc_dai_dmic_params dmic; + struct sof_ipc_dai_hda_params hda; + }; +} __packed; + +#endif diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h new file mode 100644 index 000000000000..ccb6a004b37b --- /dev/null +++ b/include/sound/sof/header.h @@ -0,0 +1,158 @@ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_HEADER_H__ +#define __INCLUDE_SOUND_SOF_HEADER_H__ + +#include + +/** \addtogroup sof_uapi uAPI + * SOF uAPI specification. + * @{ + */ + +/* + * IPC messages have a prefixed 32 bit identifier made up as follows :- + * + * 0xGCCCNNNN where + * G is global cmd type (4 bits) + * C is command type (12 bits) + * I is the ID number (16 bits) - monotonic and overflows + * + * This is sent at the start of the IPM message in the mailbox. Messages should + * not be sent in the doorbell (special exceptions for firmware . + */ + +/* Global Message - Generic */ +#define SOF_GLB_TYPE_SHIFT 28 +#define SOF_GLB_TYPE_MASK (0xf << SOF_GLB_TYPE_SHIFT) +#define SOF_GLB_TYPE(x) ((x) << SOF_GLB_TYPE_SHIFT) + +/* Command Message - Generic */ +#define SOF_CMD_TYPE_SHIFT 16 +#define SOF_CMD_TYPE_MASK (0xfff << SOF_CMD_TYPE_SHIFT) +#define SOF_CMD_TYPE(x) ((x) << SOF_CMD_TYPE_SHIFT) + +/* Global Message Types */ +#define SOF_IPC_GLB_REPLY SOF_GLB_TYPE(0x1U) +#define SOF_IPC_GLB_COMPOUND SOF_GLB_TYPE(0x2U) +#define SOF_IPC_GLB_TPLG_MSG SOF_GLB_TYPE(0x3U) +#define SOF_IPC_GLB_PM_MSG SOF_GLB_TYPE(0x4U) +#define SOF_IPC_GLB_COMP_MSG SOF_GLB_TYPE(0x5U) +#define SOF_IPC_GLB_STREAM_MSG SOF_GLB_TYPE(0x6U) +#define SOF_IPC_FW_READY SOF_GLB_TYPE(0x7U) +#define SOF_IPC_GLB_DAI_MSG SOF_GLB_TYPE(0x8U) +#define SOF_IPC_GLB_TRACE_MSG SOF_GLB_TYPE(0x9U) + +/* + * DSP Command Message Types + */ + +/* topology */ +#define SOF_IPC_TPLG_COMP_NEW SOF_CMD_TYPE(0x001) +#define SOF_IPC_TPLG_COMP_FREE SOF_CMD_TYPE(0x002) +#define SOF_IPC_TPLG_COMP_CONNECT SOF_CMD_TYPE(0x003) +#define SOF_IPC_TPLG_PIPE_NEW SOF_CMD_TYPE(0x010) +#define SOF_IPC_TPLG_PIPE_FREE SOF_CMD_TYPE(0x011) +#define SOF_IPC_TPLG_PIPE_CONNECT SOF_CMD_TYPE(0x012) +#define SOF_IPC_TPLG_PIPE_COMPLETE SOF_CMD_TYPE(0x013) +#define SOF_IPC_TPLG_BUFFER_NEW SOF_CMD_TYPE(0x020) +#define SOF_IPC_TPLG_BUFFER_FREE SOF_CMD_TYPE(0x021) + +/* PM */ +#define SOF_IPC_PM_CTX_SAVE SOF_CMD_TYPE(0x001) +#define SOF_IPC_PM_CTX_RESTORE SOF_CMD_TYPE(0x002) +#define SOF_IPC_PM_CTX_SIZE SOF_CMD_TYPE(0x003) +#define SOF_IPC_PM_CLK_SET SOF_CMD_TYPE(0x004) +#define SOF_IPC_PM_CLK_GET SOF_CMD_TYPE(0x005) +#define SOF_IPC_PM_CLK_REQ SOF_CMD_TYPE(0x006) +#define SOF_IPC_PM_CORE_ENABLE SOF_CMD_TYPE(0x007) + +/* component runtime config - multiple different types */ +#define SOF_IPC_COMP_SET_VALUE SOF_CMD_TYPE(0x001) +#define SOF_IPC_COMP_GET_VALUE SOF_CMD_TYPE(0x002) +#define SOF_IPC_COMP_SET_DATA SOF_CMD_TYPE(0x003) +#define SOF_IPC_COMP_GET_DATA SOF_CMD_TYPE(0x004) + +/* DAI messages */ +#define SOF_IPC_DAI_CONFIG SOF_CMD_TYPE(0x001) +#define SOF_IPC_DAI_LOOPBACK SOF_CMD_TYPE(0x002) + +/* stream */ +#define SOF_IPC_STREAM_PCM_PARAMS SOF_CMD_TYPE(0x001) +#define SOF_IPC_STREAM_PCM_PARAMS_REPLY SOF_CMD_TYPE(0x002) +#define SOF_IPC_STREAM_PCM_FREE SOF_CMD_TYPE(0x003) +#define SOF_IPC_STREAM_TRIG_START SOF_CMD_TYPE(0x004) +#define SOF_IPC_STREAM_TRIG_STOP SOF_CMD_TYPE(0x005) +#define SOF_IPC_STREAM_TRIG_PAUSE SOF_CMD_TYPE(0x006) +#define SOF_IPC_STREAM_TRIG_RELEASE SOF_CMD_TYPE(0x007) +#define SOF_IPC_STREAM_TRIG_DRAIN SOF_CMD_TYPE(0x008) +#define SOF_IPC_STREAM_TRIG_XRUN SOF_CMD_TYPE(0x009) +#define SOF_IPC_STREAM_POSITION SOF_CMD_TYPE(0x00a) +#define SOF_IPC_STREAM_VORBIS_PARAMS SOF_CMD_TYPE(0x010) +#define SOF_IPC_STREAM_VORBIS_FREE SOF_CMD_TYPE(0x011) + +/* trace and debug */ +#define SOF_IPC_TRACE_DMA_PARAMS SOF_CMD_TYPE(0x001) +#define SOF_IPC_TRACE_DMA_POSITION SOF_CMD_TYPE(0x002) + +/* Get message component id */ +#define SOF_IPC_MESSAGE_ID(x) ((x) & 0xffff) + +/* maximum message size for mailbox Tx/Rx */ +#define SOF_IPC_MSG_MAX_SIZE 384 + +/* + * Structure Header - Header for all IPC structures except command structs. + * The size can be greater than the structure size and that means there is + * extended bespoke data beyond the end of the structure including variable + * arrays. + */ + +struct sof_ipc_hdr { + uint32_t size; /**< size of structure */ +} __packed; + +/* + * Command Header - Header for all IPC commands. Identifies IPC message. + * The size can be greater than the structure size and that means there is + * extended bespoke data beyond the end of the structure including variable + * arrays. + */ + +struct sof_ipc_cmd_hdr { + uint32_t size; /**< size of structure */ + uint32_t cmd; /**< SOF_IPC_GLB_ + cmd */ +} __packed; + +/* + * Generic reply message. Some commands override this with their own reply + * types that must include this at start. + */ +struct sof_ipc_reply { + struct sof_ipc_cmd_hdr hdr; + int32_t error; /**< negative error numbers */ +} __packed; + +/* + * Compound commands - SOF_IPC_GLB_COMPOUND. + * + * Compound commands are sent to the DSP as a single IPC operation. The + * commands are split into blocks and each block has a header. This header + * identifies the command type and the number of commands before the next + * header. + */ + +struct sof_ipc_compound_hdr { + struct sof_ipc_cmd_hdr hdr; + uint32_t count; /**< count of 0 means end of compound sequence */ +} __packed; + +/** @}*/ + +#endif diff --git a/include/sound/sof/info.h b/include/sound/sof/info.h new file mode 100644 index 000000000000..21dae04d8183 --- /dev/null +++ b/include/sound/sof/info.h @@ -0,0 +1,118 @@ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_INFO_H__ +#define __INCLUDE_SOUND_SOF_INFO_H__ + +#include +#include + +/* + * Firmware boot and version + */ + +#define SOF_IPC_MAX_ELEMS 16 + +/* extended data types that can be appended onto end of sof_ipc_fw_ready */ +enum sof_ipc_ext_data { + SOF_IPC_EXT_DMA_BUFFER = 0, + SOF_IPC_EXT_WINDOW, +}; + +/* FW version - SOF_IPC_GLB_VERSION */ +struct sof_ipc_fw_version { + struct sof_ipc_hdr hdr; + uint16_t major; + uint16_t minor; + uint16_t micro; + uint16_t build; + uint8_t date[12]; + uint8_t time[10]; + uint8_t tag[6]; + uint32_t abi_version; + + /* reserved for future use */ + uint32_t reserved[4]; +} __packed; + +/* FW ready Message - sent by firmware when boot has completed */ +struct sof_ipc_fw_ready { + struct sof_ipc_cmd_hdr hdr; + uint32_t dspbox_offset; /* dsp initiated IPC mailbox */ + uint32_t hostbox_offset; /* host initiated IPC mailbox */ + uint32_t dspbox_size; + uint32_t hostbox_size; + struct sof_ipc_fw_version version; + + /* Miscellaneous debug flags showing build/debug features enabled */ + union { + uint64_t reserved; + struct { + uint64_t build:1; + uint64_t locks:1; + uint64_t locks_verbose:1; + uint64_t gdb:1; + } bits; + } debug; + + /* reserved for future use */ + uint32_t reserved[4]; +} __packed; + +/* + * Extended Firmware data. All optional, depends on platform/arch. + */ +enum sof_ipc_region { + SOF_IPC_REGION_DOWNBOX = 0, + SOF_IPC_REGION_UPBOX, + SOF_IPC_REGION_TRACE, + SOF_IPC_REGION_DEBUG, + SOF_IPC_REGION_STREAM, + SOF_IPC_REGION_REGS, + SOF_IPC_REGION_EXCEPTION, +}; + +struct sof_ipc_ext_data_hdr { + struct sof_ipc_cmd_hdr hdr; + uint32_t type; /**< SOF_IPC_EXT_ */ +} __packed; + +struct sof_ipc_dma_buffer_elem { + struct sof_ipc_hdr hdr; + uint32_t type; /**< SOF_IPC_REGION_ */ + uint32_t id; /**< platform specific - used to map to host memory */ + struct sof_ipc_host_buffer buffer; +} __packed; + +/* extended data DMA buffers for IPC, trace and debug */ +struct sof_ipc_dma_buffer_data { + struct sof_ipc_ext_data_hdr ext_hdr; + uint32_t num_buffers; + + /* host files in buffer[n].buffer */ + struct sof_ipc_dma_buffer_elem buffer[]; +} __packed; + +struct sof_ipc_window_elem { + struct sof_ipc_hdr hdr; + uint32_t type; /**< SOF_IPC_REGION_ */ + uint32_t id; /**< platform specific - used to map to host memory */ + uint32_t flags; /**< R, W, RW, etc - to define */ + uint32_t size; /**< size of region in bytes */ + /* offset in window region as windows can be partitioned */ + uint32_t offset; +} __packed; + +/* extended data memory windows for IPC, trace and debug */ +struct sof_ipc_window { + struct sof_ipc_ext_data_hdr ext_hdr; + uint32_t num_windows; + struct sof_ipc_window_elem window[]; +} __packed; + +#endif diff --git a/include/sound/sof/pm.h b/include/sound/sof/pm.h new file mode 100644 index 000000000000..8ae3ad45bdf7 --- /dev/null +++ b/include/sound/sof/pm.h @@ -0,0 +1,48 @@ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_PM_H__ +#define __INCLUDE_SOUND_SOF_PM_H__ + +#include + +/* + * PM + */ + +/* PM context element */ +struct sof_ipc_pm_ctx_elem { + struct sof_ipc_hdr hdr; + uint32_t type; + uint32_t size; + uint64_t addr; +} __packed; + +/* + * PM context - SOF_IPC_PM_CTX_SAVE, SOF_IPC_PM_CTX_RESTORE, + * SOF_IPC_PM_CTX_SIZE + */ +struct sof_ipc_pm_ctx { + struct sof_ipc_cmd_hdr hdr; + struct sof_ipc_host_buffer buffer; + uint32_t num_elems; + uint32_t size; + + /* reserved for future use */ + uint32_t reserved[8]; + + struct sof_ipc_pm_ctx_elem elems[]; +} __packed; + +/* enable or disable cores - SOF_IPC_PM_CORE_ENABLE */ +struct sof_ipc_pm_core_config { + struct sof_ipc_cmd_hdr hdr; + uint32_t enable_mask; +} __packed; + +#endif diff --git a/include/sound/sof/stream.h b/include/sound/sof/stream.h new file mode 100644 index 000000000000..643f175cb479 --- /dev/null +++ b/include/sound/sof/stream.h @@ -0,0 +1,148 @@ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_STREAM_H__ +#define __INCLUDE_SOUND_SOF_STREAM_H__ + +#include + +/* + * Stream configuration. + */ + +#define SOF_IPC_MAX_CHANNELS 8 + +/* common sample rates for use in masks */ +#define SOF_RATE_8000 (1 << 0) /**< 8000Hz */ +#define SOF_RATE_11025 (1 << 1) /**< 11025Hz */ +#define SOF_RATE_12000 (1 << 2) /**< 12000Hz */ +#define SOF_RATE_16000 (1 << 3) /**< 16000Hz */ +#define SOF_RATE_22050 (1 << 4) /**< 22050Hz */ +#define SOF_RATE_24000 (1 << 5) /**< 24000Hz */ +#define SOF_RATE_32000 (1 << 6) /**< 32000Hz */ +#define SOF_RATE_44100 (1 << 7) /**< 44100Hz */ +#define SOF_RATE_48000 (1 << 8) /**< 48000Hz */ +#define SOF_RATE_64000 (1 << 9) /**< 64000Hz */ +#define SOF_RATE_88200 (1 << 10) /**< 88200Hz */ +#define SOF_RATE_96000 (1 << 11) /**< 96000Hz */ +#define SOF_RATE_176400 (1 << 12) /**< 176400Hz */ +#define SOF_RATE_192000 (1 << 13) /**< 192000Hz */ + +/* continuous and non-standard rates for flexibility */ +#define SOF_RATE_CONTINUOUS (1 << 30) /**< range */ +#define SOF_RATE_KNOT (1 << 31) /**< non-continuous */ + +/* generic PCM flags for runtime settings */ +#define SOF_PCM_FLAG_XRUN_STOP (1 << 0) /**< Stop on any XRUN */ + +/* stream PCM frame format */ +enum sof_ipc_frame { + SOF_IPC_FRAME_S16_LE = 0, + SOF_IPC_FRAME_S24_4LE, + SOF_IPC_FRAME_S32_LE, + SOF_IPC_FRAME_FLOAT, + /* other formats here */ +}; + +/* stream buffer format */ +enum sof_ipc_buffer_format { + SOF_IPC_BUFFER_INTERLEAVED, + SOF_IPC_BUFFER_NONINTERLEAVED, + /* other formats here */ +}; + +/* stream direction */ +enum sof_ipc_stream_direction { + SOF_IPC_STREAM_PLAYBACK = 0, + SOF_IPC_STREAM_CAPTURE, +}; + +/* stream ring info */ +struct sof_ipc_host_buffer { + struct sof_ipc_hdr hdr; + uint32_t phy_addr; + uint32_t pages; + uint32_t size; + uint32_t reserved[3]; +} __packed; + +struct sof_ipc_stream_params { + struct sof_ipc_hdr hdr; + struct sof_ipc_host_buffer buffer; + uint32_t direction; /**< enum sof_ipc_stream_direction */ + uint32_t frame_fmt; /**< enum sof_ipc_frame */ + uint32_t buffer_fmt; /**< enum sof_ipc_buffer_format */ + uint32_t rate; + uint16_t stream_tag; + uint16_t channels; + uint16_t sample_valid_bytes; + uint16_t sample_container_bytes; + + /* for notifying host period has completed - 0 means no period IRQ */ + uint32_t host_period_bytes; + + uint32_t reserved[2]; + uint16_t chmap[SOF_IPC_MAX_CHANNELS]; /**< channel map - SOF_CHMAP_ */ +} __packed; + +/* PCM params info - SOF_IPC_STREAM_PCM_PARAMS */ +struct sof_ipc_pcm_params { + struct sof_ipc_cmd_hdr hdr; + uint32_t comp_id; + uint32_t flags; /**< generic PCM flags - SOF_PCM_FLAG_ */ + uint32_t reserved[2]; + struct sof_ipc_stream_params params; +} __packed; + +/* PCM params info reply - SOF_IPC_STREAM_PCM_PARAMS_REPLY */ +struct sof_ipc_pcm_params_reply { + struct sof_ipc_reply rhdr; + uint32_t comp_id; + uint32_t posn_offset; +} __packed; + +/* free stream - SOF_IPC_STREAM_PCM_PARAMS */ +struct sof_ipc_stream { + struct sof_ipc_cmd_hdr hdr; + uint32_t comp_id; +} __packed; + +/* flags indicating which time stamps are in sync with each other */ +#define SOF_TIME_HOST_SYNC (1 << 0) +#define SOF_TIME_DAI_SYNC (1 << 1) +#define SOF_TIME_WALL_SYNC (1 << 2) +#define SOF_TIME_STAMP_SYNC (1 << 3) + +/* flags indicating which time stamps are valid */ +#define SOF_TIME_HOST_VALID (1 << 8) +#define SOF_TIME_DAI_VALID (1 << 9) +#define SOF_TIME_WALL_VALID (1 << 10) +#define SOF_TIME_STAMP_VALID (1 << 11) + +/* flags indicating time stamps are 64bit else 3use low 32bit */ +#define SOF_TIME_HOST_64 (1 << 16) +#define SOF_TIME_DAI_64 (1 << 17) +#define SOF_TIME_WALL_64 (1 << 18) +#define SOF_TIME_STAMP_64 (1 << 19) + +struct sof_ipc_stream_posn { + struct sof_ipc_reply rhdr; + uint32_t comp_id; /**< host component ID */ + uint32_t flags; /**< SOF_TIME_ */ + uint32_t wallclock_hz; /**< frequency of wallclock in Hz */ + uint32_t timestamp_ns; /**< resolution of timestamp in ns */ + uint64_t host_posn; /**< host DMA position in bytes */ + uint64_t dai_posn; /**< DAI DMA position in bytes */ + uint64_t comp_posn; /**< comp position in bytes */ + uint64_t wallclock; /**< audio wall clock */ + uint64_t timestamp; /**< system time stamp */ + uint32_t xrun_comp_id; /**< comp ID of XRUN component */ + int32_t xrun_size; /**< XRUN size in bytes */ +} __packed; + +#endif diff --git a/include/sound/sof/trace.h b/include/sound/sof/trace.h new file mode 100644 index 000000000000..7d211f319a92 --- /dev/null +++ b/include/sound/sof/trace.h @@ -0,0 +1,67 @@ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_TRACE_H__ +#define __INCLUDE_SOUND_SOF_TRACE_H__ + +#include +#include + +/* + * DMA for Trace + */ + +#define SOF_TRACE_FILENAME_SIZE 32 + +/* DMA for Trace params info - SOF_IPC_DEBUG_DMA_PARAMS */ +struct sof_ipc_dma_trace_params { + struct sof_ipc_cmd_hdr hdr; + struct sof_ipc_host_buffer buffer; + uint32_t stream_tag; +} __packed; + +/* DMA for Trace params info - SOF_IPC_DEBUG_DMA_PARAMS */ +struct sof_ipc_dma_trace_posn { + struct sof_ipc_reply rhdr; + uint32_t host_offset; /* Offset of DMA host buffer */ + uint32_t overflow; /* overflow bytes if any */ + uint32_t messages; /* total trace messages */ +} __packed; + +/* + * Commom debug + */ + +/* + * SOF panic codes + */ +#define SOF_IPC_PANIC_MAGIC 0x0dead000 +#define SOF_IPC_PANIC_MAGIC_MASK 0x0ffff000 +#define SOF_IPC_PANIC_CODE_MASK 0x00000fff +#define SOF_IPC_PANIC_MEM (SOF_IPC_PANIC_MAGIC | 0x0) +#define SOF_IPC_PANIC_WORK (SOF_IPC_PANIC_MAGIC | 0x1) +#define SOF_IPC_PANIC_IPC (SOF_IPC_PANIC_MAGIC | 0x2) +#define SOF_IPC_PANIC_ARCH (SOF_IPC_PANIC_MAGIC | 0x3) +#define SOF_IPC_PANIC_PLATFORM (SOF_IPC_PANIC_MAGIC | 0x4) +#define SOF_IPC_PANIC_TASK (SOF_IPC_PANIC_MAGIC | 0x5) +#define SOF_IPC_PANIC_EXCEPTION (SOF_IPC_PANIC_MAGIC | 0x6) +#define SOF_IPC_PANIC_DEADLOCK (SOF_IPC_PANIC_MAGIC | 0x7) +#define SOF_IPC_PANIC_STACK (SOF_IPC_PANIC_MAGIC | 0x8) +#define SOF_IPC_PANIC_IDLE (SOF_IPC_PANIC_MAGIC | 0x9) +#define SOF_IPC_PANIC_WFI (SOF_IPC_PANIC_MAGIC | 0xa) +#define SOF_IPC_PANIC_ASSERT (SOF_IPC_PANIC_MAGIC | 0xb) + +/* panic info include filename and line number */ +struct sof_ipc_panic_info { + struct sof_ipc_hdr hdr; + uint32_t code; /* SOF_IPC_PANIC_ */ + char filename[SOF_TRACE_FILENAME_SIZE]; + uint32_t linenum; +} __packed; + +#endif diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c new file mode 100644 index 000000000000..d00373ceca12 --- /dev/null +++ b/sound/soc/sof/ipc.c @@ -0,0 +1,832 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// +// Generic IPC layer that can work over MMIO and SPI/I2C. PHY layer provided +// by platform driver code. +// + +#include +#include + +#include "sof-priv.h" +#include "ops.h" + +/* + * IPC message default size and timeout (ms). + * TODO: allow platforms to set size and timeout. + */ +#define IPC_TIMEOUT_MS 300 + +static void ipc_trace_message(struct snd_sof_dev *sdev, u32 msg_id); +static void ipc_stream_message(struct snd_sof_dev *sdev, u32 msg_cmd); + +/* + * IPC message Tx/Rx message handling. + */ + +/* SOF generic IPC data */ +struct snd_sof_ipc { + struct snd_sof_dev *sdev; + + /* protects messages and the disable flag */ + struct mutex tx_mutex; + /* disables further sending of ipc's */ + bool disable_ipc_tx; + + struct snd_sof_ipc_msg msg; +}; + +struct sof_ipc_ctrl_data_params { + size_t msg_bytes; + size_t hdr_bytes; + size_t pl_size; + size_t elems; + u32 num_msg; + u8 *src; + u8 *dst; +}; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_VERBOSE_IPC) +static void ipc_log_header(struct device *dev, u8 *text, u32 cmd) +{ + u8 *str; + u8 *str2 = NULL; + u32 glb; + u32 type; + + glb = cmd & SOF_GLB_TYPE_MASK; + type = cmd & SOF_CMD_TYPE_MASK; + + switch (glb) { + case SOF_IPC_GLB_REPLY: + str = "GLB_REPLY"; break; + case SOF_IPC_GLB_COMPOUND: + str = "GLB_COMPOUND"; break; + case SOF_IPC_GLB_TPLG_MSG: + str = "GLB_TPLG_MSG"; + switch (type) { + case SOF_IPC_TPLG_COMP_NEW: + str2 = "COMP_NEW"; break; + case SOF_IPC_TPLG_COMP_FREE: + str2 = "COMP_FREE"; break; + case SOF_IPC_TPLG_COMP_CONNECT: + str2 = "COMP_CONNECT"; break; + case SOF_IPC_TPLG_PIPE_NEW: + str2 = "PIPE_NEW"; break; + case SOF_IPC_TPLG_PIPE_FREE: + str2 = "PIPE_FREE"; break; + case SOF_IPC_TPLG_PIPE_CONNECT: + str2 = "PIPE_CONNECT"; break; + case SOF_IPC_TPLG_PIPE_COMPLETE: + str2 = "PIPE_COMPLETE"; break; + case SOF_IPC_TPLG_BUFFER_NEW: + str2 = "BUFFER_NEW"; break; + case SOF_IPC_TPLG_BUFFER_FREE: + str2 = "BUFFER_FREE"; break; + default: + str2 = "unknown type"; break; + } + break; + case SOF_IPC_GLB_PM_MSG: + str = "GLB_PM_MSG"; + switch (type) { + case SOF_IPC_PM_CTX_SAVE: + str2 = "CTX_SAVE"; break; + case SOF_IPC_PM_CTX_RESTORE: + str2 = "CTX_RESTORE"; break; + case SOF_IPC_PM_CTX_SIZE: + str2 = "CTX_SIZE"; break; + case SOF_IPC_PM_CLK_SET: + str2 = "CLK_SET"; break; + case SOF_IPC_PM_CLK_GET: + str2 = "CLK_GET"; break; + case SOF_IPC_PM_CLK_REQ: + str2 = "CLK_REQ"; break; + case SOF_IPC_PM_CORE_ENABLE: + str2 = "CORE_ENABLE"; break; + default: + str2 = "unknown type"; break; + } + break; + case SOF_IPC_GLB_COMP_MSG: + str = "GLB_COMP_MSG: SET_VALUE"; + switch (type) { + case SOF_IPC_COMP_SET_VALUE: + str2 = "SET_VALUE"; break; + case SOF_IPC_COMP_GET_VALUE: + str2 = "GET_VALUE"; break; + case SOF_IPC_COMP_SET_DATA: + str2 = "SET_DATA"; break; + case SOF_IPC_COMP_GET_DATA: + str2 = "GET_DATA"; break; + default: + str2 = "unknown type"; break; + } + break; + case SOF_IPC_GLB_STREAM_MSG: + str = "GLB_STREAM_MSG"; + switch (type) { + case SOF_IPC_STREAM_PCM_PARAMS: + str2 = "PCM_PARAMS"; break; + case SOF_IPC_STREAM_PCM_PARAMS_REPLY: + str2 = "PCM_REPLY"; break; + case SOF_IPC_STREAM_PCM_FREE: + str2 = "PCM_FREE"; break; + case SOF_IPC_STREAM_TRIG_START: + str2 = "TRIG_START"; break; + case SOF_IPC_STREAM_TRIG_STOP: + str2 = "TRIG_STOP"; break; + case SOF_IPC_STREAM_TRIG_PAUSE: + str2 = "TRIG_PAUSE"; break; + case SOF_IPC_STREAM_TRIG_RELEASE: + str2 = "TRIG_RELEASE"; break; + case SOF_IPC_STREAM_TRIG_DRAIN: + str2 = "TRIG_DRAIN"; break; + case SOF_IPC_STREAM_TRIG_XRUN: + str2 = "TRIG_XRUN"; break; + case SOF_IPC_STREAM_POSITION: + str2 = "POSITION"; break; + case SOF_IPC_STREAM_VORBIS_PARAMS: + str2 = "VORBIS_PARAMS"; break; + case SOF_IPC_STREAM_VORBIS_FREE: + str2 = "VORBIS_FREE"; break; + default: + str2 = "unknown type"; break; + } + break; + case SOF_IPC_FW_READY: + str = "FW_READY"; break; + case SOF_IPC_GLB_DAI_MSG: + str = "GLB_DAI_MSG"; + switch (type) { + case SOF_IPC_DAI_CONFIG: + str2 = "CONFIG"; break; + case SOF_IPC_DAI_LOOPBACK: + str2 = "LOOPBACK"; break; + default: + str2 = "unknown type"; break; + } + break; + case SOF_IPC_GLB_TRACE_MSG: + str = "GLB_TRACE_MSG"; break; + default: + str = "unknown GLB command"; break; + } + + if (str2) + dev_dbg(dev, "%s: 0x%x: %s: %s\n", text, cmd, str, str2); + else + dev_dbg(dev, "%s: 0x%x: %s\n", text, cmd, str); +} +#else +static inline void ipc_log_header(struct device *dev, u8 *text, u32 cmd) +{ + dev_dbg(dev, "%s: 0x%x\n", text, cmd); +} +#endif + +/* wait for IPC message reply */ +static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg, + void *reply_data) +{ + struct snd_sof_dev *sdev = ipc->sdev; + struct sof_ipc_cmd_hdr *hdr = msg->msg_data; + int ret; + + /* wait for DSP IPC completion */ + ret = wait_event_timeout(msg->waitq, msg->ipc_complete, + msecs_to_jiffies(IPC_TIMEOUT_MS)); + + if (ret == 0) { + dev_err(sdev->dev, "error: ipc timed out for 0x%x size %d\n", + hdr->cmd, hdr->size); + snd_sof_dsp_dbg_dump(ipc->sdev, SOF_DBG_REGS | SOF_DBG_MBOX); + snd_sof_trace_notify_for_error(ipc->sdev); + ret = -ETIMEDOUT; + } else { + /* copy the data returned from DSP */ + ret = msg->reply_error; + if (msg->reply_size) + memcpy(reply_data, msg->reply_data, msg->reply_size); + if (ret < 0) + dev_err(sdev->dev, "error: ipc error for 0x%x size %zu\n", + hdr->cmd, msg->reply_size); + else + ipc_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); + } + + return ret; +} + +/* send IPC message from host to DSP */ +static int sof_ipc_tx_message_unlocked(struct snd_sof_ipc *ipc, u32 header, + void *msg_data, size_t msg_bytes, + void *reply_data, size_t reply_bytes) +{ + struct snd_sof_dev *sdev = ipc->sdev; + struct snd_sof_ipc_msg *msg; + int ret; + + if (ipc->disable_ipc_tx) + return -ENODEV; + + /* + * The spin-lock is also still needed to protect message objects against + * other atomic contexts. + */ + spin_lock_irq(&sdev->ipc_lock); + + /* initialise the message */ + msg = &ipc->msg; + + msg->header = header; + msg->msg_size = msg_bytes; + msg->reply_size = reply_bytes; + msg->reply_error = 0; + + /* attach any data */ + if (msg_bytes) + memcpy(msg->msg_data, msg_data, msg_bytes); + + sdev->msg = msg; + + ret = snd_sof_dsp_send_msg(sdev, msg); + /* Next reply that we receive will be related to this message */ + if (!ret) + msg->ipc_complete = false; + + spin_unlock_irq(&sdev->ipc_lock); + + if (ret < 0) { + /* So far IPC TX never fails, consider making the above void */ + dev_err_ratelimited(sdev->dev, + "error: ipc tx failed with error %d\n", + ret); + return ret; + } + + ipc_log_header(sdev->dev, "ipc tx", msg->header); + + /* now wait for completion */ + if (!ret) + ret = tx_wait_done(ipc, msg, reply_data); + + return ret; +} + +/* send IPC message from host to DSP */ +int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header, + void *msg_data, size_t msg_bytes, void *reply_data, + size_t reply_bytes) +{ + int ret; + + if (msg_bytes > SOF_IPC_MSG_MAX_SIZE || + reply_bytes > SOF_IPC_MSG_MAX_SIZE) + return -ENOBUFS; + + /* Serialise IPC TX */ + mutex_lock(&ipc->tx_mutex); + + ret = sof_ipc_tx_message_unlocked(ipc, header, msg_data, msg_bytes, + reply_data, reply_bytes); + + mutex_unlock(&ipc->tx_mutex); + + return ret; +} +EXPORT_SYMBOL(sof_ipc_tx_message); + +/* handle reply message from DSP */ +int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id) +{ + struct snd_sof_ipc_msg *msg = &sdev->ipc->msg; + unsigned long flags; + + /* + * Protect against a theoretical race with sof_ipc_tx_message(): if the + * DSP is fast enough to receive an IPC message, reply to it, and the + * host interrupt processing calls this function on a different core + * from the one, where the sending is taking place, the message might + * not yet be marked as expecting a reply. + */ + spin_lock_irqsave(&sdev->ipc_lock, flags); + + if (msg->ipc_complete) { + spin_unlock_irqrestore(&sdev->ipc_lock, flags); + dev_err(sdev->dev, "error: no reply expected, received 0x%x", + msg_id); + return -EINVAL; + } + + /* wake up and return the error if we have waiters on this message ? */ + msg->ipc_complete = true; + wake_up(&msg->waitq); + + spin_unlock_irqrestore(&sdev->ipc_lock, flags); + + return 0; +} +EXPORT_SYMBOL(snd_sof_ipc_reply); + +/* DSP firmware has sent host a message */ +void snd_sof_ipc_msgs_rx(struct snd_sof_dev *sdev) +{ + struct sof_ipc_cmd_hdr hdr; + u32 cmd, type; + int err = 0; + + /* read back header */ + snd_sof_ipc_msg_data(sdev, NULL, &hdr, sizeof(hdr)); + ipc_log_header(sdev->dev, "ipc rx", hdr.cmd); + + cmd = hdr.cmd & SOF_GLB_TYPE_MASK; + type = hdr.cmd & SOF_CMD_TYPE_MASK; + + /* check message type */ + switch (cmd) { + case SOF_IPC_GLB_REPLY: + dev_err(sdev->dev, "error: ipc reply unknown\n"); + break; + case SOF_IPC_FW_READY: + /* check for FW boot completion */ + if (!sdev->boot_complete) { + err = sof_ops(sdev)->fw_ready(sdev, cmd); + if (err < 0) { + /* + * this indicates a mismatch in ABI + * between the driver and fw + */ + dev_err(sdev->dev, "error: ABI mismatch %d\n", + err); + } else { + /* firmware boot completed OK */ + sdev->boot_complete = true; + } + + /* wake up firmware loader */ + wake_up(&sdev->boot_wait); + } + break; + case SOF_IPC_GLB_COMPOUND: + case SOF_IPC_GLB_TPLG_MSG: + case SOF_IPC_GLB_PM_MSG: + case SOF_IPC_GLB_COMP_MSG: + break; + case SOF_IPC_GLB_STREAM_MSG: + /* need to pass msg id into the function */ + ipc_stream_message(sdev, hdr.cmd); + break; + case SOF_IPC_GLB_TRACE_MSG: + ipc_trace_message(sdev, type); + break; + default: + dev_err(sdev->dev, "error: unknown DSP message 0x%x\n", cmd); + break; + } + + ipc_log_header(sdev->dev, "ipc rx done", hdr.cmd); +} +EXPORT_SYMBOL(snd_sof_ipc_msgs_rx); + +/* + * IPC trace mechanism. + */ + +static void ipc_trace_message(struct snd_sof_dev *sdev, u32 msg_id) +{ + struct sof_ipc_dma_trace_posn posn; + + switch (msg_id) { + case SOF_IPC_TRACE_DMA_POSITION: + /* read back full message */ + snd_sof_ipc_msg_data(sdev, NULL, &posn, sizeof(posn)); + snd_sof_trace_update_pos(sdev, &posn); + break; + default: + dev_err(sdev->dev, "error: unhandled trace message %x\n", + msg_id); + break; + } +} + +/* + * IPC stream position. + */ + +static void ipc_period_elapsed(struct snd_sof_dev *sdev, u32 msg_id) +{ + struct snd_sof_pcm_stream *stream; + struct sof_ipc_stream_posn posn; + struct snd_sof_pcm *spcm; + int direction; + + spcm = snd_sof_find_spcm_comp(sdev, msg_id, &direction); + if (!spcm) { + dev_err(sdev->dev, + "error: period elapsed for unknown stream, msg_id %d\n", + msg_id); + return; + } + + stream = &spcm->stream[direction]; + snd_sof_ipc_msg_data(sdev, stream->substream, &posn, sizeof(posn)); + + dev_dbg(sdev->dev, "posn : host 0x%llx dai 0x%llx wall 0x%llx\n", + posn.host_posn, posn.dai_posn, posn.wallclock); + + memcpy(&stream->posn, &posn, sizeof(posn)); + + /* only inform ALSA for period_wakeup mode */ + if (!stream->substream->runtime->no_period_wakeup) + snd_pcm_period_elapsed(stream->substream); +} + +/* DSP notifies host of an XRUN within FW */ +static void ipc_xrun(struct snd_sof_dev *sdev, u32 msg_id) +{ + struct snd_sof_pcm_stream *stream; + struct sof_ipc_stream_posn posn; + struct snd_sof_pcm *spcm; + int direction; + + spcm = snd_sof_find_spcm_comp(sdev, msg_id, &direction); + if (!spcm) { + dev_err(sdev->dev, "error: XRUN for unknown stream, msg_id %d\n", + msg_id); + return; + } + + stream = &spcm->stream[direction]; + snd_sof_ipc_msg_data(sdev, stream->substream, &posn, sizeof(posn)); + + dev_dbg(sdev->dev, "posn XRUN: host %llx comp %d size %d\n", + posn.host_posn, posn.xrun_comp_id, posn.xrun_size); + +#if defined(CONFIG_SND_SOC_SOF_DEBUG_XRUN_STOP) + /* stop PCM on XRUN - used for pipeline debug */ + memcpy(&stream->posn, &posn, sizeof(posn)); + snd_pcm_stop_xrun(stream->substream); +#endif +} + +/* stream notifications from DSP FW */ +static void ipc_stream_message(struct snd_sof_dev *sdev, u32 msg_cmd) +{ + /* get msg cmd type and msd id */ + u32 msg_type = msg_cmd & SOF_CMD_TYPE_MASK; + u32 msg_id = SOF_IPC_MESSAGE_ID(msg_cmd); + + switch (msg_type) { + case SOF_IPC_STREAM_POSITION: + ipc_period_elapsed(sdev, msg_id); + break; + case SOF_IPC_STREAM_TRIG_XRUN: + ipc_xrun(sdev, msg_id); + break; + default: + dev_err(sdev->dev, "error: unhandled stream message %x\n", + msg_id); + break; + } +} + +/* get stream position IPC - use faster MMIO method if available on platform */ +int snd_sof_ipc_stream_posn(struct snd_sof_dev *sdev, + struct snd_sof_pcm *spcm, int direction, + struct sof_ipc_stream_posn *posn) +{ + struct sof_ipc_stream stream; + int err; + + /* read position via slower IPC */ + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_POSITION; + stream.comp_id = spcm->stream[direction].comp_id; + + /* send IPC to the DSP */ + err = sof_ipc_tx_message(sdev->ipc, + stream.hdr.cmd, &stream, sizeof(stream), &posn, + sizeof(*posn)); + if (err < 0) { + dev_err(sdev->dev, "error: failed to get stream %d position\n", + stream.comp_id); + return err; + } + + return 0; +} +EXPORT_SYMBOL(snd_sof_ipc_stream_posn); + +static int sof_get_ctrl_copy_params(enum sof_ipc_ctrl_type ctrl_type, + struct sof_ipc_ctrl_data *src, + struct sof_ipc_ctrl_data *dst, + struct sof_ipc_ctrl_data_params *sparams) +{ + switch (ctrl_type) { + case SOF_CTRL_TYPE_VALUE_CHAN_GET: + case SOF_CTRL_TYPE_VALUE_CHAN_SET: + sparams->src = (u8 *)src->chanv; + sparams->dst = (u8 *)dst->chanv; + break; + case SOF_CTRL_TYPE_VALUE_COMP_GET: + case SOF_CTRL_TYPE_VALUE_COMP_SET: + sparams->src = (u8 *)src->compv; + sparams->dst = (u8 *)dst->compv; + break; + case SOF_CTRL_TYPE_DATA_GET: + case SOF_CTRL_TYPE_DATA_SET: + sparams->src = (u8 *)src->data->data; + sparams->dst = (u8 *)dst->data->data; + break; + default: + return -EINVAL; + } + + /* calculate payload size and number of messages */ + sparams->pl_size = SOF_IPC_MSG_MAX_SIZE - sparams->hdr_bytes; + sparams->num_msg = DIV_ROUND_UP(sparams->msg_bytes, sparams->pl_size); + + return 0; +} + +static int sof_set_get_large_ctrl_data(struct snd_sof_dev *sdev, + struct sof_ipc_ctrl_data *cdata, + struct sof_ipc_ctrl_data_params *sparams, + bool send) +{ + struct sof_ipc_ctrl_data *partdata; + size_t send_bytes; + size_t offset = 0; + size_t msg_bytes; + size_t pl_size; + int err = 0; + int i; + + /* allocate max ipc size because we have at least one */ + partdata = kzalloc(SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL); + if (!partdata) + return -ENOMEM; + + if (send) + sof_get_ctrl_copy_params(cdata->type, cdata, partdata, sparams); + else + sof_get_ctrl_copy_params(cdata->type, partdata, cdata, sparams); + + msg_bytes = sparams->msg_bytes; + pl_size = sparams->pl_size; + + /* copy the header data */ + memcpy(partdata, cdata, sparams->hdr_bytes); + + /* Serialise IPC TX */ + mutex_lock(&sdev->ipc->tx_mutex); + + /* copy the payload data in a loop */ + for (i = 0; i < sparams->num_msg; i++) { + send_bytes = min(msg_bytes, pl_size); + partdata->num_elems = send_bytes; + partdata->rhdr.hdr.size = sparams->hdr_bytes + send_bytes; + partdata->msg_index = i; + msg_bytes -= send_bytes; + partdata->elems_remaining = msg_bytes; + + if (send) + memcpy(sparams->dst, sparams->src + offset, send_bytes); + + err = sof_ipc_tx_message_unlocked(sdev->ipc, + partdata->rhdr.hdr.cmd, + partdata, + partdata->rhdr.hdr.size, + partdata, + partdata->rhdr.hdr.size); + if (err < 0) + break; + + if (!send) + memcpy(sparams->dst + offset, sparams->src, send_bytes); + + offset += pl_size; + } + + mutex_unlock(&sdev->ipc->tx_mutex); + + kfree(partdata); + return err; +} + +/* + * IPC get()/set() for kcontrols. + */ +int snd_sof_ipc_set_get_comp_data(struct snd_sof_ipc *ipc, + struct snd_sof_control *scontrol, + u32 ipc_cmd, + enum sof_ipc_ctrl_type ctrl_type, + enum sof_ipc_ctrl_cmd ctrl_cmd, + bool send) +{ + struct sof_ipc_ctrl_data *cdata = scontrol->control_data; + struct snd_sof_dev *sdev = ipc->sdev; + struct sof_ipc_fw_ready *ready = &sdev->fw_ready; + struct sof_ipc_fw_version *v = &ready->version; + struct sof_ipc_ctrl_data_params sparams; + size_t send_bytes; + int err; + + /* read or write firmware volume */ + if (scontrol->readback_offset != 0) { + /* write/read value header via mmaped region */ + send_bytes = sizeof(struct sof_ipc_ctrl_value_chan) * + cdata->num_elems; + if (send) + snd_sof_dsp_block_write(sdev, sdev->mmio_bar, + scontrol->readback_offset, + cdata->chanv, send_bytes); + + else + snd_sof_dsp_block_read(sdev, sdev->mmio_bar, + scontrol->readback_offset, + cdata->chanv, send_bytes); + return 0; + } + + cdata->rhdr.hdr.cmd = SOF_IPC_GLB_COMP_MSG | ipc_cmd; + cdata->cmd = ctrl_cmd; + cdata->type = ctrl_type; + cdata->comp_id = scontrol->comp_id; + cdata->msg_index = 0; + + /* calculate header and data size */ + switch (cdata->type) { + case SOF_CTRL_TYPE_VALUE_CHAN_GET: + case SOF_CTRL_TYPE_VALUE_CHAN_SET: + sparams.msg_bytes = scontrol->num_channels * + sizeof(struct sof_ipc_ctrl_value_chan); + sparams.hdr_bytes = sizeof(struct sof_ipc_ctrl_data); + sparams.elems = scontrol->num_channels; + break; + case SOF_CTRL_TYPE_VALUE_COMP_GET: + case SOF_CTRL_TYPE_VALUE_COMP_SET: + sparams.msg_bytes = scontrol->num_channels * + sizeof(struct sof_ipc_ctrl_value_comp); + sparams.hdr_bytes = sizeof(struct sof_ipc_ctrl_data); + sparams.elems = scontrol->num_channels; + break; + case SOF_CTRL_TYPE_DATA_GET: + case SOF_CTRL_TYPE_DATA_SET: + sparams.msg_bytes = cdata->data->size; + sparams.hdr_bytes = sizeof(struct sof_ipc_ctrl_data) + + sizeof(struct sof_abi_hdr); + sparams.elems = cdata->data->size; + break; + default: + return -EINVAL; + } + + cdata->rhdr.hdr.size = sparams.msg_bytes + sparams.hdr_bytes; + cdata->num_elems = sparams.elems; + cdata->elems_remaining = 0; + + /* send normal size ipc in one part */ + if (cdata->rhdr.hdr.size <= SOF_IPC_MSG_MAX_SIZE) { + err = sof_ipc_tx_message(sdev->ipc, cdata->rhdr.hdr.cmd, cdata, + cdata->rhdr.hdr.size, cdata, + cdata->rhdr.hdr.size); + + if (err < 0) + dev_err(sdev->dev, "error: set/get ctrl ipc comp %d\n", + cdata->comp_id); + + return err; + } + + /* data is bigger than max ipc size, chop into smaller pieces */ + dev_dbg(sdev->dev, "large ipc size %u, control size %u\n", + cdata->rhdr.hdr.size, scontrol->size); + + /* large messages is only supported from ABI 3.3.0 onwards */ + if (v->abi_version < SOF_ABI_VER(3, 3, 0)) { + dev_err(sdev->dev, "error: incompatible FW ABI version\n"); + return -EINVAL; + } + + err = sof_set_get_large_ctrl_data(sdev, cdata, &sparams, send); + + if (err < 0) + dev_err(sdev->dev, "error: set/get large ctrl ipc comp %d\n", + cdata->comp_id); + + return err; +} +EXPORT_SYMBOL(snd_sof_ipc_set_get_comp_data); + +/* + * IPC layer enumeration. + */ + +int snd_sof_dsp_mailbox_init(struct snd_sof_dev *sdev, u32 dspbox, + size_t dspbox_size, u32 hostbox, + size_t hostbox_size) +{ + sdev->dsp_box.offset = dspbox; + sdev->dsp_box.size = dspbox_size; + sdev->host_box.offset = hostbox; + sdev->host_box.size = hostbox_size; + return 0; +} +EXPORT_SYMBOL(snd_sof_dsp_mailbox_init); + +int snd_sof_ipc_valid(struct snd_sof_dev *sdev) +{ + struct sof_ipc_fw_ready *ready = &sdev->fw_ready; + struct sof_ipc_fw_version *v = &ready->version; + + dev_info(sdev->dev, + "Firmware info: version %d:%d:%d-%s\n", v->major, v->minor, + v->micro, v->tag); + dev_info(sdev->dev, + "Firmware: ABI %d:%d:%d Kernel ABI %d:%d:%d\n", + SOF_ABI_VERSION_MAJOR(v->abi_version), + SOF_ABI_VERSION_MINOR(v->abi_version), + SOF_ABI_VERSION_PATCH(v->abi_version), + SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); + + if (SOF_ABI_VERSION_INCOMPATIBLE(SOF_ABI_VERSION, v->abi_version)) { + dev_err(sdev->dev, "error: incompatible FW ABI version\n"); + return -EINVAL; + } + + if (ready->debug.bits.build) { + dev_info(sdev->dev, + "Firmware debug build %d on %s-%s - options:\n" + " GDB: %s\n" + " lock debug: %s\n" + " lock vdebug: %s\n", + v->build, v->date, v->time, + ready->debug.bits.gdb ? "enabled" : "disabled", + ready->debug.bits.locks ? "enabled" : "disabled", + ready->debug.bits.locks_verbose ? "enabled" : "disabled"); + } + + /* copy the fw_version into debugfs at first boot */ + memcpy(&sdev->fw_version, v, sizeof(*v)); + + return 0; +} +EXPORT_SYMBOL(snd_sof_ipc_valid); + +struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev) +{ + struct snd_sof_ipc *ipc; + struct snd_sof_ipc_msg *msg; + + /* check if mandatory ops required for ipc are defined */ + if (!sof_ops(sdev)->fw_ready) { + dev_err(sdev->dev, "error: ipc mandatory ops not defined\n"); + return NULL; + } + + ipc = devm_kzalloc(sdev->dev, sizeof(*ipc), GFP_KERNEL); + if (!ipc) + return NULL; + + mutex_init(&ipc->tx_mutex); + ipc->sdev = sdev; + msg = &ipc->msg; + + /* indicate that we aren't sending a message ATM */ + msg->ipc_complete = true; + + /* pre-allocate message data */ + msg->msg_data = devm_kzalloc(sdev->dev, SOF_IPC_MSG_MAX_SIZE, + GFP_KERNEL); + if (!msg->msg_data) + return NULL; + + msg->reply_data = devm_kzalloc(sdev->dev, SOF_IPC_MSG_MAX_SIZE, + GFP_KERNEL); + if (!msg->reply_data) + return NULL; + + init_waitqueue_head(&msg->waitq); + + return ipc; +} +EXPORT_SYMBOL(snd_sof_ipc_init); + +void snd_sof_ipc_free(struct snd_sof_dev *sdev) +{ + struct snd_sof_ipc *ipc = sdev->ipc; + + /* disable sending of ipc's */ + mutex_lock(&ipc->tx_mutex); + ipc->disable_ipc_tx = true; + mutex_unlock(&ipc->tx_mutex); +} +EXPORT_SYMBOL(snd_sof_ipc_free); -- cgit v1.2.3-59-g8ed1b From 868bd00f495514630a52161052d2fe155af599fe Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:10 -0500 Subject: ASoC: SOF: Add PCM operations support Add support for exposing PCMs to userspace. PCMs are defined by topology and the operations in this patch map to SOF IPC calls. The .get_module_upon_open field is set to allow for module load/unload tests. There is no risk of the sof-pci/acpi-dev module being removed while the platform components are in use. This may need to be revisited when DT platforms are supported. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 719 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 719 insertions(+) create mode 100644 sound/soc/sof/pcm.c (limited to 'sound') diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c new file mode 100644 index 000000000000..be4984c4da4e --- /dev/null +++ b/sound/soc/sof/pcm.c @@ -0,0 +1,719 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// +// PCM Layer, interface between ALSA and IPC. +// + +#include +#include +#include +#include "sof-priv.h" +#include "ops.h" + +#define DRV_NAME "sof-audio-component" + +/* Create DMA buffer page table for DSP */ +static int create_page_table(struct snd_pcm_substream *substream, + unsigned char *dma_area, size_t size) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + struct snd_dma_buffer *dmab = snd_pcm_get_dma_buf(substream); + int stream = substream->stream; + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) + return -EINVAL; + + return snd_sof_create_page_table(sdev, dmab, + spcm->stream[stream].page_table.area, size); +} + +static int sof_pcm_dsp_params(struct snd_sof_pcm *spcm, struct snd_pcm_substream *substream, + const struct sof_ipc_pcm_params_reply *reply) +{ + struct snd_sof_dev *sdev = spcm->sdev; + /* validate offset */ + int ret = snd_sof_ipc_pcm_params(sdev, substream, reply); + + if (ret < 0) + dev_err(sdev->dev, "error: got wrong reply for PCM %d\n", + spcm->pcm.pcm_id); + + return ret; +} + +/* this may get called several times by oss emulation */ +static int sof_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + struct sof_ipc_pcm_params pcm; + struct sof_ipc_pcm_params_reply ipc_params_reply; + int ret; + + /* nothing to do for BE */ + if (rtd->dai_link->no_pcm) + return 0; + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) + return -EINVAL; + + dev_dbg(sdev->dev, "pcm: hw params stream %d dir %d\n", + spcm->pcm.pcm_id, substream->stream); + + memset(&pcm, 0, sizeof(pcm)); + + /* allocate audio buffer pages */ + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) { + dev_err(sdev->dev, "error: could not allocate %d bytes for PCM %d\n", + params_buffer_bytes(params), spcm->pcm.pcm_id); + return ret; + } + if (ret) { + /* + * ret == 1 means the buffer is changed + * create compressed page table for audio firmware + * ret == 0 means the buffer is not changed + * so no need to regenerate the page table + */ + ret = create_page_table(substream, runtime->dma_area, + runtime->dma_bytes); + if (ret < 0) + return ret; + } + + /* number of pages should be rounded up */ + pcm.params.buffer.pages = PFN_UP(runtime->dma_bytes); + + /* set IPC PCM parameters */ + pcm.hdr.size = sizeof(pcm); + pcm.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_PARAMS; + pcm.comp_id = spcm->stream[substream->stream].comp_id; + pcm.params.hdr.size = sizeof(pcm.params); + pcm.params.buffer.phy_addr = + spcm->stream[substream->stream].page_table.addr; + pcm.params.buffer.size = runtime->dma_bytes; + pcm.params.direction = substream->stream; + pcm.params.sample_valid_bytes = params_width(params) >> 3; + pcm.params.buffer_fmt = SOF_IPC_BUFFER_INTERLEAVED; + pcm.params.rate = params_rate(params); + pcm.params.channels = params_channels(params); + pcm.params.host_period_bytes = params_period_bytes(params); + + /* container size */ + ret = snd_pcm_format_physical_width(params_format(params)); + if (ret < 0) + return ret; + pcm.params.sample_container_bytes = ret >> 3; + + /* format */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16: + pcm.params.frame_fmt = SOF_IPC_FRAME_S16_LE; + break; + case SNDRV_PCM_FORMAT_S24: + pcm.params.frame_fmt = SOF_IPC_FRAME_S24_4LE; + break; + case SNDRV_PCM_FORMAT_S32: + pcm.params.frame_fmt = SOF_IPC_FRAME_S32_LE; + break; + case SNDRV_PCM_FORMAT_FLOAT: + pcm.params.frame_fmt = SOF_IPC_FRAME_FLOAT; + break; + default: + return -EINVAL; + } + + /* firmware already configured host stream */ + ret = snd_sof_pcm_platform_hw_params(sdev, + substream, + params, + &pcm.params); + if (ret < 0) { + dev_err(sdev->dev, "error: platform hw params failed\n"); + return ret; + } + + dev_dbg(sdev->dev, "stream_tag %d", pcm.params.stream_tag); + + /* send IPC to the DSP */ + ret = sof_ipc_tx_message(sdev->ipc, pcm.hdr.cmd, &pcm, sizeof(pcm), + &ipc_params_reply, sizeof(ipc_params_reply)); + if (ret < 0) { + dev_err(sdev->dev, "error: hw params ipc failed for stream %d\n", + pcm.params.stream_tag); + return ret; + } + + ret = sof_pcm_dsp_params(spcm, substream, &ipc_params_reply); + if (ret < 0) + return ret; + + /* save pcm hw_params */ + memcpy(&spcm->params[substream->stream], params, sizeof(*params)); + + return ret; +} + +static int sof_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + int ret; + + /* nothing to do for BE */ + if (rtd->dai_link->no_pcm) + return 0; + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) + return -EINVAL; + + dev_dbg(sdev->dev, "pcm: free stream %d dir %d\n", spcm->pcm.pcm_id, + substream->stream); + + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE; + stream.comp_id = spcm->stream[substream->stream].comp_id; + + /* send IPC to the DSP */ + ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream, + sizeof(stream), &reply, sizeof(reply)); + + snd_pcm_lib_free_pages(substream); + return ret; +} + +static int sof_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + int ret; + + /* nothing to do for BE */ + if (rtd->dai_link->no_pcm) + return 0; + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) + return -EINVAL; + + /* + * check if hw_params needs to be set-up again. + * This is only needed when resuming from system sleep. + */ + if (!spcm->hw_params_upon_resume[substream->stream]) + return 0; + + dev_dbg(sdev->dev, "pcm: prepare stream %d dir %d\n", spcm->pcm.pcm_id, + substream->stream); + + /* set hw_params */ + ret = sof_pcm_hw_params(substream, &spcm->params[substream->stream]); + if (ret < 0) { + dev_err(sdev->dev, "error: set pcm hw_params after resume\n"); + return ret; + } + + return 0; +} + +/* + * FE dai link trigger actions are always executed in non-atomic context because + * they involve IPC's. + */ +static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + int ret; + + /* nothing to do for BE */ + if (rtd->dai_link->no_pcm) + return 0; + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) + return -EINVAL; + + dev_dbg(sdev->dev, "pcm: trigger stream %d dir %d cmd %d\n", + spcm->pcm.pcm_id, substream->stream, cmd); + + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG; + stream.comp_id = spcm->stream[substream->stream].comp_id; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE; + break; + case SNDRV_PCM_TRIGGER_RESUME: + /* set up hw_params */ + ret = sof_pcm_prepare(substream); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to set up hw_params upon resume\n"); + return ret; + } + + /* fallthrough */ + case SNDRV_PCM_TRIGGER_START: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_START; + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP; + break; + default: + dev_err(sdev->dev, "error: unhandled trigger cmd %d\n", cmd); + return -EINVAL; + } + + snd_sof_pcm_platform_trigger(sdev, substream, cmd); + + /* send IPC to the DSP */ + ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream, + sizeof(stream), &reply, sizeof(reply)); + + if (ret < 0 || cmd != SNDRV_PCM_TRIGGER_SUSPEND) + return ret; + + /* + * The hw_free op is usually called when the pcm stream is closed. + * Since the stream is not closed during suspend, the DSP needs to be + * notified explicitly to free pcm to prevent errors upon resume. + */ + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE; + stream.comp_id = spcm->stream[substream->stream].comp_id; + + /* send IPC to the DSP */ + return sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream, + sizeof(stream), &reply, sizeof(reply)); +} + +static snd_pcm_uframes_t sof_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + snd_pcm_uframes_t host, dai; + + /* nothing to do for BE */ + if (rtd->dai_link->no_pcm) + return 0; + + /* use dsp ops pointer callback directly if set */ + if (sof_ops(sdev)->pcm_pointer) + return sof_ops(sdev)->pcm_pointer(sdev, substream); + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) + return -EINVAL; + + /* read position from DSP */ + host = bytes_to_frames(substream->runtime, + spcm->stream[substream->stream].posn.host_posn); + dai = bytes_to_frames(substream->runtime, + spcm->stream[substream->stream].posn.dai_posn); + + dev_dbg(sdev->dev, "PCM: stream %d dir %d DMA position %lu DAI position %lu\n", + spcm->pcm.pcm_id, substream->stream, host, dai); + + return host; +} + +static int sof_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + struct snd_soc_tplg_stream_caps *caps; + int ret; + int err; + + /* nothing to do for BE */ + if (rtd->dai_link->no_pcm) + return 0; + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) + return -EINVAL; + + dev_dbg(sdev->dev, "pcm: open stream %d dir %d\n", spcm->pcm.pcm_id, + substream->stream); + + /* clear hw_params_upon_resume flag */ + spcm->hw_params_upon_resume[substream->stream] = 0; + + caps = &spcm->pcm.caps[substream->stream]; + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err(sdev->dev, "error: pcm open failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* set any runtime constraints based on topology */ + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + le32_to_cpu(caps->period_size_min)); + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + le32_to_cpu(caps->period_size_min)); + + /* set runtime config */ + runtime->hw.info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP; + runtime->hw.formats = le64_to_cpu(caps->formats); + runtime->hw.period_bytes_min = le32_to_cpu(caps->period_size_min); + runtime->hw.period_bytes_max = le32_to_cpu(caps->period_size_max); + runtime->hw.periods_min = le32_to_cpu(caps->periods_min); + runtime->hw.periods_max = le32_to_cpu(caps->periods_max); + + /* + * caps->buffer_size_min is not used since the + * snd_pcm_hardware structure only defines buffer_bytes_max + */ + runtime->hw.buffer_bytes_max = le32_to_cpu(caps->buffer_size_max); + + dev_dbg(sdev->dev, "period min %zd max %zd bytes\n", + runtime->hw.period_bytes_min, + runtime->hw.period_bytes_max); + dev_dbg(sdev->dev, "period count %d max %d\n", + runtime->hw.periods_min, + runtime->hw.periods_max); + dev_dbg(sdev->dev, "buffer max %zd bytes\n", + runtime->hw.buffer_bytes_max); + + /* set wait time - TODO: come from topology */ + substream->wait_time = 500; + + spcm->stream[substream->stream].posn.host_posn = 0; + spcm->stream[substream->stream].posn.dai_posn = 0; + spcm->stream[substream->stream].substream = substream; + + ret = snd_sof_pcm_platform_open(sdev, substream); + if (ret < 0) { + dev_err(sdev->dev, "error: pcm open failed %d\n", + ret); + + pm_runtime_mark_last_busy(sdev->dev); + + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err(sdev->dev, "error: pcm close failed to idle %d\n", + err); + } + + return ret; +} + +static int sof_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + int err; + + /* nothing to do for BE */ + if (rtd->dai_link->no_pcm) + return 0; + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) + return -EINVAL; + + dev_dbg(sdev->dev, "pcm: close stream %d dir %d\n", spcm->pcm.pcm_id, + substream->stream); + + err = snd_sof_pcm_platform_close(sdev, substream); + if (err < 0) { + dev_err(sdev->dev, "error: pcm close failed %d\n", + err); + /* + * keep going, no point in preventing the close + * from happening + */ + } + + pm_runtime_mark_last_busy(sdev->dev); + + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err(sdev->dev, "error: pcm close failed to idle %d\n", + err); + + return 0; +} + +static struct snd_pcm_ops sof_pcm_ops = { + .open = sof_pcm_open, + .close = sof_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sof_pcm_hw_params, + .prepare = sof_pcm_prepare, + .hw_free = sof_pcm_hw_free, + .trigger = sof_pcm_trigger, + .pointer = sof_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + +/* + * Pre-allocate playback/capture audio buffer pages. + * no need to explicitly release memory preallocated by sof_pcm_new in pcm_free + * snd_pcm_lib_preallocate_free_for_all() is called by the core. + */ +static int sof_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + struct snd_pcm *pcm = rtd->pcm; + struct snd_soc_tplg_stream_caps *caps; + int stream = SNDRV_PCM_STREAM_PLAYBACK; + + /* find SOF PCM for this RTD */ + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) { + dev_warn(sdev->dev, "warn: can't find PCM with DAI ID %d\n", + rtd->dai_link->id); + return 0; + } + + dev_dbg(sdev->dev, "creating new PCM %s\n", spcm->pcm.pcm_name); + + /* do we need to pre-allocate playback audio buffer pages */ + if (!spcm->pcm.playback) + goto capture; + + caps = &spcm->pcm.caps[stream]; + + /* pre-allocate playback audio buffer pages */ + dev_dbg(sdev->dev, "spcm: allocate %s playback DMA buffer size 0x%x max 0x%x\n", + caps->name, caps->buffer_size_min, caps->buffer_size_max); + + snd_pcm_lib_preallocate_pages(pcm->streams[stream].substream, + SNDRV_DMA_TYPE_DEV_SG, sdev->dev, + le32_to_cpu(caps->buffer_size_min), + le32_to_cpu(caps->buffer_size_max)); +capture: + stream = SNDRV_PCM_STREAM_CAPTURE; + + /* do we need to pre-allocate capture audio buffer pages */ + if (!spcm->pcm.capture) + return 0; + + caps = &spcm->pcm.caps[stream]; + + /* pre-allocate capture audio buffer pages */ + dev_dbg(sdev->dev, "spcm: allocate %s capture DMA buffer size 0x%x max 0x%x\n", + caps->name, caps->buffer_size_min, caps->buffer_size_max); + + snd_pcm_lib_preallocate_pages(pcm->streams[stream].substream, + SNDRV_DMA_TYPE_DEV_SG, sdev->dev, + le32_to_cpu(caps->buffer_size_min), + le32_to_cpu(caps->buffer_size_max)); + + return 0; +} + +/* fixup the BE DAI link to match any values from topology */ +static int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_dai *dai = + snd_sof_find_dai(sdev, (char *)rtd->dai_link->name); + + /* no topology exists for this BE, try a common configuration */ + if (!dai) { + dev_warn(sdev->dev, "warning: no topology found for BE DAI %s config\n", + rtd->dai_link->name); + + /* set 48k, stereo, 16bits by default */ + rate->min = 48000; + rate->max = 48000; + + channels->min = 2; + channels->max = 2; + + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + + return 0; + } + + /* read format from topology */ + snd_mask_none(fmt); + + switch (dai->comp_dai.config.frame_fmt) { + case SOF_IPC_FRAME_S16_LE: + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + break; + case SOF_IPC_FRAME_S24_4LE: + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + break; + case SOF_IPC_FRAME_S32_LE: + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S32_LE); + break; + default: + dev_err(sdev->dev, "error: No available DAI format!\n"); + return -EINVAL; + } + + /* read rate and channels from topology */ + switch (dai->dai_config->type) { + case SOF_DAI_INTEL_SSP: + rate->min = dai->dai_config->ssp.fsync_rate; + rate->max = dai->dai_config->ssp.fsync_rate; + channels->min = dai->dai_config->ssp.tdm_slots; + channels->max = dai->dai_config->ssp.tdm_slots; + + dev_dbg(sdev->dev, + "rate_min: %d rate_max: %d\n", rate->min, rate->max); + dev_dbg(sdev->dev, + "channels_min: %d channels_max: %d\n", + channels->min, channels->max); + + break; + case SOF_DAI_INTEL_DMIC: + /* DMIC only supports 16 or 32 bit formats */ + if (dai->comp_dai.config.frame_fmt == SOF_IPC_FRAME_S24_4LE) { + dev_err(sdev->dev, + "error: invalid fmt %d for DAI type %d\n", + dai->comp_dai.config.frame_fmt, + dai->dai_config->type); + } + break; + case SOF_DAI_INTEL_HDA: + /* do nothing for HDA dai_link */ + break; + default: + dev_err(sdev->dev, "error: invalid DAI type %d\n", + dai->dai_config->type); + break; + } + + return 0; +} + +static int sof_pcm_probe(struct snd_soc_component *component) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pdata *plat_data = sdev->pdata; + const char *tplg_filename; + int ret; + + /* load the default topology */ + sdev->component = component; + + tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, + "%s/%s", + plat_data->tplg_filename_prefix, + plat_data->tplg_filename); + if (!tplg_filename) + return -ENOMEM; + + ret = snd_sof_load_topology(sdev, tplg_filename); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to load DSP topology %d\n", + ret); + return ret; + } + + /* + * Some platforms in SOF, ex: BYT, may not have their platform PM + * callbacks set. Increment the usage count so as to + * prevent the device from entering runtime suspend. + */ + if (!sof_ops(sdev)->runtime_suspend || !sof_ops(sdev)->runtime_resume) + pm_runtime_get_noresume(sdev->dev); + + return ret; +} + +static void sof_pcm_remove(struct snd_soc_component *component) +{ + /* remove topology */ + snd_soc_tplg_component_remove(component, SND_SOC_TPLG_INDEX_ALL); +} + +void snd_sof_new_platform_drv(struct snd_sof_dev *sdev) +{ + struct snd_soc_component_driver *pd = &sdev->plat_drv; + struct snd_sof_pdata *plat_data = sdev->pdata; + const char *drv_name; + + drv_name = plat_data->machine->drv_name; + + pd->name = "sof-audio-component"; + pd->probe = sof_pcm_probe; + pd->remove = sof_pcm_remove; + pd->ops = &sof_pcm_ops; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMPRESS) + pd->compr_ops = &sof_compressed_ops; +#endif + pd->pcm_new = sof_pcm_new; + pd->ignore_machine = drv_name; + pd->be_hw_params_fixup = sof_pcm_dai_link_fixup; + pd->be_pcm_base = SOF_BE_PCM_BASE; + pd->use_dai_pcm_id = true; + pd->topology_name_prefix = "sof"; + + /* increment module refcount when a pcm is opened */ + pd->module_get_upon_open = 1; +} -- cgit v1.2.3-59-g8ed1b From 311ce4fe7637d96608b6e57bf9ebbd8aabcf429e Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:11 -0500 Subject: ASoC: SOF: Add support for loading topologies SOF uses topology to define the DAPM graphs and widgets, DAIs, PCMs and set parameters for init and run time usage. This patch loads topology and maps it to IPC commands that are build the topology on the DSP. Signed-off-by: Ranjani Sridharan Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/sof/topology.h | 256 ++++ sound/soc/sof/topology.c | 3164 ++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 3420 insertions(+) create mode 100644 include/sound/sof/topology.h create mode 100644 sound/soc/sof/topology.c (limited to 'sound') diff --git a/include/sound/sof/topology.h b/include/sound/sof/topology.h new file mode 100644 index 000000000000..46b2a7e63167 --- /dev/null +++ b/include/sound/sof/topology.h @@ -0,0 +1,256 @@ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_TOPOLOGY_H__ +#define __INCLUDE_SOUND_SOF_TOPOLOGY_H__ + +#include + +/* + * Component + */ + +/* types of component */ +enum sof_comp_type { + SOF_COMP_NONE = 0, + SOF_COMP_HOST, + SOF_COMP_DAI, + SOF_COMP_SG_HOST, /**< scatter gather variant */ + SOF_COMP_SG_DAI, /**< scatter gather variant */ + SOF_COMP_VOLUME, + SOF_COMP_MIXER, + SOF_COMP_MUX, + SOF_COMP_SRC, + SOF_COMP_SPLITTER, + SOF_COMP_TONE, + SOF_COMP_SWITCH, + SOF_COMP_BUFFER, + SOF_COMP_EQ_IIR, + SOF_COMP_EQ_FIR, + SOF_COMP_KEYWORD_DETECT, + SOF_COMP_KPB, /* A key phrase buffer component */ + SOF_COMP_SELECTOR, /**< channel selector component */ + /* keep FILEREAD/FILEWRITE as the last ones */ + SOF_COMP_FILEREAD = 10000, /**< host test based file IO */ + SOF_COMP_FILEWRITE = 10001, /**< host test based file IO */ +}; + +/* XRUN action for component */ +#define SOF_XRUN_STOP 1 /**< stop stream */ +#define SOF_XRUN_UNDER_ZERO 2 /**< send 0s to sink */ +#define SOF_XRUN_OVER_NULL 4 /**< send data to NULL */ + +/* create new generic component - SOF_IPC_TPLG_COMP_NEW */ +struct sof_ipc_comp { + struct sof_ipc_cmd_hdr hdr; + uint32_t id; + enum sof_comp_type type; + uint32_t pipeline_id; + + /* reserved for future use */ + uint32_t reserved[2]; +} __packed; + +/* + * Component Buffers + */ + +/* + * SOF memory capabilities, add new ones at the end + */ +#define SOF_MEM_CAPS_RAM (1 << 0) +#define SOF_MEM_CAPS_ROM (1 << 1) +#define SOF_MEM_CAPS_EXT (1 << 2) /**< external */ +#define SOF_MEM_CAPS_LP (1 << 3) /**< low power */ +#define SOF_MEM_CAPS_HP (1 << 4) /**< high performance */ +#define SOF_MEM_CAPS_DMA (1 << 5) /**< DMA'able */ +#define SOF_MEM_CAPS_CACHE (1 << 6) /**< cacheable */ +#define SOF_MEM_CAPS_EXEC (1 << 7) /**< executable */ + +/* create new component buffer - SOF_IPC_TPLG_BUFFER_NEW */ +struct sof_ipc_buffer { + struct sof_ipc_comp comp; + uint32_t size; /**< buffer size in bytes */ + uint32_t caps; /**< SOF_MEM_CAPS_ */ +} __packed; + +/* generic component config data - must always be after struct sof_ipc_comp */ +struct sof_ipc_comp_config { + struct sof_ipc_cmd_hdr hdr; + uint32_t periods_sink; /**< 0 means variable */ + uint32_t periods_source; /**< 0 means variable */ + uint32_t reserved1; /**< reserved */ + uint32_t frame_fmt; /**< SOF_IPC_FRAME_ */ + uint32_t xrun_action; + + /* reserved for future use */ + uint32_t reserved[2]; +} __packed; + +/* generic host component */ +struct sof_ipc_comp_host { + struct sof_ipc_comp comp; + struct sof_ipc_comp_config config; + uint32_t direction; /**< SOF_IPC_STREAM_ */ + uint32_t no_irq; /**< don't send periodic IRQ to host/DSP */ + uint32_t dmac_config; /**< DMA engine specific */ +} __packed; + +/* generic DAI component */ +struct sof_ipc_comp_dai { + struct sof_ipc_comp comp; + struct sof_ipc_comp_config config; + uint32_t direction; /**< SOF_IPC_STREAM_ */ + uint32_t dai_index; /**< index of this type dai */ + uint32_t type; /**< DAI type - SOF_DAI_ */ + uint32_t reserved; /**< reserved */ +} __packed; + +/* generic mixer component */ +struct sof_ipc_comp_mixer { + struct sof_ipc_comp comp; + struct sof_ipc_comp_config config; +} __packed; + +/* volume ramping types */ +enum sof_volume_ramp { + SOF_VOLUME_LINEAR = 0, + SOF_VOLUME_LOG, + SOF_VOLUME_LINEAR_ZC, + SOF_VOLUME_LOG_ZC, +}; + +/* generic volume component */ +struct sof_ipc_comp_volume { + struct sof_ipc_comp comp; + struct sof_ipc_comp_config config; + uint32_t channels; + uint32_t min_value; + uint32_t max_value; + uint32_t ramp; /**< SOF_VOLUME_ */ + uint32_t initial_ramp; /**< ramp space in ms */ +} __packed; + +/* generic SRC component */ +struct sof_ipc_comp_src { + struct sof_ipc_comp comp; + struct sof_ipc_comp_config config; + /* either source or sink rate must be non zero */ + uint32_t source_rate; /**< source rate or 0 for variable */ + uint32_t sink_rate; /**< sink rate or 0 for variable */ + uint32_t rate_mask; /**< SOF_RATE_ supported rates */ +} __packed; + +/* generic MUX component */ +struct sof_ipc_comp_mux { + struct sof_ipc_comp comp; + struct sof_ipc_comp_config config; +} __packed; + +/* generic tone generator component */ +struct sof_ipc_comp_tone { + struct sof_ipc_comp comp; + struct sof_ipc_comp_config config; + int32_t sample_rate; + int32_t frequency; + int32_t amplitude; + int32_t freq_mult; + int32_t ampl_mult; + int32_t length; + int32_t period; + int32_t repeats; + int32_t ramp_step; +} __packed; + +/** \brief Types of processing components */ +enum sof_ipc_process_type { + SOF_PROCESS_NONE = 0, /**< None */ + SOF_PROCESS_EQFIR, /**< Intel FIR */ + SOF_PROCESS_EQIIR, /**< Intel IIR */ + SOF_PROCESS_KEYWORD_DETECT, /**< Keyword Detection */ + SOF_PROCESS_KPB, /**< KeyPhrase Buffer Manager */ + SOF_PROCESS_CHAN_SELECTOR, /**< Channel Selector */ +}; + +/* generic "effect", "codec" or proprietary processing component */ +struct sof_ipc_comp_process { + struct sof_ipc_comp comp; + struct sof_ipc_comp_config config; + uint32_t size; /**< size of bespoke data section in bytes */ + uint32_t type; /**< sof_ipc_process_type */ + + /* reserved for future use */ + uint32_t reserved[7]; + + unsigned char data[0]; +} __packed; + +/* frees components, buffers and pipelines + * SOF_IPC_TPLG_COMP_FREE, SOF_IPC_TPLG_PIPE_FREE, SOF_IPC_TPLG_BUFFER_FREE + */ +struct sof_ipc_free { + struct sof_ipc_cmd_hdr hdr; + uint32_t id; +} __packed; + +struct sof_ipc_comp_reply { + struct sof_ipc_reply rhdr; + uint32_t id; + uint32_t offset; +} __packed; + +/* + * Pipeline + */ + +/** \brief Types of pipeline scheduling time domains */ +enum sof_ipc_pipe_sched_time_domain { + SOF_TIME_DOMAIN_DMA = 0, /**< DMA interrupt */ + SOF_TIME_DOMAIN_TIMER, /**< Timer interrupt */ +}; + +/* new pipeline - SOF_IPC_TPLG_PIPE_NEW */ +struct sof_ipc_pipe_new { + struct sof_ipc_cmd_hdr hdr; + uint32_t comp_id; /**< component id for pipeline */ + uint32_t pipeline_id; /**< pipeline id */ + uint32_t sched_id; /**< Scheduling component id */ + uint32_t core; /**< core we run on */ + uint32_t period; /**< execution period in us*/ + uint32_t priority; /**< priority level 0 (low) to 10 (max) */ + uint32_t period_mips; /**< worst case instruction count per period */ + uint32_t frames_per_sched;/**< output frames of pipeline, 0 is variable */ + uint32_t xrun_limit_usecs; /**< report xruns greater than limit */ + uint32_t time_domain; /**< scheduling time domain */ +} __packed; + +/* pipeline construction complete - SOF_IPC_TPLG_PIPE_COMPLETE */ +struct sof_ipc_pipe_ready { + struct sof_ipc_cmd_hdr hdr; + uint32_t comp_id; +} __packed; + +struct sof_ipc_pipe_free { + struct sof_ipc_cmd_hdr hdr; + uint32_t comp_id; +} __packed; + +/* connect two components in pipeline - SOF_IPC_TPLG_COMP_CONNECT */ +struct sof_ipc_pipe_comp_connect { + struct sof_ipc_cmd_hdr hdr; + uint32_t source_id; + uint32_t sink_id; +} __packed; + +/* external events */ +enum sof_event_types { + SOF_EVENT_NONE = 0, + SOF_KEYWORD_DETECT_DAPM_EVENT, +}; + +#endif diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c new file mode 100644 index 000000000000..2b9de1b97447 --- /dev/null +++ b/sound/soc/sof/topology.c @@ -0,0 +1,3164 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +#include +#include +#include +#include +#include "sof-priv.h" +#include "ops.h" + +#define COMP_ID_UNASSIGNED 0xffffffff +/* + * Constants used in the computation of linear volume gain + * from dB gain 20th root of 10 in Q1.16 fixed-point notation + */ +#define VOL_TWENTIETH_ROOT_OF_TEN 73533 +/* 40th root of 10 in Q1.16 fixed-point notation*/ +#define VOL_FORTIETH_ROOT_OF_TEN 69419 +/* + * Volume fractional word length define to 16 sets + * the volume linear gain value to use Qx.16 format + */ +#define VOLUME_FWL 16 +/* 0.5 dB step value in topology TLV */ +#define VOL_HALF_DB_STEP 50 +/* Full volume for default values */ +#define VOL_ZERO_DB BIT(VOLUME_FWL) + +/* TLV data items */ +#define TLV_ITEMS 3 +#define TLV_MIN 0 +#define TLV_STEP 1 +#define TLV_MUTE 2 + +/* size of tplg abi in byte */ +#define SOF_TPLG_ABI_SIZE 3 + +/* send pcm params ipc */ +static int ipc_pcm_params(struct snd_sof_widget *swidget, int dir) +{ + struct sof_ipc_pcm_params_reply ipc_params_reply; + struct snd_sof_dev *sdev = swidget->sdev; + struct sof_ipc_pcm_params pcm; + struct snd_pcm_hw_params *params; + struct snd_sof_pcm *spcm; + int ret = 0; + + memset(&pcm, 0, sizeof(pcm)); + + /* get runtime PCM params using widget's stream name */ + spcm = snd_sof_find_spcm_name(sdev, swidget->widget->sname); + if (!spcm) { + dev_err(sdev->dev, "error: cannot find PCM for %s\n", + swidget->widget->name); + return -EINVAL; + } + + params = &spcm->params[dir]; + + /* set IPC PCM params */ + pcm.hdr.size = sizeof(pcm); + pcm.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_PARAMS; + pcm.comp_id = swidget->comp_id; + pcm.params.hdr.size = sizeof(pcm.params); + pcm.params.direction = dir; + pcm.params.sample_valid_bytes = params_width(params) >> 3; + pcm.params.buffer_fmt = SOF_IPC_BUFFER_INTERLEAVED; + pcm.params.rate = params_rate(params); + pcm.params.channels = params_channels(params); + pcm.params.host_period_bytes = params_period_bytes(params); + + /* set format */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16: + pcm.params.frame_fmt = SOF_IPC_FRAME_S16_LE; + break; + case SNDRV_PCM_FORMAT_S24: + pcm.params.frame_fmt = SOF_IPC_FRAME_S24_4LE; + break; + case SNDRV_PCM_FORMAT_S32: + pcm.params.frame_fmt = SOF_IPC_FRAME_S32_LE; + break; + default: + return -EINVAL; + } + + /* send IPC to the DSP */ + ret = sof_ipc_tx_message(sdev->ipc, pcm.hdr.cmd, &pcm, sizeof(pcm), + &ipc_params_reply, sizeof(ipc_params_reply)); + if (ret < 0) + dev_err(sdev->dev, "error: pcm params failed for %s\n", + swidget->widget->name); + + return ret; +} + + /* send stream trigger ipc */ +static int ipc_trigger(struct snd_sof_widget *swidget, int cmd) +{ + struct snd_sof_dev *sdev = swidget->sdev; + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + int ret = 0; + + /* set IPC stream params */ + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | cmd; + stream.comp_id = swidget->comp_id; + + /* send IPC to the DSP */ + ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream, + sizeof(stream), &reply, sizeof(reply)); + if (ret < 0) + dev_err(sdev->dev, "error: failed to trigger %s\n", + swidget->widget->name); + + return ret; +} + +static int sof_keyword_dapm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_sof_widget *swidget = w->dobj.private; + struct snd_sof_dev *sdev; + int ret = 0; + + if (!swidget) + return 0; + + sdev = swidget->sdev; + + dev_dbg(sdev->dev, "received event %d for widget %s\n", + event, w->name); + + /* process events */ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* set pcm params */ + ret = ipc_pcm_params(swidget, SOF_IPC_STREAM_CAPTURE); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to set pcm params for widget %s\n", + swidget->widget->name); + break; + } + + /* start trigger */ + ret = ipc_trigger(swidget, SOF_IPC_STREAM_TRIG_START); + if (ret < 0) + dev_err(sdev->dev, + "error: failed to trigger widget %s\n", + swidget->widget->name); + break; + case SND_SOC_DAPM_POST_PMD: + /* stop trigger */ + ret = ipc_trigger(swidget, SOF_IPC_STREAM_TRIG_STOP); + if (ret < 0) + dev_err(sdev->dev, + "error: failed to trigger widget %s\n", + swidget->widget->name); + + /* pcm free */ + ret = ipc_trigger(swidget, SOF_IPC_STREAM_PCM_FREE); + if (ret < 0) + dev_err(sdev->dev, + "error: failed to trigger widget %s\n", + swidget->widget->name); + break; + default: + break; + } + + return ret; +} + +/* event handlers for keyword detect component */ +static const struct snd_soc_tplg_widget_events sof_kwd_events[] = { + {SOF_KEYWORD_DETECT_DAPM_EVENT, sof_keyword_dapm_event}, +}; + +static inline int get_tlv_data(const int *p, int tlv[TLV_ITEMS]) +{ + /* we only support dB scale TLV type at the moment */ + if ((int)p[SNDRV_CTL_TLVO_TYPE] != SNDRV_CTL_TLVT_DB_SCALE) + return -EINVAL; + + /* min value in topology tlv data is multiplied by 100 */ + tlv[TLV_MIN] = (int)p[SNDRV_CTL_TLVO_DB_SCALE_MIN] / 100; + + /* volume steps */ + tlv[TLV_STEP] = (int)(p[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] & + TLV_DB_SCALE_MASK); + + /* mute ON/OFF */ + if ((p[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] & + TLV_DB_SCALE_MUTE) == 0) + tlv[TLV_MUTE] = 0; + else + tlv[TLV_MUTE] = 1; + + return 0; +} + +/* + * Function to truncate an unsigned 64-bit number + * by x bits and return 32-bit unsigned number. This + * function also takes care of rounding while truncating + */ +static inline u32 vol_shift_64(u64 i, u32 x) +{ + /* do not truncate more than 32 bits */ + if (x > 32) + x = 32; + + if (x == 0) + return (u32)i; + + return (u32)(((i >> (x - 1)) + 1) >> 1); +} + +/* + * Function to compute a ^ exp where, + * a is a fractional number represented by a fixed-point + * integer with a fractional world length of "fwl" + * exp is an integer + * fwl is the fractional word length + * Return value is a fractional number represented by a + * fixed-point integer with a fractional word length of "fwl" + */ +static u32 vol_pow32(u32 a, int exp, u32 fwl) +{ + int i, iter; + u32 power = 1 << fwl; + u64 numerator; + + /* if exponent is 0, return 1 */ + if (exp == 0) + return power; + + /* determine the number of iterations based on the exponent */ + if (exp < 0) + iter = exp * -1; + else + iter = exp; + + /* mutiply a "iter" times to compute power */ + for (i = 0; i < iter; i++) { + /* + * Product of 2 Qx.fwl fixed-point numbers yields a Q2*x.2*fwl + * Truncate product back to fwl fractional bits with rounding + */ + power = vol_shift_64((u64)power * a, fwl); + } + + if (exp > 0) { + /* if exp is positive, return the result */ + return power; + } + + /* if exp is negative, return the multiplicative inverse */ + numerator = (u64)1 << (fwl << 1); + do_div(numerator, power); + + return (u32)numerator; +} + +/* + * Function to calculate volume gain from TLV data. + * This function can only handle gain steps that are multiples of 0.5 dB + */ +static u32 vol_compute_gain(u32 value, int *tlv) +{ + int dB_gain; + u32 linear_gain; + int f_step; + + /* mute volume */ + if (value == 0 && tlv[TLV_MUTE]) + return 0; + + /* + * compute dB gain from tlv. tlv_step + * in topology is multiplied by 100 + */ + dB_gain = tlv[TLV_MIN] + (value * tlv[TLV_STEP]) / 100; + + /* + * compute linear gain represented by fixed-point + * int with VOLUME_FWL fractional bits + */ + linear_gain = vol_pow32(VOL_TWENTIETH_ROOT_OF_TEN, dB_gain, VOLUME_FWL); + + /* extract the fractional part of volume step */ + f_step = tlv[TLV_STEP] - (tlv[TLV_STEP] / 100); + + /* if volume step is an odd multiple of 0.5 dB */ + if (f_step == VOL_HALF_DB_STEP && (value & 1)) + linear_gain = vol_shift_64((u64)linear_gain * + VOL_FORTIETH_ROOT_OF_TEN, + VOLUME_FWL); + + return linear_gain; +} + +/* + * Set up volume table for kcontrols from tlv data + * "size" specifies the number of entries in the table + */ +static int set_up_volume_table(struct snd_sof_control *scontrol, + int tlv[TLV_ITEMS], int size) +{ + int j; + + /* init the volume table */ + scontrol->volume_table = kcalloc(size, sizeof(u32), GFP_KERNEL); + if (!scontrol->volume_table) + return -ENOMEM; + + /* populate the volume table */ + for (j = 0; j < size ; j++) + scontrol->volume_table[j] = vol_compute_gain(j, tlv); + + return 0; +} + +struct sof_dai_types { + const char *name; + enum sof_ipc_dai_type type; +}; + +static const struct sof_dai_types sof_dais[] = { + {"SSP", SOF_DAI_INTEL_SSP}, + {"HDA", SOF_DAI_INTEL_HDA}, + {"DMIC", SOF_DAI_INTEL_DMIC}, +}; + +static enum sof_ipc_dai_type find_dai(const char *name) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(sof_dais); i++) { + if (strcmp(name, sof_dais[i].name) == 0) + return sof_dais[i].type; + } + + return SOF_DAI_INTEL_NONE; +} + +/* + * Supported Frame format types and lookup, add new ones to end of list. + */ + +struct sof_frame_types { + const char *name; + enum sof_ipc_frame frame; +}; + +static const struct sof_frame_types sof_frames[] = { + {"s16le", SOF_IPC_FRAME_S16_LE}, + {"s24le", SOF_IPC_FRAME_S24_4LE}, + {"s32le", SOF_IPC_FRAME_S32_LE}, + {"float", SOF_IPC_FRAME_FLOAT}, +}; + +static enum sof_ipc_frame find_format(const char *name) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(sof_frames); i++) { + if (strcmp(name, sof_frames[i].name) == 0) + return sof_frames[i].frame; + } + + /* use s32le if nothing is specified */ + return SOF_IPC_FRAME_S32_LE; +} + +struct sof_process_types { + const char *name; + enum sof_ipc_process_type type; + enum sof_comp_type comp_type; +}; + +static const struct sof_process_types sof_process[] = { + {"EQFIR", SOF_PROCESS_EQFIR, SOF_COMP_EQ_FIR}, + {"EQIIR", SOF_PROCESS_EQIIR, SOF_COMP_EQ_IIR}, + {"KEYWORD_DETECT", SOF_PROCESS_KEYWORD_DETECT, SOF_COMP_KEYWORD_DETECT}, + {"KPB", SOF_PROCESS_KPB, SOF_COMP_KPB}, + {"CHAN_SELECTOR", SOF_PROCESS_CHAN_SELECTOR, SOF_COMP_SELECTOR}, +}; + +static enum sof_ipc_process_type find_process(const char *name) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(sof_process); i++) { + if (strcmp(name, sof_process[i].name) == 0) + return sof_process[i].type; + } + + return SOF_PROCESS_NONE; +} + +static enum sof_comp_type find_process_comp_type(enum sof_ipc_process_type type) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(sof_process); i++) { + if (sof_process[i].type == type) + return sof_process[i].comp_type; + } + + return SOF_COMP_NONE; +} + +/* + * Standard Kcontrols. + */ + +static int sof_control_load_volume(struct snd_soc_component *scomp, + struct snd_sof_control *scontrol, + struct snd_kcontrol_new *kc, + struct snd_soc_tplg_ctl_hdr *hdr) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_mixer_control *mc = + container_of(hdr, struct snd_soc_tplg_mixer_control, hdr); + struct sof_ipc_ctrl_data *cdata; + int tlv[TLV_ITEMS]; + unsigned int i; + int ret; + + /* validate topology data */ + if (le32_to_cpu(mc->num_channels) > SND_SOC_TPLG_MAX_CHAN) + return -EINVAL; + + /* init the volume get/put data */ + scontrol->size = sizeof(struct sof_ipc_ctrl_data) + + sizeof(struct sof_ipc_ctrl_value_chan) * + le32_to_cpu(mc->num_channels); + scontrol->control_data = kzalloc(scontrol->size, GFP_KERNEL); + if (!scontrol->control_data) + return -ENOMEM; + + scontrol->comp_id = sdev->next_comp_id; + scontrol->num_channels = le32_to_cpu(mc->num_channels); + + /* set cmd for mixer control */ + if (le32_to_cpu(mc->max) == 1) { + scontrol->cmd = SOF_CTRL_CMD_SWITCH; + goto out; + } + + scontrol->cmd = SOF_CTRL_CMD_VOLUME; + + /* extract tlv data */ + if (get_tlv_data(kc->tlv.p, tlv) < 0) { + dev_err(sdev->dev, "error: invalid TLV data\n"); + return -EINVAL; + } + + /* set up volume table */ + ret = set_up_volume_table(scontrol, tlv, le32_to_cpu(mc->max) + 1); + if (ret < 0) { + dev_err(sdev->dev, "error: setting up volume table\n"); + return ret; + } + + /* set default volume values to 0dB in control */ + cdata = scontrol->control_data; + for (i = 0; i < scontrol->num_channels; i++) { + cdata->chanv[i].channel = i; + cdata->chanv[i].value = VOL_ZERO_DB; + } + +out: + dev_dbg(sdev->dev, "tplg: load kcontrol index %d chans %d\n", + scontrol->comp_id, scontrol->num_channels); + + return 0; +} + +static int sof_control_load_enum(struct snd_soc_component *scomp, + struct snd_sof_control *scontrol, + struct snd_kcontrol_new *kc, + struct snd_soc_tplg_ctl_hdr *hdr) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_enum_control *ec = + container_of(hdr, struct snd_soc_tplg_enum_control, hdr); + + /* validate topology data */ + if (le32_to_cpu(ec->num_channels) > SND_SOC_TPLG_MAX_CHAN) + return -EINVAL; + + /* init the enum get/put data */ + scontrol->size = sizeof(struct sof_ipc_ctrl_data) + + sizeof(struct sof_ipc_ctrl_value_chan) * + le32_to_cpu(ec->num_channels); + scontrol->control_data = kzalloc(scontrol->size, GFP_KERNEL); + if (!scontrol->control_data) + return -ENOMEM; + + scontrol->comp_id = sdev->next_comp_id; + scontrol->num_channels = le32_to_cpu(ec->num_channels); + + scontrol->cmd = SOF_CTRL_CMD_ENUM; + + dev_dbg(sdev->dev, "tplg: load kcontrol index %d chans %d comp_id %d\n", + scontrol->comp_id, scontrol->num_channels, scontrol->comp_id); + + return 0; +} + +static int sof_control_load_bytes(struct snd_soc_component *scomp, + struct snd_sof_control *scontrol, + struct snd_kcontrol_new *kc, + struct snd_soc_tplg_ctl_hdr *hdr) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct sof_ipc_ctrl_data *cdata; + struct snd_soc_tplg_bytes_control *control = + container_of(hdr, struct snd_soc_tplg_bytes_control, hdr); + struct soc_bytes_ext *sbe = (struct soc_bytes_ext *)kc->private_value; + int max_size = sbe->max; + + if (le32_to_cpu(control->priv.size) > max_size) { + dev_err(sdev->dev, "err: bytes data size %d exceeds max %d.\n", + control->priv.size, max_size); + return -EINVAL; + } + + /* init the get/put bytes data */ + scontrol->size = sizeof(struct sof_ipc_ctrl_data) + + le32_to_cpu(control->priv.size); + scontrol->control_data = kzalloc(max_size, GFP_KERNEL); + cdata = scontrol->control_data; + if (!scontrol->control_data) + return -ENOMEM; + + scontrol->comp_id = sdev->next_comp_id; + scontrol->cmd = SOF_CTRL_CMD_BINARY; + + dev_dbg(sdev->dev, "tplg: load kcontrol index %d chans %d\n", + scontrol->comp_id, scontrol->num_channels); + + if (le32_to_cpu(control->priv.size) > 0) { + memcpy(cdata->data, control->priv.data, + le32_to_cpu(control->priv.size)); + + if (cdata->data->magic != SOF_ABI_MAGIC) { + dev_err(sdev->dev, "error: Wrong ABI magic 0x%08x.\n", + cdata->data->magic); + return -EINVAL; + } + if (SOF_ABI_VERSION_INCOMPATIBLE(SOF_ABI_VERSION, + cdata->data->abi)) { + dev_err(sdev->dev, + "error: Incompatible ABI version 0x%08x.\n", + cdata->data->abi); + return -EINVAL; + } + if (cdata->data->size + sizeof(const struct sof_abi_hdr) != + le32_to_cpu(control->priv.size)) { + dev_err(sdev->dev, + "error: Conflict in bytes vs. priv size.\n"); + return -EINVAL; + } + } + return 0; +} + +/* + * Topology Token Parsing. + * New tokens should be added to headers and parsing tables below. + */ + +struct sof_topology_token { + u32 token; + u32 type; + int (*get_token)(void *elem, void *object, u32 offset, u32 size); + u32 offset; + u32 size; +}; + +static int get_token_u32(void *elem, void *object, u32 offset, u32 size) +{ + struct snd_soc_tplg_vendor_value_elem *velem = elem; + u32 *val = (u32 *)((u8 *)object + offset); + + *val = le32_to_cpu(velem->value); + return 0; +} + +static int get_token_u16(void *elem, void *object, u32 offset, u32 size) +{ + struct snd_soc_tplg_vendor_value_elem *velem = elem; + u16 *val = (u16 *)((u8 *)object + offset); + + *val = (u16)le32_to_cpu(velem->value); + return 0; +} + +static int get_token_comp_format(void *elem, void *object, u32 offset, u32 size) +{ + struct snd_soc_tplg_vendor_string_elem *velem = elem; + u32 *val = (u32 *)((u8 *)object + offset); + + *val = find_format(velem->string); + return 0; +} + +static int get_token_dai_type(void *elem, void *object, u32 offset, u32 size) +{ + struct snd_soc_tplg_vendor_string_elem *velem = elem; + u32 *val = (u32 *)((u8 *)object + offset); + + *val = find_dai(velem->string); + return 0; +} + +static int get_token_process_type(void *elem, void *object, u32 offset, + u32 size) +{ + struct snd_soc_tplg_vendor_string_elem *velem = elem; + u32 *val = (u32 *)((u8 *)object + offset); + + *val = find_process(velem->string); + return 0; +} + +/* Buffers */ +static const struct sof_topology_token buffer_tokens[] = { + {SOF_TKN_BUF_SIZE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_buffer, size), 0}, + {SOF_TKN_BUF_CAPS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_buffer, caps), 0}, +}; + +/* DAI */ +static const struct sof_topology_token dai_tokens[] = { + {SOF_TKN_DAI_TYPE, SND_SOC_TPLG_TUPLE_TYPE_STRING, get_token_dai_type, + offsetof(struct sof_ipc_comp_dai, type), 0}, + {SOF_TKN_DAI_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_comp_dai, dai_index), 0}, + {SOF_TKN_DAI_DIRECTION, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_comp_dai, direction), 0}, +}; + +/* BE DAI link */ +static const struct sof_topology_token dai_link_tokens[] = { + {SOF_TKN_DAI_TYPE, SND_SOC_TPLG_TUPLE_TYPE_STRING, get_token_dai_type, + offsetof(struct sof_ipc_dai_config, type), 0}, + {SOF_TKN_DAI_INDEX, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_config, dai_index), 0}, +}; + +/* scheduling */ +static const struct sof_topology_token sched_tokens[] = { + {SOF_TKN_SCHED_PERIOD, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_pipe_new, period), 0}, + {SOF_TKN_SCHED_PRIORITY, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_pipe_new, priority), 0}, + {SOF_TKN_SCHED_MIPS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_pipe_new, period_mips), 0}, + {SOF_TKN_SCHED_CORE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_pipe_new, core), 0}, + {SOF_TKN_SCHED_FRAMES, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_pipe_new, frames_per_sched), 0}, + {SOF_TKN_SCHED_TIME_DOMAIN, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_pipe_new, time_domain), 0}, +}; + +/* volume */ +static const struct sof_topology_token volume_tokens[] = { + {SOF_TKN_VOLUME_RAMP_STEP_TYPE, SND_SOC_TPLG_TUPLE_TYPE_WORD, + get_token_u32, offsetof(struct sof_ipc_comp_volume, ramp), 0}, + {SOF_TKN_VOLUME_RAMP_STEP_MS, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_comp_volume, initial_ramp), 0}, +}; + +/* SRC */ +static const struct sof_topology_token src_tokens[] = { + {SOF_TKN_SRC_RATE_IN, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_comp_src, source_rate), 0}, + {SOF_TKN_SRC_RATE_OUT, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_comp_src, sink_rate), 0}, +}; + +/* Tone */ +static const struct sof_topology_token tone_tokens[] = { +}; + +/* EFFECT */ +static const struct sof_topology_token process_tokens[] = { + {SOF_TKN_PROCESS_TYPE, SND_SOC_TPLG_TUPLE_TYPE_STRING, + get_token_process_type, + offsetof(struct sof_ipc_comp_process, type), 0}, +}; + +/* PCM */ +static const struct sof_topology_token pcm_tokens[] = { + {SOF_TKN_PCM_DMAC_CONFIG, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_comp_host, dmac_config), 0}, +}; + +/* Generic components */ +static const struct sof_topology_token comp_tokens[] = { + {SOF_TKN_COMP_PERIOD_SINK_COUNT, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_comp_config, periods_sink), 0}, + {SOF_TKN_COMP_PERIOD_SOURCE_COUNT, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_comp_config, periods_source), 0}, + {SOF_TKN_COMP_FORMAT, + SND_SOC_TPLG_TUPLE_TYPE_STRING, get_token_comp_format, + offsetof(struct sof_ipc_comp_config, frame_fmt), 0}, +}; + +/* SSP */ +static const struct sof_topology_token ssp_tokens[] = { + {SOF_TKN_INTEL_SSP_CLKS_CONTROL, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_ssp_params, clks_control), 0}, + {SOF_TKN_INTEL_SSP_MCLK_ID, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_ssp_params, mclk_id), 0}, + {SOF_TKN_INTEL_SSP_SAMPLE_BITS, SND_SOC_TPLG_TUPLE_TYPE_WORD, + get_token_u32, + offsetof(struct sof_ipc_dai_ssp_params, sample_valid_bits), 0}, + {SOF_TKN_INTEL_SSP_FRAME_PULSE_WIDTH, SND_SOC_TPLG_TUPLE_TYPE_SHORT, + get_token_u16, + offsetof(struct sof_ipc_dai_ssp_params, frame_pulse_width), 0}, + {SOF_TKN_INTEL_SSP_QUIRKS, SND_SOC_TPLG_TUPLE_TYPE_WORD, + get_token_u32, + offsetof(struct sof_ipc_dai_ssp_params, quirks), 0}, + {SOF_TKN_INTEL_SSP_TDM_PADDING_PER_SLOT, SND_SOC_TPLG_TUPLE_TYPE_BOOL, + get_token_u16, + offsetof(struct sof_ipc_dai_ssp_params, + tdm_per_slot_padding_flag), 0}, + +}; + +/* DMIC */ +static const struct sof_topology_token dmic_tokens[] = { + {SOF_TKN_INTEL_DMIC_DRIVER_VERSION, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_dmic_params, driver_ipc_version), + 0}, + {SOF_TKN_INTEL_DMIC_CLK_MIN, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_dmic_params, pdmclk_min), 0}, + {SOF_TKN_INTEL_DMIC_CLK_MAX, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_dmic_params, pdmclk_max), 0}, + {SOF_TKN_INTEL_DMIC_SAMPLE_RATE, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_dmic_params, fifo_fs), 0}, + {SOF_TKN_INTEL_DMIC_DUTY_MIN, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_dmic_params, duty_min), 0}, + {SOF_TKN_INTEL_DMIC_DUTY_MAX, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_dmic_params, duty_max), 0}, + {SOF_TKN_INTEL_DMIC_NUM_PDM_ACTIVE, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_dmic_params, + num_pdm_active), 0}, + {SOF_TKN_INTEL_DMIC_FIFO_WORD_LENGTH, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_dmic_params, fifo_bits), 0}, +}; + +/* + * DMIC PDM Tokens + * SOF_TKN_INTEL_DMIC_PDM_CTRL_ID should be the first token + * as it increments the index while parsing the array of pdm tokens + * and determines the correct offset + */ +static const struct sof_topology_token dmic_pdm_tokens[] = { + {SOF_TKN_INTEL_DMIC_PDM_CTRL_ID, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_dmic_pdm_ctrl, id), + 0}, + {SOF_TKN_INTEL_DMIC_PDM_MIC_A_Enable, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_dmic_pdm_ctrl, enable_mic_a), + 0}, + {SOF_TKN_INTEL_DMIC_PDM_MIC_B_Enable, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_dmic_pdm_ctrl, enable_mic_b), + 0}, + {SOF_TKN_INTEL_DMIC_PDM_POLARITY_A, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_dmic_pdm_ctrl, polarity_mic_a), + 0}, + {SOF_TKN_INTEL_DMIC_PDM_POLARITY_B, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_dmic_pdm_ctrl, polarity_mic_b), + 0}, + {SOF_TKN_INTEL_DMIC_PDM_CLK_EDGE, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_dmic_pdm_ctrl, clk_edge), + 0}, + {SOF_TKN_INTEL_DMIC_PDM_SKEW, + SND_SOC_TPLG_TUPLE_TYPE_SHORT, get_token_u16, + offsetof(struct sof_ipc_dai_dmic_pdm_ctrl, skew), + 0}, +}; + +/* HDA */ +static const struct sof_topology_token hda_tokens[] = { +}; + +static void sof_parse_uuid_tokens(struct snd_soc_component *scomp, + void *object, + const struct sof_topology_token *tokens, + int count, + struct snd_soc_tplg_vendor_array *array) +{ + struct snd_soc_tplg_vendor_uuid_elem *elem; + int i, j; + + /* parse element by element */ + for (i = 0; i < le32_to_cpu(array->num_elems); i++) { + elem = &array->uuid[i]; + + /* search for token */ + for (j = 0; j < count; j++) { + /* match token type */ + if (tokens[j].type != SND_SOC_TPLG_TUPLE_TYPE_UUID) + continue; + + /* match token id */ + if (tokens[j].token != le32_to_cpu(elem->token)) + continue; + + /* matched - now load token */ + tokens[j].get_token(elem, object, tokens[j].offset, + tokens[j].size); + } + } +} + +static void sof_parse_string_tokens(struct snd_soc_component *scomp, + void *object, + const struct sof_topology_token *tokens, + int count, + struct snd_soc_tplg_vendor_array *array) +{ + struct snd_soc_tplg_vendor_string_elem *elem; + int i, j; + + /* parse element by element */ + for (i = 0; i < le32_to_cpu(array->num_elems); i++) { + elem = &array->string[i]; + + /* search for token */ + for (j = 0; j < count; j++) { + /* match token type */ + if (tokens[j].type != SND_SOC_TPLG_TUPLE_TYPE_STRING) + continue; + + /* match token id */ + if (tokens[j].token != le32_to_cpu(elem->token)) + continue; + + /* matched - now load token */ + tokens[j].get_token(elem, object, tokens[j].offset, + tokens[j].size); + } + } +} + +static void sof_parse_word_tokens(struct snd_soc_component *scomp, + void *object, + const struct sof_topology_token *tokens, + int count, + struct snd_soc_tplg_vendor_array *array) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_vendor_value_elem *elem; + size_t size = sizeof(struct sof_ipc_dai_dmic_pdm_ctrl); + int i, j; + u32 offset; + u32 *index = NULL; + + /* parse element by element */ + for (i = 0; i < le32_to_cpu(array->num_elems); i++) { + elem = &array->value[i]; + + /* search for token */ + for (j = 0; j < count; j++) { + /* match token type */ + if (!(tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_WORD || + tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT)) + continue; + + /* match token id */ + if (tokens[j].token != le32_to_cpu(elem->token)) + continue; + + /* pdm config array index */ + if (sdev->private) + index = sdev->private; + + /* matched - determine offset */ + switch (tokens[j].token) { + case SOF_TKN_INTEL_DMIC_PDM_CTRL_ID: + + /* inc number of pdm array index */ + if (index) + (*index)++; + /* fallthrough */ + case SOF_TKN_INTEL_DMIC_PDM_MIC_A_Enable: + case SOF_TKN_INTEL_DMIC_PDM_MIC_B_Enable: + case SOF_TKN_INTEL_DMIC_PDM_POLARITY_A: + case SOF_TKN_INTEL_DMIC_PDM_POLARITY_B: + case SOF_TKN_INTEL_DMIC_PDM_CLK_EDGE: + case SOF_TKN_INTEL_DMIC_PDM_SKEW: + + /* check if array index is valid */ + if (!index || *index == 0) { + dev_err(sdev->dev, + "error: invalid array offset\n"); + continue; + } else { + /* offset within the pdm config array */ + offset = size * (*index - 1); + } + break; + default: + offset = 0; + break; + } + + /* load token */ + tokens[j].get_token(elem, object, + offset + tokens[j].offset, + tokens[j].size); + } + } +} + +static int sof_parse_tokens(struct snd_soc_component *scomp, + void *object, + const struct sof_topology_token *tokens, + int count, + struct snd_soc_tplg_vendor_array *array, + int priv_size) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + int asize; + + while (priv_size > 0) { + asize = le32_to_cpu(array->size); + + /* validate asize */ + if (asize < 0) { /* FIXME: A zero-size array makes no sense */ + dev_err(sdev->dev, "error: invalid array size 0x%x\n", + asize); + return -EINVAL; + } + + /* make sure there is enough data before parsing */ + priv_size -= asize; + if (priv_size < 0) { + dev_err(sdev->dev, "error: invalid array size 0x%x\n", + asize); + return -EINVAL; + } + + /* call correct parser depending on type */ + switch (le32_to_cpu(array->type)) { + case SND_SOC_TPLG_TUPLE_TYPE_UUID: + sof_parse_uuid_tokens(scomp, object, tokens, count, + array); + break; + case SND_SOC_TPLG_TUPLE_TYPE_STRING: + sof_parse_string_tokens(scomp, object, tokens, count, + array); + break; + case SND_SOC_TPLG_TUPLE_TYPE_BOOL: + case SND_SOC_TPLG_TUPLE_TYPE_BYTE: + case SND_SOC_TPLG_TUPLE_TYPE_WORD: + case SND_SOC_TPLG_TUPLE_TYPE_SHORT: + sof_parse_word_tokens(scomp, object, tokens, count, + array); + break; + default: + dev_err(sdev->dev, "error: unknown token type %d\n", + array->type); + return -EINVAL; + } + + /* next array */ + array = (struct snd_soc_tplg_vendor_array *)((u8 *)array + + asize); + } + return 0; +} + +static void sof_dbg_comp_config(struct snd_soc_component *scomp, + struct sof_ipc_comp_config *config) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + + dev_dbg(sdev->dev, " config: periods snk %d src %d fmt %d\n", + config->periods_sink, config->periods_source, + config->frame_fmt); +} + +/* external kcontrol init - used for any driver specific init */ +static int sof_control_load(struct snd_soc_component *scomp, int index, + struct snd_kcontrol_new *kc, + struct snd_soc_tplg_ctl_hdr *hdr) +{ + struct soc_mixer_control *sm; + struct soc_bytes_ext *sbe; + struct soc_enum *se; + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_dobj *dobj; + struct snd_sof_control *scontrol; + int ret = -EINVAL; + + dev_dbg(sdev->dev, "tplg: load control type %d name : %s\n", + hdr->type, hdr->name); + + scontrol = kzalloc(sizeof(*scontrol), GFP_KERNEL); + if (!scontrol) + return -ENOMEM; + + scontrol->sdev = sdev; + + switch (le32_to_cpu(hdr->ops.info)) { + case SND_SOC_TPLG_CTL_VOLSW: + case SND_SOC_TPLG_CTL_VOLSW_SX: + case SND_SOC_TPLG_CTL_VOLSW_XR_SX: + sm = (struct soc_mixer_control *)kc->private_value; + dobj = &sm->dobj; + ret = sof_control_load_volume(scomp, scontrol, kc, hdr); + break; + case SND_SOC_TPLG_CTL_BYTES: + sbe = (struct soc_bytes_ext *)kc->private_value; + dobj = &sbe->dobj; + ret = sof_control_load_bytes(scomp, scontrol, kc, hdr); + break; + case SND_SOC_TPLG_CTL_ENUM: + case SND_SOC_TPLG_CTL_ENUM_VALUE: + se = (struct soc_enum *)kc->private_value; + dobj = &se->dobj; + ret = sof_control_load_enum(scomp, scontrol, kc, hdr); + break; + case SND_SOC_TPLG_CTL_RANGE: + case SND_SOC_TPLG_CTL_STROBE: + case SND_SOC_TPLG_DAPM_CTL_VOLSW: + case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: + case SND_SOC_TPLG_DAPM_CTL_PIN: + default: + dev_warn(sdev->dev, "control type not supported %d:%d:%d\n", + hdr->ops.get, hdr->ops.put, hdr->ops.info); + kfree(scontrol); + return 0; + } + + dobj->private = scontrol; + list_add(&scontrol->list, &sdev->kcontrol_list); + return ret; +} + +static int sof_control_unload(struct snd_soc_component *scomp, + struct snd_soc_dobj *dobj) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct sof_ipc_free fcomp; + struct snd_sof_control *scontrol = dobj->private; + + dev_dbg(sdev->dev, "tplg: unload control name : %s\n", scomp->name); + + fcomp.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_FREE; + fcomp.hdr.size = sizeof(fcomp); + fcomp.id = scontrol->comp_id; + + kfree(scontrol->control_data); + list_del(&scontrol->list); + kfree(scontrol); + /* send IPC to the DSP */ + return sof_ipc_tx_message(sdev->ipc, + fcomp.hdr.cmd, &fcomp, sizeof(fcomp), + NULL, 0); +} + +/* + * DAI Topology + */ + +static int sof_connect_dai_widget(struct snd_soc_component *scomp, + struct snd_soc_dapm_widget *w, + struct snd_soc_tplg_dapm_widget *tw, + struct snd_sof_dai *dai) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_card *card = scomp->card; + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + dev_vdbg(sdev->dev, "tplg: check widget: %s stream: %s dai stream: %s\n", + w->name, w->sname, rtd->dai_link->stream_name); + + if (!w->sname || !rtd->dai_link->stream_name) + continue; + + /* does stream match DAI link ? */ + if (strcmp(w->sname, rtd->dai_link->stream_name)) + continue; + + switch (w->id) { + case snd_soc_dapm_dai_out: + rtd->cpu_dai->capture_widget = w; + if (dai) + dai->name = rtd->dai_link->name; + dev_dbg(sdev->dev, "tplg: connected widget %s -> DAI link %s\n", + w->name, rtd->dai_link->name); + break; + case snd_soc_dapm_dai_in: + rtd->cpu_dai->playback_widget = w; + if (dai) + dai->name = rtd->dai_link->name; + dev_dbg(sdev->dev, "tplg: connected widget %s -> DAI link %s\n", + w->name, rtd->dai_link->name); + break; + default: + break; + } + } + + /* check we have a connection */ + if (!dai->name) { + dev_err(sdev->dev, "error: can't connect DAI %s stream %s\n", + w->name, w->sname); + return -EINVAL; + } + + return 0; +} + +static int sof_widget_load_dai(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r, + struct snd_sof_dai *dai) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_comp_dai comp_dai; + int ret; + + /* configure dai IPC message */ + memset(&comp_dai, 0, sizeof(comp_dai)); + comp_dai.comp.hdr.size = sizeof(comp_dai); + comp_dai.comp.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; + comp_dai.comp.id = swidget->comp_id; + comp_dai.comp.type = SOF_COMP_DAI; + comp_dai.comp.pipeline_id = index; + comp_dai.config.hdr.size = sizeof(comp_dai.config); + + ret = sof_parse_tokens(scomp, &comp_dai, dai_tokens, + ARRAY_SIZE(dai_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse dai tokens failed %d\n", + le32_to_cpu(private->size)); + return ret; + } + + ret = sof_parse_tokens(scomp, &comp_dai.config, comp_tokens, + ARRAY_SIZE(comp_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse dai.cfg tokens failed %d\n", + private->size); + return ret; + } + + dev_dbg(sdev->dev, "dai %s: type %d index %d\n", + swidget->widget->name, comp_dai.type, comp_dai.dai_index); + sof_dbg_comp_config(scomp, &comp_dai.config); + + ret = sof_ipc_tx_message(sdev->ipc, comp_dai.comp.hdr.cmd, + &comp_dai, sizeof(comp_dai), r, sizeof(*r)); + + if (ret == 0 && dai) { + dai->sdev = sdev; + memcpy(&dai->comp_dai, &comp_dai, sizeof(comp_dai)); + } + + return ret; +} + +/* + * Buffer topology + */ + +static int sof_widget_load_buffer(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_buffer *buffer; + int ret; + + buffer = kzalloc(sizeof(*buffer), GFP_KERNEL); + if (!buffer) + return -ENOMEM; + + /* configure dai IPC message */ + buffer->comp.hdr.size = sizeof(*buffer); + buffer->comp.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_BUFFER_NEW; + buffer->comp.id = swidget->comp_id; + buffer->comp.type = SOF_COMP_BUFFER; + buffer->comp.pipeline_id = index; + + ret = sof_parse_tokens(scomp, buffer, buffer_tokens, + ARRAY_SIZE(buffer_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse buffer tokens failed %d\n", + private->size); + kfree(buffer); + return ret; + } + + dev_dbg(sdev->dev, "buffer %s: size %d caps 0x%x\n", + swidget->widget->name, buffer->size, buffer->caps); + + swidget->private = buffer; + + ret = sof_ipc_tx_message(sdev->ipc, buffer->comp.hdr.cmd, buffer, + sizeof(*buffer), r, sizeof(*r)); + if (ret < 0) { + dev_err(sdev->dev, "error: buffer %s load failed\n", + swidget->widget->name); + kfree(buffer); + } + + return ret; +} + +/* bind PCM ID to host component ID */ +static int spcm_bind(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm, + int dir) +{ + struct snd_sof_widget *host_widget; + + host_widget = snd_sof_find_swidget_sname(sdev, + spcm->pcm.caps[dir].name, + dir); + if (!host_widget) { + dev_err(sdev->dev, "can't find host comp to bind pcm\n"); + return -EINVAL; + } + + spcm->stream[dir].comp_id = host_widget->comp_id; + + return 0; +} + +/* + * PCM Topology + */ + +static int sof_widget_load_pcm(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + enum sof_ipc_stream_direction dir, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_comp_host *host; + int ret; + + host = kzalloc(sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* configure host comp IPC message */ + host->comp.hdr.size = sizeof(*host); + host->comp.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; + host->comp.id = swidget->comp_id; + host->comp.type = SOF_COMP_HOST; + host->comp.pipeline_id = index; + host->direction = dir; + host->config.hdr.size = sizeof(host->config); + + ret = sof_parse_tokens(scomp, host, pcm_tokens, + ARRAY_SIZE(pcm_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse host tokens failed %d\n", + private->size); + goto err; + } + + ret = sof_parse_tokens(scomp, &host->config, comp_tokens, + ARRAY_SIZE(comp_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse host.cfg tokens failed %d\n", + le32_to_cpu(private->size)); + goto err; + } + + dev_dbg(sdev->dev, "loaded host %s\n", swidget->widget->name); + sof_dbg_comp_config(scomp, &host->config); + + swidget->private = host; + + ret = sof_ipc_tx_message(sdev->ipc, host->comp.hdr.cmd, host, + sizeof(*host), r, sizeof(*r)); + if (ret >= 0) + return ret; +err: + kfree(host); + return ret; +} + +/* + * Pipeline Topology + */ +int sof_load_pipeline_ipc(struct snd_sof_dev *sdev, + struct sof_ipc_pipe_new *pipeline, + struct sof_ipc_comp_reply *r) +{ + struct sof_ipc_pm_core_config pm_core_config; + int ret; + + ret = sof_ipc_tx_message(sdev->ipc, pipeline->hdr.cmd, pipeline, + sizeof(*pipeline), r, sizeof(*r)); + if (ret < 0) { + dev_err(sdev->dev, "error: load pipeline ipc failure\n"); + return ret; + } + + /* power up the core that this pipeline is scheduled on */ + ret = snd_sof_dsp_core_power_up(sdev, 1 << pipeline->core); + if (ret < 0) { + dev_err(sdev->dev, "error: powering up pipeline schedule core %d\n", + pipeline->core); + return ret; + } + + /* update enabled cores mask */ + sdev->enabled_cores_mask |= 1 << pipeline->core; + + /* + * Now notify DSP that the core that this pipeline is scheduled on + * has been powered up + */ + memset(&pm_core_config, 0, sizeof(pm_core_config)); + pm_core_config.enable_mask = sdev->enabled_cores_mask; + + /* configure CORE_ENABLE ipc message */ + pm_core_config.hdr.size = sizeof(pm_core_config); + pm_core_config.hdr.cmd = SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CORE_ENABLE; + + /* send ipc */ + ret = sof_ipc_tx_message(sdev->ipc, pm_core_config.hdr.cmd, + &pm_core_config, sizeof(pm_core_config), + &pm_core_config, sizeof(pm_core_config)); + if (ret < 0) + dev_err(sdev->dev, "error: core enable ipc failure\n"); + + return ret; +} + +static int sof_widget_load_pipeline(struct snd_soc_component *scomp, + int index, struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_pipe_new *pipeline; + struct snd_sof_widget *comp_swidget; + int ret; + + pipeline = kzalloc(sizeof(*pipeline), GFP_KERNEL); + if (!pipeline) + return -ENOMEM; + + /* configure dai IPC message */ + pipeline->hdr.size = sizeof(*pipeline); + pipeline->hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_PIPE_NEW; + pipeline->pipeline_id = index; + pipeline->comp_id = swidget->comp_id; + + /* component at start of pipeline is our stream id */ + comp_swidget = snd_sof_find_swidget(sdev, tw->sname); + if (!comp_swidget) { + dev_err(sdev->dev, "error: widget %s refers to non existent widget %s\n", + tw->name, tw->sname); + ret = -EINVAL; + goto err; + } + + pipeline->sched_id = comp_swidget->comp_id; + + dev_dbg(sdev->dev, "tplg: pipeline id %d comp %d scheduling comp id %d\n", + pipeline->pipeline_id, pipeline->comp_id, pipeline->sched_id); + + ret = sof_parse_tokens(scomp, pipeline, sched_tokens, + ARRAY_SIZE(sched_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse pipeline tokens failed %d\n", + private->size); + goto err; + } + + dev_dbg(sdev->dev, "pipeline %s: period %d pri %d mips %d core %d frames %d\n", + swidget->widget->name, pipeline->period, pipeline->priority, + pipeline->period_mips, pipeline->core, pipeline->frames_per_sched); + + swidget->private = pipeline; + + /* send ipc's to create pipeline comp and power up schedule core */ + ret = sof_load_pipeline_ipc(sdev, pipeline, r); + if (ret >= 0) + return ret; +err: + kfree(pipeline); + return ret; +} + +/* + * Mixer topology + */ + +static int sof_widget_load_mixer(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_comp_mixer *mixer; + int ret; + + mixer = kzalloc(sizeof(*mixer), GFP_KERNEL); + if (!mixer) + return -ENOMEM; + + /* configure mixer IPC message */ + mixer->comp.hdr.size = sizeof(*mixer); + mixer->comp.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; + mixer->comp.id = swidget->comp_id; + mixer->comp.type = SOF_COMP_MIXER; + mixer->comp.pipeline_id = index; + mixer->config.hdr.size = sizeof(mixer->config); + + ret = sof_parse_tokens(scomp, &mixer->config, comp_tokens, + ARRAY_SIZE(comp_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse mixer.cfg tokens failed %d\n", + private->size); + kfree(mixer); + return ret; + } + + sof_dbg_comp_config(scomp, &mixer->config); + + swidget->private = mixer; + + ret = sof_ipc_tx_message(sdev->ipc, mixer->comp.hdr.cmd, mixer, + sizeof(*mixer), r, sizeof(*r)); + if (ret < 0) + kfree(mixer); + + return ret; +} + +/* + * Mux topology + */ +static int sof_widget_load_mux(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_comp_mux *mux; + int ret; + + mux = kzalloc(sizeof(*mux), GFP_KERNEL); + if (!mux) + return -ENOMEM; + + /* configure mux IPC message */ + mux->comp.hdr.size = sizeof(*mux); + mux->comp.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; + mux->comp.id = swidget->comp_id; + mux->comp.type = SOF_COMP_MUX; + mux->comp.pipeline_id = index; + mux->config.hdr.size = sizeof(mux->config); + + ret = sof_parse_tokens(scomp, &mux->config, comp_tokens, + ARRAY_SIZE(comp_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse mux.cfg tokens failed %d\n", + private->size); + kfree(mux); + return ret; + } + + sof_dbg_comp_config(scomp, &mux->config); + + swidget->private = mux; + + ret = sof_ipc_tx_message(sdev->ipc, mux->comp.hdr.cmd, mux, + sizeof(*mux), r, sizeof(*r)); + if (ret < 0) + kfree(mux); + + return ret; +} + +/* + * PGA Topology + */ + +static int sof_widget_load_pga(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_comp_volume *volume; + int ret; + + volume = kzalloc(sizeof(*volume), GFP_KERNEL); + if (!volume) + return -ENOMEM; + + if (le32_to_cpu(tw->num_kcontrols) != 1) { + dev_err(sdev->dev, "error: invalid kcontrol count %d for volume\n", + tw->num_kcontrols); + ret = -EINVAL; + goto err; + } + + /* configure volume IPC message */ + volume->comp.hdr.size = sizeof(*volume); + volume->comp.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; + volume->comp.id = swidget->comp_id; + volume->comp.type = SOF_COMP_VOLUME; + volume->comp.pipeline_id = index; + volume->config.hdr.size = sizeof(volume->config); + + ret = sof_parse_tokens(scomp, volume, volume_tokens, + ARRAY_SIZE(volume_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse volume tokens failed %d\n", + private->size); + goto err; + } + ret = sof_parse_tokens(scomp, &volume->config, comp_tokens, + ARRAY_SIZE(comp_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse volume.cfg tokens failed %d\n", + le32_to_cpu(private->size)); + goto err; + } + + sof_dbg_comp_config(scomp, &volume->config); + + swidget->private = volume; + + ret = sof_ipc_tx_message(sdev->ipc, volume->comp.hdr.cmd, volume, + sizeof(*volume), r, sizeof(*r)); + if (ret >= 0) + return ret; +err: + kfree(volume); + return ret; +} + +/* + * SRC Topology + */ + +static int sof_widget_load_src(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_comp_src *src; + int ret; + + src = kzalloc(sizeof(*src), GFP_KERNEL); + if (!src) + return -ENOMEM; + + /* configure src IPC message */ + src->comp.hdr.size = sizeof(*src); + src->comp.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; + src->comp.id = swidget->comp_id; + src->comp.type = SOF_COMP_SRC; + src->comp.pipeline_id = index; + src->config.hdr.size = sizeof(src->config); + + ret = sof_parse_tokens(scomp, src, src_tokens, + ARRAY_SIZE(src_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse src tokens failed %d\n", + private->size); + goto err; + } + + ret = sof_parse_tokens(scomp, &src->config, comp_tokens, + ARRAY_SIZE(comp_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse src.cfg tokens failed %d\n", + le32_to_cpu(private->size)); + goto err; + } + + dev_dbg(sdev->dev, "src %s: source rate %d sink rate %d\n", + swidget->widget->name, src->source_rate, src->sink_rate); + sof_dbg_comp_config(scomp, &src->config); + + swidget->private = src; + + ret = sof_ipc_tx_message(sdev->ipc, src->comp.hdr.cmd, src, + sizeof(*src), r, sizeof(*r)); + if (ret >= 0) + return ret; +err: + kfree(src); + return ret; +} + +/* + * Signal Generator Topology + */ + +static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_comp_tone *tone; + int ret; + + tone = kzalloc(sizeof(*tone), GFP_KERNEL); + if (!tone) + return -ENOMEM; + + /* configure siggen IPC message */ + tone->comp.hdr.size = sizeof(*tone); + tone->comp.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; + tone->comp.id = swidget->comp_id; + tone->comp.type = SOF_COMP_TONE; + tone->comp.pipeline_id = index; + tone->config.hdr.size = sizeof(tone->config); + + ret = sof_parse_tokens(scomp, tone, tone_tokens, + ARRAY_SIZE(tone_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse tone tokens failed %d\n", + le32_to_cpu(private->size)); + goto err; + } + + ret = sof_parse_tokens(scomp, &tone->config, comp_tokens, + ARRAY_SIZE(comp_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse tone.cfg tokens failed %d\n", + le32_to_cpu(private->size)); + goto err; + } + + dev_dbg(sdev->dev, "tone %s: frequency %d amplitude %d\n", + swidget->widget->name, tone->frequency, tone->amplitude); + sof_dbg_comp_config(scomp, &tone->config); + + swidget->private = tone; + + ret = sof_ipc_tx_message(sdev->ipc, tone->comp.hdr.cmd, tone, + sizeof(*tone), r, sizeof(*r)); + if (ret >= 0) + return ret; +err: + kfree(tone); + return ret; +} + +static int sof_process_load(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r, + int type) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct snd_soc_dapm_widget *widget = swidget->widget; + const struct snd_kcontrol_new *kc; + struct soc_bytes_ext *sbe; + struct soc_mixer_control *sm; + struct soc_enum *se; + struct snd_sof_control *scontrol = NULL; + struct sof_abi_hdr *pdata = NULL; + struct sof_ipc_comp_process *process; + size_t ipc_size, ipc_data_size = 0; + int ret, i, offset = 0; + + if (type == SOF_COMP_NONE) { + dev_err(sdev->dev, "error: invalid process comp type %d\n", + type); + return -EINVAL; + } + + /* + * get possible component controls - get size of all pdata, + * then memcpy with headers + */ + for (i = 0; i < widget->num_kcontrols; i++) { + + kc = &widget->kcontrol_news[i]; + + switch (widget->dobj.widget.kcontrol_type) { + case SND_SOC_TPLG_TYPE_MIXER: + sm = (struct soc_mixer_control *)kc->private_value; + scontrol = sm->dobj.private; + break; + case SND_SOC_TPLG_TYPE_BYTES: + sbe = (struct soc_bytes_ext *)kc->private_value; + scontrol = sbe->dobj.private; + break; + case SND_SOC_TPLG_TYPE_ENUM: + se = (struct soc_enum *)kc->private_value; + scontrol = se->dobj.private; + break; + default: + dev_err(sdev->dev, "error: unknown kcontrol type %d in widget %s\n", + widget->dobj.widget.kcontrol_type, + widget->name); + return -EINVAL; + } + + if (!scontrol) { + dev_err(sdev->dev, "error: no scontrol for widget %s\n", + widget->name); + return -EINVAL; + } + + /* don't include if no private data */ + pdata = scontrol->control_data->data; + if (!pdata) + continue; + + /* make sure data is valid - data can be updated at runtime */ + if (pdata->magic != SOF_ABI_MAGIC) + continue; + + ipc_data_size += pdata->size; + } + + ipc_size = sizeof(struct sof_ipc_comp_process) + + le32_to_cpu(private->size) + + ipc_data_size; + + process = kzalloc(ipc_size, GFP_KERNEL); + if (!process) + return -ENOMEM; + + /* configure iir IPC message */ + process->comp.hdr.size = ipc_size; + process->comp.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; + process->comp.id = swidget->comp_id; + process->comp.type = type; + process->comp.pipeline_id = index; + process->config.hdr.size = sizeof(process->config); + + ret = sof_parse_tokens(scomp, &process->config, comp_tokens, + ARRAY_SIZE(comp_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse process.cfg tokens failed %d\n", + le32_to_cpu(private->size)); + goto err; + } + + sof_dbg_comp_config(scomp, &process->config); + + /* + * found private data in control, so copy it. + * get possible component controls - get size of all pdata, + * then memcpy with headers + */ + for (i = 0; i < widget->num_kcontrols; i++) { + kc = &widget->kcontrol_news[i]; + + switch (widget->dobj.widget.kcontrol_type) { + case SND_SOC_TPLG_TYPE_MIXER: + sm = (struct soc_mixer_control *)kc->private_value; + scontrol = sm->dobj.private; + break; + case SND_SOC_TPLG_TYPE_BYTES: + sbe = (struct soc_bytes_ext *)kc->private_value; + scontrol = sbe->dobj.private; + break; + case SND_SOC_TPLG_TYPE_ENUM: + se = (struct soc_enum *)kc->private_value; + scontrol = se->dobj.private; + break; + default: + dev_err(sdev->dev, "error: unknown kcontrol type %d in widget %s\n", + widget->dobj.widget.kcontrol_type, + widget->name); + return -EINVAL; + } + + /* don't include if no private data */ + pdata = scontrol->control_data->data; + if (!pdata) + continue; + + /* make sure data is valid - data can be updated at runtime */ + if (pdata->magic != SOF_ABI_MAGIC) + continue; + + memcpy(&process->data + offset, pdata->data, pdata->size); + offset += pdata->size; + } + + process->size = ipc_data_size; + swidget->private = process; + + ret = sof_ipc_tx_message(sdev->ipc, process->comp.hdr.cmd, process, + ipc_size, r, sizeof(*r)); + if (ret >= 0) + return ret; +err: + kfree(process); + return ret; +} + +/* + * Processing Component Topology - can be "effect", "codec", or general + * "processing". + */ + +static int sof_widget_load_process(struct snd_soc_component *scomp, int index, + struct snd_sof_widget *swidget, + struct snd_soc_tplg_dapm_widget *tw, + struct sof_ipc_comp_reply *r) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &tw->priv; + struct sof_ipc_comp_process config; + int ret; + + /* check we have some tokens - we need at least process type */ + if (le32_to_cpu(private->size) == 0) { + dev_err(sdev->dev, "error: process tokens not found\n"); + return -EINVAL; + } + + memset(&config, 0, sizeof(config)); + + /* get the process token */ + ret = sof_parse_tokens(scomp, &config, process_tokens, + ARRAY_SIZE(process_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse process tokens failed %d\n", + le32_to_cpu(private->size)); + return ret; + } + + /* now load process specific data and send IPC */ + ret = sof_process_load(scomp, index, swidget, tw, r, + find_process_comp_type(config.type)); + if (ret < 0) { + dev_err(sdev->dev, "error: process loading failed\n"); + return ret; + } + + return 0; +} + +static int sof_widget_bind_event(struct snd_sof_dev *sdev, + struct snd_sof_widget *swidget, + u16 event_type) +{ + struct sof_ipc_comp *ipc_comp; + + /* validate widget event type */ + switch (event_type) { + case SOF_KEYWORD_DETECT_DAPM_EVENT: + /* only KEYWORD_DETECT comps should handle this */ + if (swidget->id != snd_soc_dapm_effect) + break; + + ipc_comp = swidget->private; + if (ipc_comp && ipc_comp->type != SOF_COMP_KEYWORD_DETECT) + break; + + /* bind event to keyword detect comp */ + return snd_soc_tplg_widget_bind_event(swidget->widget, + sof_kwd_events, + ARRAY_SIZE(sof_kwd_events), + event_type); + default: + break; + } + + dev_err(sdev->dev, + "error: invalid event type %d for widget %s\n", + event_type, swidget->widget->name); + return -EINVAL; +} + +/* external widget init - used for any driver specific init */ +static int sof_widget_ready(struct snd_soc_component *scomp, int index, + struct snd_soc_dapm_widget *w, + struct snd_soc_tplg_dapm_widget *tw) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_sof_widget *swidget; + struct snd_sof_dai *dai; + struct sof_ipc_comp_reply reply; + struct snd_sof_control *scontrol; + int ret = 0; + + swidget = kzalloc(sizeof(*swidget), GFP_KERNEL); + if (!swidget) + return -ENOMEM; + + swidget->sdev = sdev; + swidget->widget = w; + swidget->comp_id = sdev->next_comp_id++; + swidget->complete = 0; + swidget->id = w->id; + swidget->pipeline_id = index; + swidget->private = NULL; + memset(&reply, 0, sizeof(reply)); + + dev_dbg(sdev->dev, "tplg: ready widget id %d pipe %d type %d name : %s stream %s\n", + swidget->comp_id, index, swidget->id, tw->name, + strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0 + ? tw->sname : "none"); + + /* handle any special case widgets */ + switch (w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + dai = kzalloc(sizeof(*dai), GFP_KERNEL); + if (!dai) { + kfree(swidget); + return -ENOMEM; + } + + ret = sof_widget_load_dai(scomp, index, swidget, tw, &reply, + dai); + if (ret == 0) { + sof_connect_dai_widget(scomp, w, tw, dai); + list_add(&dai->list, &sdev->dai_list); + swidget->private = dai; + } else { + kfree(dai); + } + break; + case snd_soc_dapm_mixer: + ret = sof_widget_load_mixer(scomp, index, swidget, tw, &reply); + break; + case snd_soc_dapm_pga: + ret = sof_widget_load_pga(scomp, index, swidget, tw, &reply); + /* Find scontrol for this pga and set readback offset*/ + list_for_each_entry(scontrol, &sdev->kcontrol_list, list) { + if (scontrol->comp_id == swidget->comp_id) { + scontrol->readback_offset = reply.offset; + break; + } + } + break; + case snd_soc_dapm_buffer: + ret = sof_widget_load_buffer(scomp, index, swidget, tw, &reply); + break; + case snd_soc_dapm_scheduler: + ret = sof_widget_load_pipeline(scomp, index, swidget, tw, + &reply); + break; + case snd_soc_dapm_aif_out: + ret = sof_widget_load_pcm(scomp, index, swidget, + SOF_IPC_STREAM_CAPTURE, tw, &reply); + break; + case snd_soc_dapm_aif_in: + ret = sof_widget_load_pcm(scomp, index, swidget, + SOF_IPC_STREAM_PLAYBACK, tw, &reply); + break; + case snd_soc_dapm_src: + ret = sof_widget_load_src(scomp, index, swidget, tw, &reply); + break; + case snd_soc_dapm_siggen: + ret = sof_widget_load_siggen(scomp, index, swidget, tw, &reply); + break; + case snd_soc_dapm_effect: + ret = sof_widget_load_process(scomp, index, swidget, tw, + &reply); + break; + case snd_soc_dapm_mux: + case snd_soc_dapm_demux: + ret = sof_widget_load_mux(scomp, index, swidget, tw, &reply); + break; + case snd_soc_dapm_switch: + case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: + default: + dev_warn(sdev->dev, "warning: widget type %d name %s not handled\n", + swidget->id, tw->name); + break; + } + + /* check IPC reply */ + if (ret < 0 || reply.rhdr.error < 0) { + dev_err(sdev->dev, + "error: DSP failed to add widget id %d type %d name : %s stream %s reply %d\n", + tw->shift, swidget->id, tw->name, + strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0 + ? tw->sname : "none", reply.rhdr.error); + kfree(swidget); + return ret; + } + + /* bind widget to external event */ + if (tw->event_type) { + ret = sof_widget_bind_event(sdev, swidget, + le16_to_cpu(tw->event_type)); + if (ret) { + dev_err(sdev->dev, "error: widget event binding failed\n"); + kfree(swidget->private); + kfree(swidget); + return ret; + } + } + + w->dobj.private = swidget; + list_add(&swidget->list, &sdev->widget_list); + return ret; +} + +static int sof_route_unload(struct snd_soc_component *scomp, + struct snd_soc_dobj *dobj) +{ + struct snd_sof_route *sroute; + + sroute = dobj->private; + if (!sroute) + return 0; + + /* free sroute and its private data */ + kfree(sroute->private); + list_del(&sroute->list); + kfree(sroute); + + return 0; +} + +static int sof_widget_unload(struct snd_soc_component *scomp, + struct snd_soc_dobj *dobj) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + const struct snd_kcontrol_new *kc; + struct snd_soc_dapm_widget *widget; + struct sof_ipc_pipe_new *pipeline; + struct snd_sof_control *scontrol; + struct snd_sof_widget *swidget; + struct soc_mixer_control *sm; + struct soc_bytes_ext *sbe; + struct snd_sof_dai *dai; + struct soc_enum *se; + int ret = 0; + int i; + + swidget = dobj->private; + if (!swidget) + return 0; + + widget = swidget->widget; + + switch (swidget->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + dai = swidget->private; + + if (dai) { + /* free dai config */ + kfree(dai->dai_config); + list_del(&dai->list); + } + break; + case snd_soc_dapm_scheduler: + + /* power down the pipeline schedule core */ + pipeline = swidget->private; + ret = snd_sof_dsp_core_power_down(sdev, 1 << pipeline->core); + if (ret < 0) + dev_err(sdev->dev, "error: powering down pipeline schedule core %d\n", + pipeline->core); + + /* update enabled cores mask */ + sdev->enabled_cores_mask &= ~(1 << pipeline->core); + + break; + default: + break; + } + for (i = 0; i < widget->num_kcontrols; i++) { + kc = &widget->kcontrol_news[i]; + switch (dobj->widget.kcontrol_type) { + case SND_SOC_TPLG_TYPE_MIXER: + sm = (struct soc_mixer_control *)kc->private_value; + scontrol = sm->dobj.private; + if (sm->max > 1) + kfree(scontrol->volume_table); + break; + case SND_SOC_TPLG_TYPE_ENUM: + se = (struct soc_enum *)kc->private_value; + scontrol = se->dobj.private; + break; + case SND_SOC_TPLG_TYPE_BYTES: + sbe = (struct soc_bytes_ext *)kc->private_value; + scontrol = sbe->dobj.private; + break; + default: + dev_warn(sdev->dev, "unsupported kcontrol_type\n"); + goto out; + } + kfree(scontrol->control_data); + list_del(&scontrol->list); + kfree(scontrol); + } + +out: + /* free private value */ + kfree(swidget->private); + + /* remove and free swidget object */ + list_del(&swidget->list); + kfree(swidget); + + return ret; +} + +/* + * DAI HW configuration. + */ + +/* FE DAI - used for any driver specific init */ +static int sof_dai_load(struct snd_soc_component *scomp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_stream_caps *caps; + struct snd_sof_pcm *spcm; + int stream = SNDRV_PCM_STREAM_PLAYBACK; + int ret = 0; + + /* nothing to do for BEs atm */ + if (!pcm) + return 0; + + spcm = kzalloc(sizeof(*spcm), GFP_KERNEL); + if (!spcm) + return -ENOMEM; + + spcm->sdev = sdev; + spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].comp_id = COMP_ID_UNASSIGNED; + spcm->stream[SNDRV_PCM_STREAM_CAPTURE].comp_id = COMP_ID_UNASSIGNED; + + if (pcm) { + spcm->pcm = *pcm; + dev_dbg(sdev->dev, "tplg: load pcm %s\n", pcm->dai_name); + } + dai_drv->dobj.private = spcm; + list_add(&spcm->list, &sdev->pcm_list); + + /* do we need to allocate playback PCM DMA pages */ + if (!spcm->pcm.playback) + goto capture; + + caps = &spcm->pcm.caps[stream]; + + /* allocate playback page table buffer */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, sdev->dev, + PAGE_SIZE, &spcm->stream[stream].page_table); + if (ret < 0) { + dev_err(sdev->dev, "error: can't alloc page table for %s %d\n", + caps->name, ret); + + return ret; + } + + /* bind pcm to host comp */ + ret = spcm_bind(sdev, spcm, stream); + if (ret) { + dev_err(sdev->dev, + "error: can't bind pcm to host\n"); + goto free_playback_tables; + } + +capture: + stream = SNDRV_PCM_STREAM_CAPTURE; + + /* do we need to allocate capture PCM DMA pages */ + if (!spcm->pcm.capture) + return ret; + + caps = &spcm->pcm.caps[stream]; + + /* allocate capture page table buffer */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, sdev->dev, + PAGE_SIZE, &spcm->stream[stream].page_table); + if (ret < 0) { + dev_err(sdev->dev, "error: can't alloc page table for %s %d\n", + caps->name, ret); + goto free_playback_tables; + } + + /* bind pcm to host comp */ + ret = spcm_bind(sdev, spcm, stream); + if (ret) { + dev_err(sdev->dev, + "error: can't bind pcm to host\n"); + snd_dma_free_pages(&spcm->stream[stream].page_table); + goto free_playback_tables; + } + + return ret; + +free_playback_tables: + if (spcm->pcm.playback) + snd_dma_free_pages(&spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].page_table); + + return ret; +} + +static int sof_dai_unload(struct snd_soc_component *scomp, + struct snd_soc_dobj *dobj) +{ + struct snd_sof_pcm *spcm = dobj->private; + + /* free PCM DMA pages */ + if (spcm->pcm.playback) + snd_dma_free_pages(&spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].page_table); + + if (spcm->pcm.capture) + snd_dma_free_pages(&spcm->stream[SNDRV_PCM_STREAM_CAPTURE].page_table); + + /* remove from list and free spcm */ + list_del(&spcm->list); + kfree(spcm); + + return 0; +} + +static void sof_dai_set_format(struct snd_soc_tplg_hw_config *hw_config, + struct sof_ipc_dai_config *config) +{ + /* clock directions wrt codec */ + if (hw_config->bclk_master == SND_SOC_TPLG_BCLK_CM) { + /* codec is bclk master */ + if (hw_config->fsync_master == SND_SOC_TPLG_FSYNC_CM) + config->format |= SOF_DAI_FMT_CBM_CFM; + else + config->format |= SOF_DAI_FMT_CBM_CFS; + } else { + /* codec is bclk slave */ + if (hw_config->fsync_master == SND_SOC_TPLG_FSYNC_CM) + config->format |= SOF_DAI_FMT_CBS_CFM; + else + config->format |= SOF_DAI_FMT_CBS_CFS; + } + + /* inverted clocks ? */ + if (hw_config->invert_bclk) { + if (hw_config->invert_fsync) + config->format |= SOF_DAI_FMT_IB_IF; + else + config->format |= SOF_DAI_FMT_IB_NF; + } else { + if (hw_config->invert_fsync) + config->format |= SOF_DAI_FMT_NB_IF; + else + config->format |= SOF_DAI_FMT_NB_NF; + } +} + +/* set config for all DAI's with name matching the link name */ +static int sof_set_dai_config(struct snd_sof_dev *sdev, u32 size, + struct snd_soc_dai_link *link, + struct sof_ipc_dai_config *config) +{ + struct snd_sof_dai *dai; + int found = 0; + + list_for_each_entry(dai, &sdev->dai_list, list) { + if (!dai->name) + continue; + + if (strcmp(link->name, dai->name) == 0) { + dai->dai_config = kmemdup(config, size, GFP_KERNEL); + if (!dai->dai_config) + return -ENOMEM; + + found = 1; + } + } + + /* + * machine driver may define a dai link with playback and capture + * dai enabled, but the dai link in topology would support both, one + * or none of them. Here print a warning message to notify user + */ + if (!found) { + dev_warn(sdev->dev, "warning: failed to find dai for dai link %s", + link->name); + } + + return 0; +} + +static int sof_link_ssp_load(struct snd_soc_component *scomp, int index, + struct snd_soc_dai_link *link, + struct snd_soc_tplg_link_config *cfg, + struct snd_soc_tplg_hw_config *hw_config, + struct sof_ipc_dai_config *config) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &cfg->priv; + struct sof_ipc_reply reply; + u32 size = sizeof(*config); + int ret; + + /* handle master/slave and inverted clocks */ + sof_dai_set_format(hw_config, config); + + /* init IPC */ + memset(&config->ssp, 0, sizeof(struct sof_ipc_dai_ssp_params)); + config->hdr.size = size; + + ret = sof_parse_tokens(scomp, &config->ssp, ssp_tokens, + ARRAY_SIZE(ssp_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse ssp tokens failed %d\n", + le32_to_cpu(private->size)); + return ret; + } + + config->ssp.mclk_rate = le32_to_cpu(hw_config->mclk_rate); + config->ssp.bclk_rate = le32_to_cpu(hw_config->bclk_rate); + config->ssp.fsync_rate = le32_to_cpu(hw_config->fsync_rate); + config->ssp.tdm_slots = le32_to_cpu(hw_config->tdm_slots); + config->ssp.tdm_slot_width = le32_to_cpu(hw_config->tdm_slot_width); + config->ssp.mclk_direction = hw_config->mclk_direction; + config->ssp.rx_slots = le32_to_cpu(hw_config->rx_slots); + config->ssp.tx_slots = le32_to_cpu(hw_config->tx_slots); + + dev_dbg(sdev->dev, "tplg: config SSP%d fmt 0x%x mclk %d bclk %d fclk %d width (%d)%d slots %d mclk id %d quirks %d\n", + config->dai_index, config->format, + config->ssp.mclk_rate, config->ssp.bclk_rate, + config->ssp.fsync_rate, config->ssp.sample_valid_bits, + config->ssp.tdm_slot_width, config->ssp.tdm_slots, + config->ssp.mclk_id, config->ssp.quirks); + + /* validate SSP fsync rate and channel count */ + if (config->ssp.fsync_rate < 8000 || config->ssp.fsync_rate > 192000) { + dev_err(sdev->dev, "error: invalid fsync rate for SSP%d\n", + config->dai_index); + return -EINVAL; + } + + if (config->ssp.tdm_slots < 1 || config->ssp.tdm_slots > 8) { + dev_err(sdev->dev, "error: invalid channel count for SSP%d\n", + config->dai_index); + return -EINVAL; + } + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config->hdr.cmd, config, size, &reply, + sizeof(reply)); + + if (ret < 0) { + dev_err(sdev->dev, "error: failed to set DAI config for SSP%d\n", + config->dai_index); + return ret; + } + + /* set config for all DAI's with name matching the link name */ + ret = sof_set_dai_config(sdev, size, link, config); + if (ret < 0) + dev_err(sdev->dev, "error: failed to save DAI config for SSP%d\n", + config->dai_index); + + return ret; +} + +static int sof_link_dmic_load(struct snd_soc_component *scomp, int index, + struct snd_soc_dai_link *link, + struct snd_soc_tplg_link_config *cfg, + struct snd_soc_tplg_hw_config *hw_config, + struct sof_ipc_dai_config *config) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &cfg->priv; + struct sof_ipc_dai_config *ipc_config; + struct sof_ipc_reply reply; + struct sof_ipc_fw_ready *ready = &sdev->fw_ready; + struct sof_ipc_fw_version *v = &ready->version; + u32 size; + int ret, j; + + /* + * config is only used for the common params in dmic_params structure + * that does not include the PDM controller config array + * Set the common params to 0. + */ + memset(&config->dmic, 0, sizeof(struct sof_ipc_dai_dmic_params)); + + /* get DMIC tokens */ + ret = sof_parse_tokens(scomp, &config->dmic, dmic_tokens, + ARRAY_SIZE(dmic_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse dmic tokens failed %d\n", + le32_to_cpu(private->size)); + return ret; + } + + /* + * allocate memory for dmic dai config accounting for the + * variable number of active pdm controllers + * This will be the ipc payload for setting dai config + */ + size = sizeof(*config) + sizeof(struct sof_ipc_dai_dmic_pdm_ctrl) * + config->dmic.num_pdm_active; + + ipc_config = kzalloc(size, GFP_KERNEL); + if (!ipc_config) + return -ENOMEM; + + /* copy the common dai config and dmic params */ + memcpy(ipc_config, config, sizeof(*config)); + + /* + * alloc memory for private member + * Used to track the pdm config array index currently being parsed + */ + sdev->private = kzalloc(sizeof(u32), GFP_KERNEL); + if (!sdev->private) { + kfree(ipc_config); + return -ENOMEM; + } + + /* get DMIC PDM tokens */ + ret = sof_parse_tokens(scomp, &ipc_config->dmic.pdm[0], dmic_pdm_tokens, + ARRAY_SIZE(dmic_pdm_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse dmic pdm tokens failed %d\n", + le32_to_cpu(private->size)); + goto err; + } + + /* set IPC header size */ + ipc_config->hdr.size = size; + + /* debug messages */ + dev_dbg(sdev->dev, "tplg: config DMIC%d driver version %d\n", + ipc_config->dai_index, ipc_config->dmic.driver_ipc_version); + dev_dbg(sdev->dev, "pdmclk_min %d pdm_clkmax %d duty_min %hd\n", + ipc_config->dmic.pdmclk_min, ipc_config->dmic.pdmclk_max, + ipc_config->dmic.duty_min); + dev_dbg(sdev->dev, "duty_max %hd fifo_fs %d num_pdms active %d\n", + ipc_config->dmic.duty_max, ipc_config->dmic.fifo_fs, + ipc_config->dmic.num_pdm_active); + dev_dbg(sdev->dev, "fifo word length %hd\n", + ipc_config->dmic.fifo_bits); + + for (j = 0; j < ipc_config->dmic.num_pdm_active; j++) { + dev_dbg(sdev->dev, "pdm %hd mic a %hd mic b %hd\n", + ipc_config->dmic.pdm[j].id, + ipc_config->dmic.pdm[j].enable_mic_a, + ipc_config->dmic.pdm[j].enable_mic_b); + dev_dbg(sdev->dev, "pdm %hd polarity a %hd polarity b %hd\n", + ipc_config->dmic.pdm[j].id, + ipc_config->dmic.pdm[j].polarity_mic_a, + ipc_config->dmic.pdm[j].polarity_mic_b); + dev_dbg(sdev->dev, "pdm %hd clk_edge %hd skew %hd\n", + ipc_config->dmic.pdm[j].id, + ipc_config->dmic.pdm[j].clk_edge, + ipc_config->dmic.pdm[j].skew); + } + + if (SOF_ABI_VER(v->major, v->minor, v->micro) < SOF_ABI_VER(3, 0, 1)) { + /* this takes care of backwards compatible handling of fifo_bits_b */ + ipc_config->dmic.reserved_2 = ipc_config->dmic.fifo_bits; + } + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + ipc_config->hdr.cmd, ipc_config, size, &reply, + sizeof(reply)); + + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to set DAI config for DMIC%d\n", + config->dai_index); + goto err; + } + + /* set config for all DAI's with name matching the link name */ + ret = sof_set_dai_config(sdev, size, link, ipc_config); + if (ret < 0) + dev_err(sdev->dev, "error: failed to save DAI config for DMIC%d\n", + config->dai_index); + +err: + kfree(sdev->private); + kfree(ipc_config); + + return ret; +} + +/* + * for hda link, playback and capture are supported by different dai + * in FW. Here get the dai_index, set dma channel of each dai + * and send config to FW. In FW, each dai sets config by dai_index + */ +static int sof_link_hda_process(struct snd_sof_dev *sdev, + struct snd_soc_dai_link *link, + struct sof_ipc_dai_config *config, + int tx_slot, + int rx_slot) +{ + struct sof_ipc_reply reply; + u32 size = sizeof(*config); + struct snd_sof_dai *sof_dai; + int found = 0; + int ret; + + list_for_each_entry(sof_dai, &sdev->dai_list, list) { + if (!sof_dai->name) + continue; + + if (strcmp(link->name, sof_dai->name) == 0) { + if (sof_dai->comp_dai.direction == + SNDRV_PCM_STREAM_PLAYBACK) { + if (!link->dpcm_playback) + return -EINVAL; + + config->hda.link_dma_ch = tx_slot; + } else { + if (!link->dpcm_capture) + return -EINVAL; + + config->hda.link_dma_ch = rx_slot; + } + + config->dai_index = sof_dai->comp_dai.dai_index; + found = 1; + + /* save config in dai component */ + sof_dai->dai_config = kmemdup(config, size, GFP_KERNEL); + if (!sof_dai->dai_config) + return -ENOMEM; + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config->hdr.cmd, config, size, + &reply, sizeof(reply)); + + if (ret < 0) { + dev_err(sdev->dev, "error: failed to set DAI config for direction:%d of HDA dai %d\n", + sof_dai->comp_dai.direction, + config->dai_index); + + return ret; + } + } + } + + /* + * machine driver may define a dai link with playback and capture + * dai enabled, but the dai link in topology would support both, one + * or none of them. Here print a warning message to notify user + */ + if (!found) { + dev_warn(sdev->dev, "warning: failed to find dai for dai link %s", + link->name); + } + + return 0; +} + +static int sof_link_hda_load(struct snd_soc_component *scomp, int index, + struct snd_soc_dai_link *link, + struct snd_soc_tplg_link_config *cfg, + struct snd_soc_tplg_hw_config *hw_config, + struct sof_ipc_dai_config *config) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_dai_link_component dai_component; + struct snd_soc_tplg_private *private = &cfg->priv; + struct snd_soc_dai *dai; + u32 size = sizeof(*config); + u32 tx_num = 0; + u32 tx_slot = 0; + u32 rx_num = 0; + u32 rx_slot = 0; + int ret; + + /* init IPC */ + memset(&dai_component, 0, sizeof(dai_component)); + memset(&config->hda, 0, sizeof(struct sof_ipc_dai_hda_params)); + config->hdr.size = size; + + /* get any bespoke DAI tokens */ + ret = sof_parse_tokens(scomp, config, hda_tokens, + ARRAY_SIZE(hda_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse hda tokens failed %d\n", + le32_to_cpu(private->size)); + return ret; + } + + dai_component.dai_name = link->cpu_dai_name; + dai = snd_soc_find_dai(&dai_component); + if (!dai) { + dev_err(sdev->dev, "error: failed to find dai %s in %s", + dai_component.dai_name, __func__); + return -EINVAL; + } + + if (link->dpcm_playback) + tx_num = 1; + + if (link->dpcm_capture) + rx_num = 1; + + ret = snd_soc_dai_get_channel_map(dai, &tx_num, &tx_slot, + &rx_num, &rx_slot); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to get dma channel for HDA%d\n", + config->dai_index); + + return ret; + } + + ret = sof_link_hda_process(sdev, link, config, tx_slot, rx_slot); + if (ret < 0) + dev_err(sdev->dev, "error: failed to process hda dai link %s", + link->name); + + return ret; +} + +/* DAI link - used for any driver specific init */ +static int sof_link_load(struct snd_soc_component *scomp, int index, + struct snd_soc_dai_link *link, + struct snd_soc_tplg_link_config *cfg) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_tplg_private *private = &cfg->priv; + struct sof_ipc_dai_config config; + struct snd_soc_tplg_hw_config *hw_config; + int num_hw_configs; + int ret; + int i = 0; + + link->platform_name = dev_name(sdev->dev); + + /* + * Set nonatomic property for FE dai links as their trigger action + * involves IPC's. + */ + if (!link->no_pcm) { + link->nonatomic = true; + + /* nothing more to do for FE dai links */ + return 0; + } + + /* check we have some tokens - we need at least DAI type */ + if (le32_to_cpu(private->size) == 0) { + dev_err(sdev->dev, "error: expected tokens for DAI, none found\n"); + return -EINVAL; + } + + /* Send BE DAI link configurations to DSP */ + memset(&config, 0, sizeof(config)); + + /* get any common DAI tokens */ + ret = sof_parse_tokens(scomp, &config, dai_link_tokens, + ARRAY_SIZE(dai_link_tokens), private->array, + le32_to_cpu(private->size)); + if (ret != 0) { + dev_err(sdev->dev, "error: parse link tokens failed %d\n", + le32_to_cpu(private->size)); + return ret; + } + + /* + * DAI links are expected to have at least 1 hw_config. + * But some older topologies might have no hw_config for HDA dai links. + */ + num_hw_configs = le32_to_cpu(cfg->num_hw_configs); + if (!num_hw_configs) { + if (config.type != SOF_DAI_INTEL_HDA) { + dev_err(sdev->dev, "error: unexpected DAI config count %d!\n", + le32_to_cpu(cfg->num_hw_configs)); + return -EINVAL; + } + } else { + dev_dbg(sdev->dev, "tplg: %d hw_configs found, default id: %d!\n", + cfg->num_hw_configs, le32_to_cpu(cfg->default_hw_config_id)); + + for (i = 0; i < num_hw_configs; i++) { + if (cfg->hw_config[i].id == cfg->default_hw_config_id) + break; + } + + if (i == num_hw_configs) { + dev_err(sdev->dev, "error: default hw_config id: %d not found!\n", + le32_to_cpu(cfg->default_hw_config_id)); + return -EINVAL; + } + } + + /* configure dai IPC message */ + hw_config = &cfg->hw_config[i]; + + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.format = le32_to_cpu(hw_config->fmt); + + /* now load DAI specific data and send IPC - type comes from token */ + switch (config.type) { + case SOF_DAI_INTEL_SSP: + ret = sof_link_ssp_load(scomp, index, link, cfg, hw_config, + &config); + break; + case SOF_DAI_INTEL_DMIC: + ret = sof_link_dmic_load(scomp, index, link, cfg, hw_config, + &config); + break; + case SOF_DAI_INTEL_HDA: + ret = sof_link_hda_load(scomp, index, link, cfg, hw_config, + &config); + break; + default: + dev_err(sdev->dev, "error: invalid DAI type %d\n", config.type); + ret = -EINVAL; + break; + } + if (ret < 0) + return ret; + + return 0; +} + +static int sof_link_hda_unload(struct snd_sof_dev *sdev, + struct snd_soc_dai_link *link) +{ + struct snd_soc_dai_link_component dai_component; + struct snd_soc_dai *dai; + int ret = 0; + + memset(&dai_component, 0, sizeof(dai_component)); + dai_component.dai_name = link->cpu_dai_name; + dai = snd_soc_find_dai(&dai_component); + if (!dai) { + dev_err(sdev->dev, "error: failed to find dai %s in %s", + dai_component.dai_name, __func__); + return -EINVAL; + } + + /* + * FIXME: this call to hw_free is mainly to release the link DMA ID. + * This is abusing the API and handling SOC internals is not + * recommended. This part will be reworked. + */ + if (dai->driver->ops->hw_free) + ret = dai->driver->ops->hw_free(NULL, dai); + if (ret < 0) + dev_err(sdev->dev, "error: failed to free hda resource for %s\n", + link->name); + + return ret; +} + +static int sof_link_unload(struct snd_soc_component *scomp, + struct snd_soc_dobj *dobj) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_soc_dai_link *link = + container_of(dobj, struct snd_soc_dai_link, dobj); + + struct snd_sof_dai *sof_dai; + int ret = 0; + + /* only BE link is loaded by sof */ + if (!link->no_pcm) + return 0; + + list_for_each_entry(sof_dai, &sdev->dai_list, list) { + if (!sof_dai->name) + continue; + + if (strcmp(link->name, sof_dai->name) == 0) + goto found; + } + + dev_err(sdev->dev, "error: failed to find dai %s in %s", + link->name, __func__); + return -EINVAL; +found: + + switch (sof_dai->dai_config->type) { + case SOF_DAI_INTEL_SSP: + case SOF_DAI_INTEL_DMIC: + /* no resource needs to be released for SSP and DMIC */ + break; + case SOF_DAI_INTEL_HDA: + ret = sof_link_hda_unload(sdev, link); + break; + default: + dev_err(sdev->dev, "error: invalid DAI type %d\n", + sof_dai->dai_config->type); + ret = -EINVAL; + break; + } + + return ret; +} + +/* DAI link - used for any driver specific init */ +static int sof_route_load(struct snd_soc_component *scomp, int index, + struct snd_soc_dapm_route *route) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct sof_ipc_pipe_comp_connect *connect; + struct snd_sof_widget *source_swidget, *sink_swidget; + struct snd_soc_dobj *dobj = &route->dobj; + struct snd_sof_route *sroute; + struct sof_ipc_reply reply; + int ret = 0; + + /* allocate memory for sroute and connect */ + sroute = kzalloc(sizeof(*sroute), GFP_KERNEL); + if (!sroute) + return -ENOMEM; + + sroute->sdev = sdev; + + connect = kzalloc(sizeof(*connect), GFP_KERNEL); + if (!connect) { + kfree(sroute); + return -ENOMEM; + } + + connect->hdr.size = sizeof(*connect); + connect->hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_CONNECT; + + dev_dbg(sdev->dev, "sink %s control %s source %s\n", + route->sink, route->control ? route->control : "none", + route->source); + + /* source component */ + source_swidget = snd_sof_find_swidget(sdev, (char *)route->source); + if (!source_swidget) { + dev_err(sdev->dev, "error: source %s not found\n", + route->source); + ret = -EINVAL; + goto err; + } + + /* + * Virtual widgets of type output/out_drv may be added in topology + * for compatibility. These are not handled by the FW. + * So, don't send routes whose source/sink widget is of such types + * to the DSP. + */ + if (source_swidget->id == snd_soc_dapm_out_drv || + source_swidget->id == snd_soc_dapm_output) + goto err; + + connect->source_id = source_swidget->comp_id; + + /* sink component */ + sink_swidget = snd_sof_find_swidget(sdev, (char *)route->sink); + if (!sink_swidget) { + dev_err(sdev->dev, "error: sink %s not found\n", + route->sink); + ret = -EINVAL; + goto err; + } + + /* + * Don't send routes whose sink widget is of type + * output or out_drv to the DSP + */ + if (sink_swidget->id == snd_soc_dapm_out_drv || + sink_swidget->id == snd_soc_dapm_output) + goto err; + + connect->sink_id = sink_swidget->comp_id; + + /* + * For virtual routes, both sink and source are not + * buffer. Since only buffer linked to component is supported by + * FW, others are reported as error, add check in route function, + * do not send it to FW when both source and sink are not buffer + */ + if (source_swidget->id != snd_soc_dapm_buffer && + sink_swidget->id != snd_soc_dapm_buffer) { + dev_dbg(sdev->dev, "warning: neither Linked source component %s nor sink component %s is of buffer type, ignoring link\n", + route->source, route->sink); + ret = 0; + goto err; + } else { + ret = sof_ipc_tx_message(sdev->ipc, + connect->hdr.cmd, + connect, sizeof(*connect), + &reply, sizeof(reply)); + + /* check IPC return value */ + if (ret < 0) { + dev_err(sdev->dev, "error: failed to add route sink %s control %s source %s\n", + route->sink, + route->control ? route->control : "none", + route->source); + goto err; + } + + /* check IPC reply */ + if (reply.error < 0) { + dev_err(sdev->dev, "error: DSP failed to add route sink %s control %s source %s result %d\n", + route->sink, + route->control ? route->control : "none", + route->source, reply.error); + ret = reply.error; + goto err; + } + + sroute->route = route; + dobj->private = sroute; + sroute->private = connect; + + /* add route to route list */ + list_add(&sroute->list, &sdev->route_list); + + return ret; + } + +err: + kfree(connect); + kfree(sroute); + return ret; +} + +int snd_sof_complete_pipeline(struct snd_sof_dev *sdev, + struct snd_sof_widget *swidget) +{ + struct sof_ipc_pipe_ready ready; + struct sof_ipc_reply reply; + int ret; + + dev_dbg(sdev->dev, "tplg: complete pipeline %s id %d\n", + swidget->widget->name, swidget->comp_id); + + memset(&ready, 0, sizeof(ready)); + ready.hdr.size = sizeof(ready); + ready.hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_PIPE_COMPLETE; + ready.comp_id = swidget->comp_id; + + ret = sof_ipc_tx_message(sdev->ipc, + ready.hdr.cmd, &ready, sizeof(ready), &reply, + sizeof(reply)); + if (ret < 0) + return ret; + return 1; +} + +/* completion - called at completion of firmware loading */ +static void sof_complete(struct snd_soc_component *scomp) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + struct snd_sof_widget *swidget; + + /* some widget types require completion notificattion */ + list_for_each_entry(swidget, &sdev->widget_list, list) { + if (swidget->complete) + continue; + + switch (swidget->id) { + case snd_soc_dapm_scheduler: + swidget->complete = + snd_sof_complete_pipeline(sdev, swidget); + break; + default: + break; + } + } +} + +/* manifest - optional to inform component of manifest */ +static int sof_manifest(struct snd_soc_component *scomp, int index, + struct snd_soc_tplg_manifest *man) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); + u32 size; + + size = le32_to_cpu(man->priv.size); + + /* backward compatible with tplg without ABI info */ + if (!size) { + dev_dbg(sdev->dev, "No topology ABI info\n"); + return 0; + } + + if (size == SOF_TPLG_ABI_SIZE) { + dev_info(sdev->dev, + "Topology: ABI %d:%d:%d Kernel ABI %d:%d:%d\n", + man->priv.data[0], man->priv.data[1], + man->priv.data[2], SOF_ABI_MAJOR, SOF_ABI_MINOR, + SOF_ABI_PATCH); + if (SOF_ABI_VER(man->priv.data[0], man->priv.data[1], + man->priv.data[2]) <= SOF_ABI_VERSION) + return 0; + } + dev_err(sdev->dev, + "error: Incompatible ABI version %d:%d:%d\n", + man->priv.data[0], man->priv.data[1], man->priv.data[2]); + return -EINVAL; +} + +/* vendor specific kcontrol handlers available for binding */ +static const struct snd_soc_tplg_kcontrol_ops sof_io_ops[] = { + {SOF_TPLG_KCTL_VOL_ID, snd_sof_volume_get, snd_sof_volume_put}, + {SOF_TPLG_KCTL_BYTES_ID, snd_sof_bytes_get, snd_sof_bytes_put}, + {SOF_TPLG_KCTL_ENUM_ID, snd_sof_enum_get, snd_sof_enum_put}, + {SOF_TPLG_KCTL_SWITCH_ID, snd_sof_switch_get, snd_sof_switch_put}, +}; + +/* vendor specific bytes ext handlers available for binding */ +static const struct snd_soc_tplg_bytes_ext_ops sof_bytes_ext_ops[] = { + {SOF_TPLG_KCTL_BYTES_ID, snd_sof_bytes_ext_get, snd_sof_bytes_ext_put}, +}; + +static struct snd_soc_tplg_ops sof_tplg_ops = { + /* external kcontrol init - used for any driver specific init */ + .control_load = sof_control_load, + .control_unload = sof_control_unload, + + /* external kcontrol init - used for any driver specific init */ + .dapm_route_load = sof_route_load, + .dapm_route_unload = sof_route_unload, + + /* external widget init - used for any driver specific init */ + /* .widget_load is not currently used */ + .widget_ready = sof_widget_ready, + .widget_unload = sof_widget_unload, + + /* FE DAI - used for any driver specific init */ + .dai_load = sof_dai_load, + .dai_unload = sof_dai_unload, + + /* DAI link - used for any driver specific init */ + .link_load = sof_link_load, + .link_unload = sof_link_unload, + + /* completion - called at completion of firmware loading */ + .complete = sof_complete, + + /* manifest - optional to inform component of manifest */ + .manifest = sof_manifest, + + /* vendor specific kcontrol handlers available for binding */ + .io_ops = sof_io_ops, + .io_ops_count = ARRAY_SIZE(sof_io_ops), + + /* vendor specific bytes ext handlers available for binding */ + .bytes_ext_ops = sof_bytes_ext_ops, + .bytes_ext_ops_count = ARRAY_SIZE(sof_bytes_ext_ops), +}; + +int snd_sof_init_topology(struct snd_sof_dev *sdev, + struct snd_soc_tplg_ops *ops) +{ + /* TODO: support linked list of topologies */ + sdev->tplg_ops = ops; + return 0; +} +EXPORT_SYMBOL(snd_sof_init_topology); + +int snd_sof_load_topology(struct snd_sof_dev *sdev, const char *file) +{ + const struct firmware *fw; + int ret; + + dev_dbg(sdev->dev, "loading topology:%s\n", file); + + ret = request_firmware(&fw, file, sdev->dev); + if (ret < 0) { + dev_err(sdev->dev, "error: tplg request firmware %s failed err: %d\n", + file, ret); + return ret; + } + + ret = snd_soc_tplg_component_load(sdev->component, + &sof_tplg_ops, fw, + SND_SOC_TPLG_INDEX_ALL); + if (ret < 0) { + dev_err(sdev->dev, "error: tplg component load failed %d\n", + ret); + ret = -EINVAL; + } + + release_firmware(fw); + return ret; +} +EXPORT_SYMBOL(snd_sof_load_topology); -- cgit v1.2.3-59-g8ed1b From fd51c47b352b7da0e78fc88b9e4da7f298bfdd70 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:12 -0500 Subject: ASoC: SOF: Add DSP firmware logger support This patch adds support for real-time DSP logging (timestamped events and bespoke binary data) for firmware debug. The current solution relies on DMA transfers to system memory that is then accessed by userspace tools such as sof-logger. For Intel platforms, two types of DMAs are currently used (GP-DMA for Baytrail/CherryTrail and HDaudio DMA for SKL+) Due to historical reasons, the driver code follows the DSP firmware conventions and refers to 'traces', but it is currently unrelated to the Linux trace subsystem. Future solutions will include support for more advanced hardware (e.g. MIPI Sys-T), additional formats and the ability to enable/disable specific traces dynamically. Signed-off-by: Pan Xiuli Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/trace.c | 297 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 297 insertions(+) create mode 100644 sound/soc/sof/trace.c (limited to 'sound') diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c new file mode 100644 index 000000000000..d588e4b70fad --- /dev/null +++ b/sound/soc/sof/trace.c @@ -0,0 +1,297 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +#include +#include +#include "sof-priv.h" +#include "ops.h" + +static size_t sof_wait_trace_avail(struct snd_sof_dev *sdev, + loff_t pos, size_t buffer_size) +{ + wait_queue_entry_t wait; + loff_t host_offset = READ_ONCE(sdev->host_offset); + + /* + * If host offset is less than local pos, it means write pointer of + * host DMA buffer has been wrapped. We should output the trace data + * at the end of host DMA buffer at first. + */ + if (host_offset < pos) + return buffer_size - pos; + + /* If there is available trace data now, it is unnecessary to wait. */ + if (host_offset > pos) + return host_offset - pos; + + /* wait for available trace data from FW */ + init_waitqueue_entry(&wait, current); + set_current_state(TASK_INTERRUPTIBLE); + add_wait_queue(&sdev->trace_sleep, &wait); + + if (!signal_pending(current)) { + /* set timeout to max value, no error code */ + schedule_timeout(MAX_SCHEDULE_TIMEOUT); + } + remove_wait_queue(&sdev->trace_sleep, &wait); + + /* return bytes available for copy */ + host_offset = READ_ONCE(sdev->host_offset); + if (host_offset < pos) + return buffer_size - pos; + + return host_offset - pos; +} + +static ssize_t sof_dfsentry_trace_read(struct file *file, char __user *buffer, + size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + unsigned long rem; + loff_t lpos = *ppos; + size_t avail, buffer_size = dfse->size; + u64 lpos_64; + + /* make sure we know about any failures on the DSP side */ + sdev->dtrace_error = false; + + /* check pos and count */ + if (lpos < 0) + return -EINVAL; + if (!count) + return 0; + + /* check for buffer wrap and count overflow */ + lpos_64 = lpos; + lpos = do_div(lpos_64, buffer_size); + + if (count > buffer_size - lpos) /* min() not used to avoid sparse warnings */ + count = buffer_size - lpos; + + /* get available count based on current host offset */ + avail = sof_wait_trace_avail(sdev, lpos, buffer_size); + if (sdev->dtrace_error) { + dev_err(sdev->dev, "error: trace IO error\n"); + return -EIO; + } + + /* make sure count is <= avail */ + count = avail > count ? count : avail; + + /* copy available trace data to debugfs */ + rem = copy_to_user(buffer, ((u8 *)(dfse->buf) + lpos), count); + if (rem) + return -EFAULT; + + *ppos += count; + + /* move debugfs reading position */ + return count; +} + +static const struct file_operations sof_dfs_trace_fops = { + .open = simple_open, + .read = sof_dfsentry_trace_read, + .llseek = default_llseek, +}; + +static int trace_debugfs_create(struct snd_sof_dev *sdev) +{ + struct snd_sof_dfsentry *dfse; + + if (!sdev) + return -EINVAL; + + dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL); + if (!dfse) + return -ENOMEM; + + dfse->type = SOF_DFSENTRY_TYPE_BUF; + dfse->buf = sdev->dmatb.area; + dfse->size = sdev->dmatb.bytes; + dfse->sdev = sdev; + + dfse->dfsentry = debugfs_create_file("trace", 0444, sdev->debugfs_root, + dfse, &sof_dfs_trace_fops); + if (!dfse->dfsentry) { + /* can't rely on debugfs, only log error and keep going */ + dev_err(sdev->dev, + "error: cannot create debugfs entry for trace\n"); + } + + return 0; +} + +int snd_sof_init_trace_ipc(struct snd_sof_dev *sdev) +{ + struct sof_ipc_dma_trace_params params; + struct sof_ipc_reply ipc_reply; + int ret; + + if (sdev->dtrace_is_enabled || !sdev->dma_trace_pages) + return -EINVAL; + + /* set IPC parameters */ + params.hdr.size = sizeof(params); + params.hdr.cmd = SOF_IPC_GLB_TRACE_MSG | SOF_IPC_TRACE_DMA_PARAMS; + params.buffer.phy_addr = sdev->dmatp.addr; + params.buffer.size = sdev->dmatb.bytes; + params.buffer.pages = sdev->dma_trace_pages; + params.stream_tag = 0; + + sdev->host_offset = 0; + + ret = snd_sof_dma_trace_init(sdev, ¶ms.stream_tag); + if (ret < 0) { + dev_err(sdev->dev, + "error: fail in snd_sof_dma_trace_init %d\n", ret); + return ret; + } + dev_dbg(sdev->dev, "stream_tag: %d\n", params.stream_tag); + + /* send IPC to the DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + params.hdr.cmd, ¶ms, sizeof(params), + &ipc_reply, sizeof(ipc_reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: can't set params for DMA for trace %d\n", ret); + goto trace_release; + } + + ret = snd_sof_dma_trace_trigger(sdev, SNDRV_PCM_TRIGGER_START); + if (ret < 0) { + dev_err(sdev->dev, + "error: snd_sof_dma_trace_trigger: start: %d\n", ret); + goto trace_release; + } + + sdev->dtrace_is_enabled = true; + + return 0; + +trace_release: + snd_sof_dma_trace_release(sdev); + return ret; +} + +int snd_sof_init_trace(struct snd_sof_dev *sdev) +{ + int ret; + + /* set false before start initialization */ + sdev->dtrace_is_enabled = false; + + /* allocate trace page table buffer */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, sdev->dev, + PAGE_SIZE, &sdev->dmatp); + if (ret < 0) { + dev_err(sdev->dev, + "error: can't alloc page table for trace %d\n", ret); + return ret; + } + + /* allocate trace data buffer */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, sdev->dev, + DMA_BUF_SIZE_FOR_TRACE, &sdev->dmatb); + if (ret < 0) { + dev_err(sdev->dev, + "error: can't alloc buffer for trace %d\n", ret); + goto page_err; + } + + /* create compressed page table for audio firmware */ + ret = snd_sof_create_page_table(sdev, &sdev->dmatb, sdev->dmatp.area, + sdev->dmatb.bytes); + if (ret < 0) + goto table_err; + + sdev->dma_trace_pages = ret; + dev_dbg(sdev->dev, "dma_trace_pages: %d\n", sdev->dma_trace_pages); + + if (sdev->first_boot) { + ret = trace_debugfs_create(sdev); + if (ret < 0) + goto table_err; + } + + init_waitqueue_head(&sdev->trace_sleep); + + ret = snd_sof_init_trace_ipc(sdev); + if (ret < 0) + goto table_err; + + return 0; +table_err: + sdev->dma_trace_pages = 0; + snd_dma_free_pages(&sdev->dmatb); +page_err: + snd_dma_free_pages(&sdev->dmatp); + return ret; +} +EXPORT_SYMBOL(snd_sof_init_trace); + +int snd_sof_trace_update_pos(struct snd_sof_dev *sdev, + struct sof_ipc_dma_trace_posn *posn) +{ + if (sdev->dtrace_is_enabled && sdev->host_offset != posn->host_offset) { + sdev->host_offset = posn->host_offset; + wake_up(&sdev->trace_sleep); + } + + if (posn->overflow != 0) + dev_err(sdev->dev, + "error: DSP trace buffer overflow %u bytes. Total messages %d\n", + posn->overflow, posn->messages); + + return 0; +} + +/* an error has occurred within the DSP that prevents further trace */ +void snd_sof_trace_notify_for_error(struct snd_sof_dev *sdev) +{ + if (sdev->dtrace_is_enabled) { + dev_err(sdev->dev, "error: waking up any trace sleepers\n"); + sdev->dtrace_error = true; + wake_up(&sdev->trace_sleep); + } +} +EXPORT_SYMBOL(snd_sof_trace_notify_for_error); + +void snd_sof_release_trace(struct snd_sof_dev *sdev) +{ + int ret; + + if (!sdev->dtrace_is_enabled) + return; + + ret = snd_sof_dma_trace_trigger(sdev, SNDRV_PCM_TRIGGER_STOP); + if (ret < 0) + dev_err(sdev->dev, + "error: snd_sof_dma_trace_trigger: stop: %d\n", ret); + + ret = snd_sof_dma_trace_release(sdev); + if (ret < 0) + dev_err(sdev->dev, + "error: fail in snd_sof_dma_trace_release %d\n", ret); + + sdev->dtrace_is_enabled = false; +} +EXPORT_SYMBOL(snd_sof_release_trace); + +void snd_sof_free_trace(struct snd_sof_dev *sdev) +{ + snd_sof_release_trace(sdev); + + snd_dma_free_pages(&sdev->dmatb); + snd_dma_free_pages(&sdev->dmatp); +} +EXPORT_SYMBOL(snd_sof_free_trace); -- cgit v1.2.3-59-g8ed1b From d1d95fcb63e3b83245ad06484b6905ab6c21afc3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:13 -0500 Subject: ASoC: SOF: Add DSP HW abstraction operations Add operation pointers that can be called by core to control a wide variety of DSP targets. The DSP HW drivers will fill in these operations. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/ops.c | 163 +++++++++++++++++++++ sound/soc/sof/ops.h | 399 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 562 insertions(+) create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h (limited to 'sound') diff --git a/sound/soc/sof/ops.c b/sound/soc/sof/ops.c new file mode 100644 index 000000000000..80f907740b82 --- /dev/null +++ b/sound/soc/sof/ops.c @@ -0,0 +1,163 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +#include +#include "ops.h" + +static +bool snd_sof_pci_update_bits_unlocked(struct snd_sof_dev *sdev, u32 offset, + u32 mask, u32 value) +{ + struct pci_dev *pci = to_pci_dev(sdev->dev); + unsigned int old, new; + u32 ret; + + pci_read_config_dword(pci, offset, &ret); + old = ret; + dev_dbg(sdev->dev, "Debug PCIR: %8.8x at %8.8x\n", old & mask, offset); + + new = (old & ~mask) | (value & mask); + + if (old == new) + return false; + + pci_write_config_dword(pci, offset, new); + dev_dbg(sdev->dev, "Debug PCIW: %8.8x at %8.8x\n", value, + offset); + + return true; +} + +bool snd_sof_pci_update_bits(struct snd_sof_dev *sdev, u32 offset, + u32 mask, u32 value) +{ + unsigned long flags; + bool change; + + spin_lock_irqsave(&sdev->hw_lock, flags); + change = snd_sof_pci_update_bits_unlocked(sdev, offset, mask, value); + spin_unlock_irqrestore(&sdev->hw_lock, flags); + return change; +} +EXPORT_SYMBOL(snd_sof_pci_update_bits); + +bool snd_sof_dsp_update_bits_unlocked(struct snd_sof_dev *sdev, u32 bar, + u32 offset, u32 mask, u32 value) +{ + unsigned int old, new; + u32 ret; + + ret = snd_sof_dsp_read(sdev, bar, offset); + + old = ret; + new = (old & ~mask) | (value & mask); + + if (old == new) + return false; + + snd_sof_dsp_write(sdev, bar, offset, new); + + return true; +} +EXPORT_SYMBOL(snd_sof_dsp_update_bits_unlocked); + +bool snd_sof_dsp_update_bits64_unlocked(struct snd_sof_dev *sdev, u32 bar, + u32 offset, u64 mask, u64 value) +{ + u64 old, new; + + old = snd_sof_dsp_read64(sdev, bar, offset); + + new = (old & ~mask) | (value & mask); + + if (old == new) + return false; + + snd_sof_dsp_write64(sdev, bar, offset, new); + + return true; +} +EXPORT_SYMBOL(snd_sof_dsp_update_bits64_unlocked); + +/* This is for registers bits with attribute RWC */ +bool snd_sof_dsp_update_bits(struct snd_sof_dev *sdev, u32 bar, u32 offset, + u32 mask, u32 value) +{ + unsigned long flags; + bool change; + + spin_lock_irqsave(&sdev->hw_lock, flags); + change = snd_sof_dsp_update_bits_unlocked(sdev, bar, offset, mask, + value); + spin_unlock_irqrestore(&sdev->hw_lock, flags); + return change; +} +EXPORT_SYMBOL(snd_sof_dsp_update_bits); + +bool snd_sof_dsp_update_bits64(struct snd_sof_dev *sdev, u32 bar, u32 offset, + u64 mask, u64 value) +{ + unsigned long flags; + bool change; + + spin_lock_irqsave(&sdev->hw_lock, flags); + change = snd_sof_dsp_update_bits64_unlocked(sdev, bar, offset, mask, + value); + spin_unlock_irqrestore(&sdev->hw_lock, flags); + return change; +} +EXPORT_SYMBOL(snd_sof_dsp_update_bits64); + +static +void snd_sof_dsp_update_bits_forced_unlocked(struct snd_sof_dev *sdev, u32 bar, + u32 offset, u32 mask, u32 value) +{ + unsigned int old, new; + u32 ret; + + ret = snd_sof_dsp_read(sdev, bar, offset); + + old = ret; + new = (old & ~mask) | (value & mask); + + snd_sof_dsp_write(sdev, bar, offset, new); +} + +/* This is for registers bits with attribute RWC */ +void snd_sof_dsp_update_bits_forced(struct snd_sof_dev *sdev, u32 bar, + u32 offset, u32 mask, u32 value) +{ + unsigned long flags; + + spin_lock_irqsave(&sdev->hw_lock, flags); + snd_sof_dsp_update_bits_forced_unlocked(sdev, bar, offset, mask, value); + spin_unlock_irqrestore(&sdev->hw_lock, flags); +} +EXPORT_SYMBOL(snd_sof_dsp_update_bits_forced); + +void snd_sof_dsp_panic(struct snd_sof_dev *sdev, u32 offset) +{ + dev_err(sdev->dev, "error : DSP panic!\n"); + + /* + * check if DSP is not ready and did not set the dsp_oops_offset. + * if the dsp_oops_offset is not set, set it from the panic message. + * Also add a check to memory window setting with panic message. + */ + if (!sdev->dsp_oops_offset) + sdev->dsp_oops_offset = offset; + else + dev_dbg(sdev->dev, "panic: dsp_oops_offset %zu offset %d\n", + sdev->dsp_oops_offset, offset); + + snd_sof_dsp_dbg_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX); + snd_sof_trace_notify_for_error(sdev); +} +EXPORT_SYMBOL(snd_sof_dsp_panic); diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h new file mode 100644 index 000000000000..2a5d4c63f160 --- /dev/null +++ b/sound/soc/sof/ops.h @@ -0,0 +1,399 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + * + * Author: Liam Girdwood + */ + +#ifndef __SOUND_SOC_SOF_IO_H +#define __SOUND_SOC_SOF_IO_H + +#include +#include +#include +#include +#include +#include "sof-priv.h" + +#define sof_ops(sdev) \ + ((sdev)->pdata->desc->ops) + +/* Mandatory operations are verified during probing */ + +/* init */ +static inline int snd_sof_probe(struct snd_sof_dev *sdev) +{ + return sof_ops(sdev)->probe(sdev); +} + +static inline int snd_sof_remove(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->remove) + return sof_ops(sdev)->remove(sdev); + + return 0; +} + +/* control */ + +/* + * snd_sof_dsp_run returns the core mask of the cores that are available + * after successful fw boot + */ +static inline int snd_sof_dsp_run(struct snd_sof_dev *sdev) +{ + return sof_ops(sdev)->run(sdev); +} + +static inline int snd_sof_dsp_stall(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->stall) + return sof_ops(sdev)->stall(sdev); + + return 0; +} + +static inline int snd_sof_dsp_reset(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->reset) + return sof_ops(sdev)->reset(sdev); + + return 0; +} + +/* dsp core power up/power down */ +static inline int snd_sof_dsp_core_power_up(struct snd_sof_dev *sdev, + unsigned int core_mask) +{ + if (sof_ops(sdev)->core_power_up) + return sof_ops(sdev)->core_power_up(sdev, core_mask); + + return 0; +} + +static inline int snd_sof_dsp_core_power_down(struct snd_sof_dev *sdev, + unsigned int core_mask) +{ + if (sof_ops(sdev)->core_power_down) + return sof_ops(sdev)->core_power_down(sdev, core_mask); + + return 0; +} + +/* pre/post fw load */ +static inline int snd_sof_dsp_pre_fw_run(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->pre_fw_run) + return sof_ops(sdev)->pre_fw_run(sdev); + + return 0; +} + +static inline int snd_sof_dsp_post_fw_run(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->post_fw_run) + return sof_ops(sdev)->post_fw_run(sdev); + + return 0; +} + +/* power management */ +static inline int snd_sof_dsp_resume(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->resume) + return sof_ops(sdev)->resume(sdev); + + return 0; +} + +static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev, int state) +{ + if (sof_ops(sdev)->suspend) + return sof_ops(sdev)->suspend(sdev, state); + + return 0; +} + +static inline int snd_sof_dsp_runtime_resume(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->runtime_resume) + return sof_ops(sdev)->runtime_resume(sdev); + + return 0; +} + +static inline int snd_sof_dsp_runtime_suspend(struct snd_sof_dev *sdev, + int state) +{ + if (sof_ops(sdev)->runtime_suspend) + return sof_ops(sdev)->runtime_suspend(sdev, state); + + return 0; +} + +static inline int snd_sof_dsp_set_clk(struct snd_sof_dev *sdev, u32 freq) +{ + if (sof_ops(sdev)->set_clk) + return sof_ops(sdev)->set_clk(sdev, freq); + + return 0; +} + +/* debug */ +static inline void snd_sof_dsp_dbg_dump(struct snd_sof_dev *sdev, u32 flags) +{ + if (sof_ops(sdev)->dbg_dump) + return sof_ops(sdev)->dbg_dump(sdev, flags); +} + +/* register IO */ +static inline void snd_sof_dsp_write(struct snd_sof_dev *sdev, u32 bar, + u32 offset, u32 value) +{ + if (sof_ops(sdev)->write) { + sof_ops(sdev)->write(sdev, sdev->bar[bar] + offset, value); + return; + } + + dev_err_ratelimited(sdev->dev, "error: %s not defined\n", __func__); +} + +static inline void snd_sof_dsp_write64(struct snd_sof_dev *sdev, u32 bar, + u32 offset, u64 value) +{ + if (sof_ops(sdev)->write64) { + sof_ops(sdev)->write64(sdev, sdev->bar[bar] + offset, value); + return; + } + + dev_err_ratelimited(sdev->dev, "error: %s not defined\n", __func__); +} + +static inline u32 snd_sof_dsp_read(struct snd_sof_dev *sdev, u32 bar, + u32 offset) +{ + if (sof_ops(sdev)->read) + return sof_ops(sdev)->read(sdev, sdev->bar[bar] + offset); + + dev_err(sdev->dev, "error: %s not defined\n", __func__); + return -ENOTSUPP; +} + +static inline u64 snd_sof_dsp_read64(struct snd_sof_dev *sdev, u32 bar, + u32 offset) +{ + if (sof_ops(sdev)->read64) + return sof_ops(sdev)->read64(sdev, sdev->bar[bar] + offset); + + dev_err(sdev->dev, "error: %s not defined\n", __func__); + return -ENOTSUPP; +} + +/* block IO */ +static inline void snd_sof_dsp_block_read(struct snd_sof_dev *sdev, u32 bar, + u32 offset, void *dest, size_t bytes) +{ + sof_ops(sdev)->block_read(sdev, bar, offset, dest, bytes); +} + +static inline void snd_sof_dsp_block_write(struct snd_sof_dev *sdev, u32 bar, + u32 offset, void *src, size_t bytes) +{ + sof_ops(sdev)->block_write(sdev, bar, offset, src, bytes); +} + +/* ipc */ +static inline int snd_sof_dsp_send_msg(struct snd_sof_dev *sdev, + struct snd_sof_ipc_msg *msg) +{ + return sof_ops(sdev)->send_msg(sdev, msg); +} + +/* host DMA trace */ +static inline int snd_sof_dma_trace_init(struct snd_sof_dev *sdev, + u32 *stream_tag) +{ + if (sof_ops(sdev)->trace_init) + return sof_ops(sdev)->trace_init(sdev, stream_tag); + + return 0; +} + +static inline int snd_sof_dma_trace_release(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->trace_release) + return sof_ops(sdev)->trace_release(sdev); + + return 0; +} + +static inline int snd_sof_dma_trace_trigger(struct snd_sof_dev *sdev, int cmd) +{ + if (sof_ops(sdev)->trace_trigger) + return sof_ops(sdev)->trace_trigger(sdev, cmd); + + return 0; +} + +/* host PCM ops */ +static inline int +snd_sof_pcm_platform_open(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream) +{ + if (sof_ops(sdev) && sof_ops(sdev)->pcm_open) + return sof_ops(sdev)->pcm_open(sdev, substream); + + return 0; +} + +/* disconnect pcm substream to a host stream */ +static inline int +snd_sof_pcm_platform_close(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream) +{ + if (sof_ops(sdev) && sof_ops(sdev)->pcm_close) + return sof_ops(sdev)->pcm_close(sdev, substream); + + return 0; +} + +/* host stream hw params */ +static inline int +snd_sof_pcm_platform_hw_params(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct sof_ipc_stream_params *ipc_params) +{ + if (sof_ops(sdev) && sof_ops(sdev)->pcm_hw_params) + return sof_ops(sdev)->pcm_hw_params(sdev, substream, + params, ipc_params); + + return 0; +} + +/* host stream trigger */ +static inline int +snd_sof_pcm_platform_trigger(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, int cmd) +{ + if (sof_ops(sdev) && sof_ops(sdev)->pcm_trigger) + return sof_ops(sdev)->pcm_trigger(sdev, substream, cmd); + + return 0; +} + +/* host DSP message data */ +static inline void snd_sof_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz) +{ + sof_ops(sdev)->ipc_msg_data(sdev, substream, p, sz); +} + +/* host configure DSP HW parameters */ +static inline int +snd_sof_ipc_pcm_params(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + const struct sof_ipc_pcm_params_reply *reply) +{ + return sof_ops(sdev)->ipc_pcm_params(sdev, substream, reply); +} + +/* host stream pointer */ +static inline snd_pcm_uframes_t +snd_sof_pcm_platform_pointer(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream) +{ + if (sof_ops(sdev) && sof_ops(sdev)->pcm_pointer) + return sof_ops(sdev)->pcm_pointer(sdev, substream); + + return 0; +} + +static inline const struct snd_sof_dsp_ops +*sof_get_ops(const struct sof_dev_desc *d, + const struct sof_ops_table mach_ops[], int asize) +{ + int i; + + for (i = 0; i < asize; i++) { + if (d == mach_ops[i].desc) + return mach_ops[i].ops; + } + + /* not found */ + return NULL; +} + +/** + * snd_sof_dsp_register_poll_timeout - Periodically poll an address + * until a condition is met or a timeout occurs + * @op: accessor function (takes @addr as its only argument) + * @addr: Address to poll + * @val: Variable to read the value into + * @cond: Break condition (usually involving @val) + * @sleep_us: Maximum time to sleep between reads in us (0 + * tight-loops). Should be less than ~20ms since usleep_range + * is used (see Documentation/timers/timers-howto.txt). + * @timeout_us: Timeout in us, 0 means never timeout + * + * Returns 0 on success and -ETIMEDOUT upon a timeout. In either + * case, the last read value at @addr is stored in @val. Must not + * be called from atomic context if sleep_us or timeout_us are used. + * + * This is modelled after the readx_poll_timeout macros in linux/iopoll.h. + */ +#define snd_sof_dsp_read_poll_timeout(sdev, bar, offset, val, cond, sleep_us, timeout_us) \ +({ \ + u64 __timeout_us = (timeout_us); \ + unsigned long __sleep_us = (sleep_us); \ + ktime_t __timeout = ktime_add_us(ktime_get(), __timeout_us); \ + might_sleep_if((__sleep_us) != 0); \ + for (;;) { \ + (val) = snd_sof_dsp_read(sdev, bar, offset); \ + if (cond) { \ + dev_dbg(sdev->dev, \ + "FW Poll Status: reg=%#x successful\n", (val)); \ + break; \ + } \ + if (__timeout_us && \ + ktime_compare(ktime_get(), __timeout) > 0) { \ + (val) = snd_sof_dsp_read(sdev, bar, offset); \ + dev_dbg(sdev->dev, \ + "FW Poll Status: reg=%#x timedout\n", (val)); \ + break; \ + } \ + if (__sleep_us) \ + usleep_range((__sleep_us >> 2) + 1, __sleep_us); \ + } \ + (cond) ? 0 : -ETIMEDOUT; \ +}) + +/* This is for registers bits with attribute RWC */ +bool snd_sof_pci_update_bits(struct snd_sof_dev *sdev, u32 offset, + u32 mask, u32 value); + +bool snd_sof_dsp_update_bits_unlocked(struct snd_sof_dev *sdev, u32 bar, + u32 offset, u32 mask, u32 value); + +bool snd_sof_dsp_update_bits64_unlocked(struct snd_sof_dev *sdev, u32 bar, + u32 offset, u64 mask, u64 value); + +bool snd_sof_dsp_update_bits(struct snd_sof_dev *sdev, u32 bar, u32 offset, + u32 mask, u32 value); + +bool snd_sof_dsp_update_bits64(struct snd_sof_dev *sdev, u32 bar, + u32 offset, u64 mask, u64 value); + +void snd_sof_dsp_update_bits_forced(struct snd_sof_dev *sdev, u32 bar, + u32 offset, u32 mask, u32 value); + +int snd_sof_dsp_register_poll(struct snd_sof_dev *sdev, u32 bar, u32 offset, + u32 mask, u32 target, u32 timeout_ms, + u32 interval_us); + +void snd_sof_dsp_panic(struct snd_sof_dev *sdev, u32 offset); +#endif -- cgit v1.2.3-59-g8ed1b From 70cd52544b95d02f8b20fadccb7400ce68a2cb84 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:14 -0500 Subject: ASoC: SOF: Add firmware loader support The firmware loader exports APIs that can be called by core to load and process multiple different file formats. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/loader.c | 400 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 400 insertions(+) create mode 100644 sound/soc/sof/loader.c (limited to 'sound') diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c new file mode 100644 index 000000000000..81c7452aae17 --- /dev/null +++ b/sound/soc/sof/loader.c @@ -0,0 +1,400 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// +// Generic firmware loader. +// + +#include +#include +#include "ops.h" + +static int get_ext_windows(struct snd_sof_dev *sdev, + struct sof_ipc_ext_data_hdr *ext_hdr) +{ + struct sof_ipc_window *w = + container_of(ext_hdr, struct sof_ipc_window, ext_hdr); + size_t size; + + if (w->num_windows == 0 || w->num_windows > SOF_IPC_MAX_ELEMS) + return -EINVAL; + + size = sizeof(*w) + sizeof(struct sof_ipc_window_elem) * w->num_windows; + + /* keep a local copy of the data */ + sdev->info_window = kmemdup(w, size, GFP_KERNEL); + if (!sdev->info_window) + return -ENOMEM; + + return 0; +} + +/* parse the extended FW boot data structures from FW boot message */ +int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) +{ + struct sof_ipc_ext_data_hdr *ext_hdr; + void *ext_data; + int ret = 0; + + ext_data = kzalloc(PAGE_SIZE, GFP_KERNEL); + if (!ext_data) + return -ENOMEM; + + /* get first header */ + snd_sof_dsp_block_read(sdev, bar, offset, ext_data, + sizeof(*ext_hdr)); + ext_hdr = ext_data; + + while (ext_hdr->hdr.cmd == SOF_IPC_FW_READY) { + /* read in ext structure */ + offset += sizeof(*ext_hdr); + snd_sof_dsp_block_read(sdev, bar, offset, + (void *)((u8 *)ext_data + sizeof(*ext_hdr)), + ext_hdr->hdr.size - sizeof(*ext_hdr)); + + dev_dbg(sdev->dev, "found ext header type %d size 0x%x\n", + ext_hdr->type, ext_hdr->hdr.size); + + /* process structure data */ + switch (ext_hdr->type) { + case SOF_IPC_EXT_DMA_BUFFER: + break; + case SOF_IPC_EXT_WINDOW: + ret = get_ext_windows(sdev, ext_hdr); + break; + default: + break; + } + + if (ret < 0) { + dev_err(sdev->dev, "error: failed to parse ext data type %d\n", + ext_hdr->type); + break; + } + + /* move to next header */ + offset += ext_hdr->hdr.size; + snd_sof_dsp_block_read(sdev, bar, offset, ext_data, + sizeof(*ext_hdr)); + ext_hdr = ext_data; + } + + kfree(ext_data); + return ret; +} +EXPORT_SYMBOL(snd_sof_fw_parse_ext_data); + +/* generic module parser for mmaped DSPs */ +int snd_sof_parse_module_memcpy(struct snd_sof_dev *sdev, + struct snd_sof_mod_hdr *module) +{ + struct snd_sof_blk_hdr *block; + int count; + u32 offset; + size_t remaining; + + dev_dbg(sdev->dev, "new module size 0x%x blocks 0x%x type 0x%x\n", + module->size, module->num_blocks, module->type); + + block = (struct snd_sof_blk_hdr *)((u8 *)module + sizeof(*module)); + + /* module->size doesn't include header size */ + remaining = module->size; + for (count = 0; count < module->num_blocks; count++) { + /* check for wrap */ + if (remaining < sizeof(*block)) { + dev_err(sdev->dev, "error: not enough data remaining\n"); + return -EINVAL; + } + + /* minus header size of block */ + remaining -= sizeof(*block); + + if (block->size == 0) { + dev_warn(sdev->dev, + "warning: block %d size zero\n", count); + dev_warn(sdev->dev, " type 0x%x offset 0x%x\n", + block->type, block->offset); + continue; + } + + switch (block->type) { + case SOF_FW_BLK_TYPE_RSRVD0: + case SOF_FW_BLK_TYPE_SRAM...SOF_FW_BLK_TYPE_RSRVD14: + continue; /* not handled atm */ + case SOF_FW_BLK_TYPE_IRAM: + case SOF_FW_BLK_TYPE_DRAM: + offset = block->offset; + break; + default: + dev_err(sdev->dev, "error: bad type 0x%x for block 0x%x\n", + block->type, count); + return -EINVAL; + } + + dev_dbg(sdev->dev, + "block %d type 0x%x size 0x%x ==> offset 0x%x\n", + count, block->type, block->size, offset); + + /* checking block->size to avoid unaligned access */ + if (block->size % sizeof(u32)) { + dev_err(sdev->dev, "error: invalid block size 0x%x\n", + block->size); + return -EINVAL; + } + snd_sof_dsp_block_write(sdev, sdev->mmio_bar, offset, + block + 1, block->size); + + if (remaining < block->size) { + dev_err(sdev->dev, "error: not enough data remaining\n"); + return -EINVAL; + } + + /* minus body size of block */ + remaining -= block->size; + /* next block */ + block = (struct snd_sof_blk_hdr *)((u8 *)block + sizeof(*block) + + block->size); + } + + return 0; +} +EXPORT_SYMBOL(snd_sof_parse_module_memcpy); + +static int check_header(struct snd_sof_dev *sdev, const struct firmware *fw) +{ + struct snd_sof_fw_header *header; + + /* Read the header information from the data pointer */ + header = (struct snd_sof_fw_header *)fw->data; + + /* verify FW sig */ + if (strncmp(header->sig, SND_SOF_FW_SIG, SND_SOF_FW_SIG_SIZE) != 0) { + dev_err(sdev->dev, "error: invalid firmware signature\n"); + return -EINVAL; + } + + /* check size is valid */ + if (fw->size != header->file_size + sizeof(*header)) { + dev_err(sdev->dev, "error: invalid filesize mismatch got 0x%zx expected 0x%zx\n", + fw->size, header->file_size + sizeof(*header)); + return -EINVAL; + } + + dev_dbg(sdev->dev, "header size=0x%x modules=0x%x abi=0x%x size=%zu\n", + header->file_size, header->num_modules, + header->abi, sizeof(*header)); + + return 0; +} + +static int load_modules(struct snd_sof_dev *sdev, const struct firmware *fw) +{ + struct snd_sof_fw_header *header; + struct snd_sof_mod_hdr *module; + int (*load_module)(struct snd_sof_dev *sof_dev, + struct snd_sof_mod_hdr *hdr); + int ret, count; + size_t remaining; + + header = (struct snd_sof_fw_header *)fw->data; + load_module = sof_ops(sdev)->load_module; + if (!load_module) + return -EINVAL; + + /* parse each module */ + module = (struct snd_sof_mod_hdr *)((u8 *)(fw->data) + sizeof(*header)); + remaining = fw->size - sizeof(*header); + /* check for wrap */ + if (remaining > fw->size) { + dev_err(sdev->dev, "error: fw size smaller than header size\n"); + return -EINVAL; + } + + for (count = 0; count < header->num_modules; count++) { + /* check for wrap */ + if (remaining < sizeof(*module)) { + dev_err(sdev->dev, "error: not enough data remaining\n"); + return -EINVAL; + } + + /* minus header size of module */ + remaining -= sizeof(*module); + + /* module */ + ret = load_module(sdev, module); + if (ret < 0) { + dev_err(sdev->dev, "error: invalid module %d\n", count); + return ret; + } + + if (remaining < module->size) { + dev_err(sdev->dev, "error: not enough data remaining\n"); + return -EINVAL; + } + + /* minus body size of module */ + remaining -= module->size; + module = (struct snd_sof_mod_hdr *)((u8 *)module + + sizeof(*module) + module->size); + } + + return 0; +} + +int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *plat_data = sdev->pdata; + const char *fw_filename; + int ret; + + /* set code loading condition to true */ + sdev->code_loading = 1; + + /* Don't request firmware again if firmware is already requested */ + if (plat_data->fw) + return 0; + + fw_filename = kasprintf(GFP_KERNEL, "%s/%s", + plat_data->fw_filename_prefix, + plat_data->fw_filename); + if (!fw_filename) + return -ENOMEM; + + ret = request_firmware(&plat_data->fw, fw_filename, sdev->dev); + + if (ret < 0) { + dev_err(sdev->dev, "error: request firmware %s failed err: %d\n", + fw_filename, ret); + } + + kfree(fw_filename); + + return ret; +} +EXPORT_SYMBOL(snd_sof_load_firmware_raw); + +int snd_sof_load_firmware_memcpy(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *plat_data = sdev->pdata; + int ret; + + ret = snd_sof_load_firmware_raw(sdev); + if (ret < 0) + return ret; + + /* make sure the FW header and file is valid */ + ret = check_header(sdev, plat_data->fw); + if (ret < 0) { + dev_err(sdev->dev, "error: invalid FW header\n"); + goto error; + } + + /* prepare the DSP for FW loading */ + ret = snd_sof_dsp_reset(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to reset DSP\n"); + goto error; + } + + /* parse and load firmware modules to DSP */ + ret = load_modules(sdev, plat_data->fw); + if (ret < 0) { + dev_err(sdev->dev, "error: invalid FW modules\n"); + goto error; + } + + return 0; + +error: + release_firmware(plat_data->fw); + plat_data->fw = NULL; + return ret; + +} +EXPORT_SYMBOL(snd_sof_load_firmware_memcpy); + +int snd_sof_load_firmware(struct snd_sof_dev *sdev) +{ + dev_dbg(sdev->dev, "loading firmware\n"); + + if (sof_ops(sdev)->load_firmware) + return sof_ops(sdev)->load_firmware(sdev); + return 0; +} +EXPORT_SYMBOL(snd_sof_load_firmware); + +int snd_sof_run_firmware(struct snd_sof_dev *sdev) +{ + int ret; + int init_core_mask; + + init_waitqueue_head(&sdev->boot_wait); + sdev->boot_complete = false; + + /* create fw_version debugfs to store boot version info */ + if (sdev->first_boot) { + ret = snd_sof_debugfs_buf_item(sdev, &sdev->fw_version, + sizeof(sdev->fw_version), + "fw_version"); + /* errors are only due to memory allocation, not debugfs */ + if (ret < 0) { + dev_err(sdev->dev, "error: snd_sof_debugfs_buf_item failed\n"); + return ret; + } + } + + /* perform pre fw run operations */ + ret = snd_sof_dsp_pre_fw_run(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed pre fw run op\n"); + return ret; + } + + dev_dbg(sdev->dev, "booting DSP firmware\n"); + + /* boot the firmware on the DSP */ + ret = snd_sof_dsp_run(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to reset DSP\n"); + return ret; + } + + init_core_mask = ret; + + /* now wait for the DSP to boot */ + ret = wait_event_timeout(sdev->boot_wait, sdev->boot_complete, + msecs_to_jiffies(sdev->boot_timeout)); + if (ret == 0) { + dev_err(sdev->dev, "error: firmware boot failure\n"); + snd_sof_dsp_dbg_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX | + SOF_DBG_TEXT | SOF_DBG_PCI); + return -EIO; + } + + dev_info(sdev->dev, "firmware boot complete\n"); + + /* perform post fw run operations */ + ret = snd_sof_dsp_post_fw_run(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed post fw run op\n"); + return ret; + } + + /* fw boot is complete. Update the active cores mask */ + sdev->enabled_cores_mask = init_core_mask; + + return 0; +} +EXPORT_SYMBOL(snd_sof_run_firmware); + +void snd_sof_fw_unload(struct snd_sof_dev *sdev) +{ + /* TODO: support module unloading at runtime */ +} +EXPORT_SYMBOL(snd_sof_fw_unload); -- cgit v1.2.3-59-g8ed1b From 8920153c6461da69e49014a35275dc80cbf6ba14 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:16 -0500 Subject: ASoC: SOF: Add PM support Add support for saving and restoring DSP context in D3 to host DDR. The suspend callback includes: suspend all pcm's stream that are running, send CTX_SAVE ipc, drop all ipc's, release trace dma and then power off the DSP. And the resume callback performs the following steps: load FW, run FW, re-initialize trace, restore pipeline, restore the kcontrol values and finally send the ctx restore ipc to the dsp. The streams that are suspended are resumed by the ALSA resume trigger. If the streams are paused during system suspend, they are marked explicitly so they can be restored during PAUSE_RELEASE. Signed-off-by: Ranjani Sridharan Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 385 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 385 insertions(+) create mode 100644 sound/soc/sof/pm.c (limited to 'sound') diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c new file mode 100644 index 000000000000..fc599e1b6f65 --- /dev/null +++ b/sound/soc/sof/pm.c @@ -0,0 +1,385 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +#include "ops.h" +#include "sof-priv.h" + +static int sof_restore_kcontrols(struct snd_sof_dev *sdev) +{ + struct snd_sof_control *scontrol; + int ipc_cmd, ctrl_type; + int ret = 0; + + /* restore kcontrol values */ + list_for_each_entry(scontrol, &sdev->kcontrol_list, list) { + /* reset readback offset for scontrol after resuming */ + scontrol->readback_offset = 0; + + /* notify DSP of kcontrol values */ + switch (scontrol->cmd) { + case SOF_CTRL_CMD_VOLUME: + case SOF_CTRL_CMD_ENUM: + case SOF_CTRL_CMD_SWITCH: + ipc_cmd = SOF_IPC_COMP_SET_VALUE; + ctrl_type = SOF_CTRL_TYPE_VALUE_CHAN_SET; + ret = snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + ipc_cmd, ctrl_type, + scontrol->cmd, + true); + break; + case SOF_CTRL_CMD_BINARY: + ipc_cmd = SOF_IPC_COMP_SET_DATA; + ctrl_type = SOF_CTRL_TYPE_DATA_SET; + ret = snd_sof_ipc_set_get_comp_data(sdev->ipc, scontrol, + ipc_cmd, ctrl_type, + scontrol->cmd, + true); + break; + + default: + break; + } + + if (ret < 0) { + dev_err(sdev->dev, + "error: failed kcontrol value set for widget: %d\n", + scontrol->comp_id); + + return ret; + } + } + + return 0; +} + +static int sof_restore_pipelines(struct snd_sof_dev *sdev) +{ + struct snd_sof_widget *swidget; + struct snd_sof_route *sroute; + struct sof_ipc_pipe_new *pipeline; + struct snd_sof_dai *dai; + struct sof_ipc_comp_dai *comp_dai; + struct sof_ipc_cmd_hdr *hdr; + int ret; + + /* restore pipeline components */ + list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { + struct sof_ipc_comp_reply r; + + /* skip if there is no private data */ + if (!swidget->private) + continue; + + switch (swidget->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + dai = swidget->private; + comp_dai = &dai->comp_dai; + ret = sof_ipc_tx_message(sdev->ipc, + comp_dai->comp.hdr.cmd, + comp_dai, sizeof(*comp_dai), + &r, sizeof(r)); + break; + case snd_soc_dapm_scheduler: + + /* + * During suspend, all DSP cores are powered off. + * Therefore upon resume, create the pipeline comp + * and power up the core that the pipeline is + * scheduled on. + */ + pipeline = swidget->private; + ret = sof_load_pipeline_ipc(sdev, pipeline, &r); + break; + default: + hdr = swidget->private; + ret = sof_ipc_tx_message(sdev->ipc, hdr->cmd, + swidget->private, hdr->size, + &r, sizeof(r)); + break; + } + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to load widget type %d with ID: %d\n", + swidget->widget->id, swidget->comp_id); + + return ret; + } + } + + /* restore pipeline connections */ + list_for_each_entry_reverse(sroute, &sdev->route_list, list) { + struct sof_ipc_pipe_comp_connect *connect; + struct sof_ipc_reply reply; + + /* skip if there's no private data */ + if (!sroute->private) + continue; + + connect = sroute->private; + + /* send ipc */ + ret = sof_ipc_tx_message(sdev->ipc, + connect->hdr.cmd, + connect, sizeof(*connect), + &reply, sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to load route sink %s control %s source %s\n", + sroute->route->sink, + sroute->route->control ? sroute->route->control + : "none", + sroute->route->source); + + return ret; + } + } + + /* restore dai links */ + list_for_each_entry_reverse(dai, &sdev->dai_list, list) { + struct sof_ipc_reply reply; + struct sof_ipc_dai_config *config = dai->dai_config; + + if (!config) { + dev_err(sdev->dev, "error: no config for DAI %s\n", + dai->name); + continue; + } + + ret = sof_ipc_tx_message(sdev->ipc, + config->hdr.cmd, config, + config->hdr.size, + &reply, sizeof(reply)); + + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to set dai config for %s\n", + dai->name); + + return ret; + } + } + + /* complete pipeline */ + list_for_each_entry(swidget, &sdev->widget_list, list) { + switch (swidget->id) { + case snd_soc_dapm_scheduler: + swidget->complete = + snd_sof_complete_pipeline(sdev, swidget); + break; + default: + break; + } + } + + /* restore pipeline kcontrols */ + ret = sof_restore_kcontrols(sdev); + if (ret < 0) + dev_err(sdev->dev, + "error: restoring kcontrols after resume\n"); + + return ret; +} + +static int sof_send_pm_ipc(struct snd_sof_dev *sdev, int cmd) +{ + struct sof_ipc_pm_ctx pm_ctx; + struct sof_ipc_reply reply; + + memset(&pm_ctx, 0, sizeof(pm_ctx)); + + /* configure ctx save ipc message */ + pm_ctx.hdr.size = sizeof(pm_ctx); + pm_ctx.hdr.cmd = SOF_IPC_GLB_PM_MSG | cmd; + + /* send ctx save ipc to dsp */ + return sof_ipc_tx_message(sdev->ipc, pm_ctx.hdr.cmd, &pm_ctx, + sizeof(pm_ctx), &reply, sizeof(reply)); +} + +static void sof_set_hw_params_upon_resume(struct snd_sof_dev *sdev) +{ + struct snd_pcm_substream *substream; + struct snd_sof_pcm *spcm; + snd_pcm_state_t state; + int dir; + + /* + * SOF requires hw_params to be set-up internally upon resume. + * So, set the flag to indicate this for those streams that + * have been suspended. + */ + list_for_each_entry(spcm, &sdev->pcm_list, list) { + for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + substream = spcm->stream[dir].substream; + if (!substream || !substream->runtime) + continue; + + state = substream->runtime->status->state; + if (state == SNDRV_PCM_STATE_SUSPENDED) + spcm->hw_params_upon_resume[dir] = 1; + } + } +} + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) +static void sof_cache_debugfs(struct snd_sof_dev *sdev) +{ + struct snd_sof_dfsentry *dfse; + + list_for_each_entry(dfse, &sdev->dfsentry_list, list) { + + /* nothing to do if debugfs buffer is not IO mem */ + if (dfse->type == SOF_DFSENTRY_TYPE_BUF) + continue; + + /* cache memory that is only accessible in D0 */ + if (dfse->access_type == SOF_DEBUGFS_ACCESS_D0_ONLY) + memcpy_fromio(dfse->cache_buf, dfse->io_mem, + dfse->size); + } +} +#endif + +static int sof_resume(struct device *dev, bool runtime_resume) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + int ret; + + /* do nothing if dsp resume callbacks are not set */ + if (!sof_ops(sdev)->resume || !sof_ops(sdev)->runtime_resume) + return 0; + + /* + * if the runtime_resume flag is set, call the runtime_resume routine + * or else call the system resume routine + */ + if (runtime_resume) + ret = snd_sof_dsp_runtime_resume(sdev); + else + ret = snd_sof_dsp_resume(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to power up DSP after resume\n"); + return ret; + } + + /* load the firmware */ + ret = snd_sof_load_firmware(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to load DSP firmware after resume %d\n", + ret); + return ret; + } + + /* boot the firmware */ + ret = snd_sof_run_firmware(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to boot DSP firmware after resume %d\n", + ret); + return ret; + } + + /* resume DMA trace, only need send ipc */ + ret = snd_sof_init_trace_ipc(sdev); + if (ret < 0) { + /* non fatal */ + dev_warn(sdev->dev, + "warning: failed to init trace after resume %d\n", + ret); + } + + /* restore pipelines */ + ret = sof_restore_pipelines(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to restore pipeline after resume %d\n", + ret); + return ret; + } + + /* notify DSP of system resume */ + ret = sof_send_pm_ipc(sdev, SOF_IPC_PM_CTX_RESTORE); + if (ret < 0) + dev_err(sdev->dev, + "error: ctx_restore ipc error during resume %d\n", + ret); + + return ret; +} + +static int sof_suspend(struct device *dev, bool runtime_suspend) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + int ret; + + /* do nothing if dsp suspend callback is not set */ + if (!sof_ops(sdev)->suspend) + return 0; + + /* release trace */ + snd_sof_release_trace(sdev); + + /* set restore_stream for all streams during system suspend */ + if (!runtime_suspend) + sof_set_hw_params_upon_resume(sdev); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) + /* cache debugfs contents during runtime suspend */ + if (runtime_suspend) + sof_cache_debugfs(sdev); +#endif + /* notify DSP of upcoming power down */ + ret = sof_send_pm_ipc(sdev, SOF_IPC_PM_CTX_SAVE); + if (ret < 0) { + dev_err(sdev->dev, + "error: ctx_save ipc error during suspend %d\n", + ret); + return ret; + } + + /* power down all DSP cores */ + if (runtime_suspend) + ret = snd_sof_dsp_runtime_suspend(sdev, 0); + else + ret = snd_sof_dsp_suspend(sdev, 0); + if (ret < 0) + dev_err(sdev->dev, + "error: failed to power down DSP during suspend %d\n", + ret); + + return ret; +} + +int snd_sof_runtime_suspend(struct device *dev) +{ + return sof_suspend(dev, true); +} +EXPORT_SYMBOL(snd_sof_runtime_suspend); + +int snd_sof_runtime_resume(struct device *dev) +{ + return sof_resume(dev, true); +} +EXPORT_SYMBOL(snd_sof_runtime_resume); + +int snd_sof_resume(struct device *dev) +{ + return sof_resume(dev, false); +} +EXPORT_SYMBOL(snd_sof_resume); + +int snd_sof_suspend(struct device *dev) +{ + return sof_suspend(dev, false); +} +EXPORT_SYMBOL(snd_sof_suspend); -- cgit v1.2.3-59-g8ed1b From 8017b8fd37bf5ed4e9c0596f83295b019ff1c287 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:05:17 -0500 Subject: ASoC: SOF: Add Nocodec machine driver support Add a simple "fallback" machine driver that can be used to enable SOF on boards with no codec device. This machine driver can also be forced for debug/development. Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/sof.h | 6 +++ sound/soc/sof/nocodec.c | 109 ++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 115 insertions(+) create mode 100644 sound/soc/sof/nocodec.c (limited to 'sound') diff --git a/include/sound/sof.h b/include/sound/sof.h index 54f65ec33a6c..4640566b54fe 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -12,6 +12,7 @@ #define __INCLUDE_SOUND_SOF_H #include +#include #include struct snd_sof_dsp_ops; @@ -91,4 +92,9 @@ struct sof_dev_desc { const struct sof_arch_ops *arch_ops; }; +int sof_nocodec_setup(struct device *dev, + struct snd_sof_pdata *sof_pdata, + struct snd_soc_acpi_mach *mach, + const struct sof_dev_desc *desc, + const struct snd_sof_dsp_ops *ops); #endif diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c new file mode 100644 index 000000000000..f84b4344dcc3 --- /dev/null +++ b/sound/soc/sof/nocodec.c @@ -0,0 +1,109 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +#include +#include +#include "sof-priv.h" + +static struct snd_soc_card sof_nocodec_card = { + .name = "nocodec", /* the sof- prefix is added by the core */ +}; + +static int sof_nocodec_bes_setup(struct device *dev, + const struct snd_sof_dsp_ops *ops, + struct snd_soc_dai_link *links, + int link_num, struct snd_soc_card *card) +{ + int i; + + if (!ops || !links || !card) + return -EINVAL; + + /* set up BE dai_links */ + for (i = 0; i < link_num; i++) { + links[i].name = devm_kasprintf(dev, GFP_KERNEL, + "NoCodec-%d", i); + if (!links[i].name) + return -ENOMEM; + + links[i].id = i; + links[i].no_pcm = 1; + links[i].cpu_dai_name = ops->drv[i].name; + links[i].platform_name = dev_name(dev); + links[i].codec_dai_name = "snd-soc-dummy-dai"; + links[i].codec_name = "snd-soc-dummy"; + links[i].dpcm_playback = 1; + links[i].dpcm_capture = 1; + } + + card->dai_link = links; + card->num_links = link_num; + + return 0; +} + +int sof_nocodec_setup(struct device *dev, + struct snd_sof_pdata *sof_pdata, + struct snd_soc_acpi_mach *mach, + const struct sof_dev_desc *desc, + const struct snd_sof_dsp_ops *ops) +{ + struct snd_soc_dai_link *links; + int ret; + + if (!mach) + return -EINVAL; + + sof_pdata->drv_name = "sof-nocodec"; + + mach->drv_name = "sof-nocodec"; + sof_pdata->fw_filename = desc->nocodec_fw_filename; + sof_pdata->tplg_filename = desc->nocodec_tplg_filename; + + /* create dummy BE dai_links */ + links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * + ops->num_drv, GFP_KERNEL); + if (!links) + return -ENOMEM; + + ret = sof_nocodec_bes_setup(dev, ops, links, ops->num_drv, + &sof_nocodec_card); + return ret; +} +EXPORT_SYMBOL(sof_nocodec_setup); + +static int sof_nocodec_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &sof_nocodec_card; + + card->dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, card); +} + +static int sof_nocodec_remove(struct platform_device *pdev) +{ + return 0; +} + +static struct platform_driver sof_nocodec_audio = { + .probe = sof_nocodec_probe, + .remove = sof_nocodec_remove, + .driver = { + .name = "sof-nocodec", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(sof_nocodec_audio) + +MODULE_DESCRIPTION("ASoC sof nocodec"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_ALIAS("platform:sof-nocodec"); -- cgit v1.2.3-59-g8ed1b From e657c18a01c85d2c4ec0e96d52be8ba42b956593 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 12 Apr 2019 11:05:18 -0500 Subject: ASoC: SOF: Add xtensa support Add common directory for xtensa architecture Signed-off-by: Pan Xiuli Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/sof/xtensa.h | 44 ++++++++++++++ sound/soc/sof/xtensa/Kconfig | 2 + sound/soc/sof/xtensa/Makefile | 5 ++ sound/soc/sof/xtensa/core.c | 138 ++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 189 insertions(+) create mode 100644 include/sound/sof/xtensa.h create mode 100644 sound/soc/sof/xtensa/Kconfig create mode 100644 sound/soc/sof/xtensa/Makefile create mode 100644 sound/soc/sof/xtensa/core.c (limited to 'sound') diff --git a/include/sound/sof/xtensa.h b/include/sound/sof/xtensa.h new file mode 100644 index 000000000000..a7189984000d --- /dev/null +++ b/include/sound/sof/xtensa.h @@ -0,0 +1,44 @@ +/* SPDX-License-Identifier: ((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2018 Intel Corporation. All rights reserved. + */ + +#ifndef __INCLUDE_SOUND_SOF_XTENSA_H__ +#define __INCLUDE_SOUND_SOF_XTENSA_H__ + +#include + +/* + * Architecture specific debug + */ + +/* Xtensa Firmware Oops data */ +struct sof_ipc_dsp_oops_xtensa { + struct sof_ipc_hdr hdr; + uint32_t exccause; + uint32_t excvaddr; + uint32_t ps; + uint32_t epc1; + uint32_t epc2; + uint32_t epc3; + uint32_t epc4; + uint32_t epc5; + uint32_t epc6; + uint32_t epc7; + uint32_t eps2; + uint32_t eps3; + uint32_t eps4; + uint32_t eps5; + uint32_t eps6; + uint32_t eps7; + uint32_t depc; + uint32_t intenable; + uint32_t interrupt; + uint32_t sar; + uint32_t stack; +} __packed; + +#endif diff --git a/sound/soc/sof/xtensa/Kconfig b/sound/soc/sof/xtensa/Kconfig new file mode 100644 index 000000000000..8a9343b85146 --- /dev/null +++ b/sound/soc/sof/xtensa/Kconfig @@ -0,0 +1,2 @@ +config SND_SOC_SOF_XTENSA + tristate diff --git a/sound/soc/sof/xtensa/Makefile b/sound/soc/sof/xtensa/Makefile new file mode 100644 index 000000000000..cc89c7472a38 --- /dev/null +++ b/sound/soc/sof/xtensa/Makefile @@ -0,0 +1,5 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) + +snd-sof-xtensa-dsp-objs := core.o + +obj-$(CONFIG_SND_SOC_SOF_XTENSA) += snd-sof-xtensa-dsp.o diff --git a/sound/soc/sof/xtensa/core.c b/sound/soc/sof/xtensa/core.c new file mode 100644 index 000000000000..c3ad23a85b99 --- /dev/null +++ b/sound/soc/sof/xtensa/core.c @@ -0,0 +1,138 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Pan Xiuli +// + +#include +#include +#include +#include "../sof-priv.h" + +struct xtensa_exception_cause { + u32 id; + const char *msg; + const char *description; +}; + +/* + * From 4.4.1.5 table 4-64 Exception Causes of Xtensa + * Instruction Set Architecture (ISA) Reference Manual + */ +static const struct xtensa_exception_cause xtensa_exception_causes[] = { + {0, "IllegalInstructionCause", "Illegal instruction"}, + {1, "SyscallCause", "SYSCALL instruction"}, + {2, "InstructionFetchErrorCause", + "Processor internal physical address or data error during instruction fetch"}, + {3, "LoadStoreErrorCause", + "Processor internal physical address or data error during load or store"}, + {4, "Level1InterruptCause", + "Level-1 interrupt as indicated by set level-1 bits in the INTERRUPT register"}, + {5, "AllocaCause", + "MOVSP instruction, if caller’s registers are not in the register file"}, + {6, "IntegerDivideByZeroCause", + "QUOS, QUOU, REMS, or REMU divisor operand is zero"}, + {8, "PrivilegedCause", + "Attempt to execute a privileged operation when CRING ? 0"}, + {9, "LoadStoreAlignmentCause", "Load or store to an unaligned address"}, + {12, "InstrPIFDataErrorCause", + "PIF data error during instruction fetch"}, + {13, "LoadStorePIFDataErrorCause", + "Synchronous PIF data error during LoadStore access"}, + {14, "InstrPIFAddrErrorCause", + "PIF address error during instruction fetch"}, + {15, "LoadStorePIFAddrErrorCause", + "Synchronous PIF address error during LoadStore access"}, + {16, "InstTLBMissCause", "Error during Instruction TLB refill"}, + {17, "InstTLBMultiHitCause", + "Multiple instruction TLB entries matched"}, + {18, "InstFetchPrivilegeCause", + "An instruction fetch referenced a virtual address at a ring level less than CRING"}, + {20, "InstFetchProhibitedCause", + "An instruction fetch referenced a page mapped with an attribute that does not permit instruction fetch"}, + {24, "LoadStoreTLBMissCause", + "Error during TLB refill for a load or store"}, + {25, "LoadStoreTLBMultiHitCause", + "Multiple TLB entries matched for a load or store"}, + {26, "LoadStorePrivilegeCause", + "A load or store referenced a virtual address at a ring level less than CRING"}, + {28, "LoadProhibitedCause", + "A load referenced a page mapped with an attribute that does not permit loads"}, + {32, "Coprocessor0Disabled", + "Coprocessor 0 instruction when cp0 disabled"}, + {33, "Coprocessor1Disabled", + "Coprocessor 1 instruction when cp1 disabled"}, + {34, "Coprocessor2Disabled", + "Coprocessor 2 instruction when cp2 disabled"}, + {35, "Coprocessor3Disabled", + "Coprocessor 3 instruction when cp3 disabled"}, + {36, "Coprocessor4Disabled", + "Coprocessor 4 instruction when cp4 disabled"}, + {37, "Coprocessor5Disabled", + "Coprocessor 5 instruction when cp5 disabled"}, + {38, "Coprocessor6Disabled", + "Coprocessor 6 instruction when cp6 disabled"}, + {39, "Coprocessor7Disabled", + "Coprocessor 7 instruction when cp7 disabled"}, +}; + +/* only need xtensa atm */ +static void xtensa_dsp_oops(struct snd_sof_dev *sdev, void *oops) +{ + struct sof_ipc_dsp_oops_xtensa *xoops = oops; + int i; + + dev_err(sdev->dev, "error: DSP Firmware Oops\n"); + for (i = 0; i < ARRAY_SIZE(xtensa_exception_causes); i++) { + if (xtensa_exception_causes[i].id == xoops->exccause) { + dev_err(sdev->dev, "error: Exception Cause: %s, %s\n", + xtensa_exception_causes[i].msg, + xtensa_exception_causes[i].description); + } + } + dev_err(sdev->dev, "EXCCAUSE 0x%8.8x EXCVADDR 0x%8.8x PS 0x%8.8x SAR 0x%8.8x\n", + xoops->exccause, xoops->excvaddr, xoops->ps, xoops->sar); + dev_err(sdev->dev, "EPC1 0x%8.8x EPC2 0x%8.8x EPC3 0x%8.8x EPC4 0x%8.8x", + xoops->epc1, xoops->epc2, xoops->epc3, xoops->epc4); + dev_err(sdev->dev, "EPC5 0x%8.8x EPC6 0x%8.8x EPC7 0x%8.8x DEPC 0x%8.8x", + xoops->epc5, xoops->epc6, xoops->epc7, xoops->depc); + dev_err(sdev->dev, "EPS2 0x%8.8x EPS3 0x%8.8x EPS4 0x%8.8x EPS5 0x%8.8x", + xoops->eps2, xoops->eps3, xoops->eps4, xoops->eps5); + dev_err(sdev->dev, "EPS6 0x%8.8x EPS7 0x%8.8x INTENABL 0x%8.8x INTERRU 0x%8.8x", + xoops->eps6, xoops->eps7, xoops->intenable, xoops->interrupt); +} + +static void xtensa_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack, + u32 stack_words) +{ + struct sof_ipc_dsp_oops_xtensa *xoops = oops; + u32 stack_ptr = xoops->stack; + /* 4 * 8chars + 3 ws + 1 terminating NUL */ + unsigned char buf[4 * 8 + 3 + 1]; + int i; + + dev_err(sdev->dev, "stack dump from 0x%8.8x\n", stack_ptr); + + /* + * example output: + * 0x0049fbb0: 8000f2d0 0049fc00 6f6c6c61 00632e63 + */ + for (i = 0; i < stack_words; i += 4) { + hex_dump_to_buffer(stack + i * 4, 16, 16, 4, + buf, sizeof(buf), false); + dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i, buf); + } +} + +const struct sof_arch_ops sof_xtensa_arch_ops = { + .dsp_oops = xtensa_dsp_oops, + .dsp_stack = xtensa_stack, +}; +EXPORT_SYMBOL(sof_xtensa_arch_ops); + +MODULE_DESCRIPTION("SOF Xtensa DSP support"); +MODULE_LICENSE("Dual BSD/GPL"); -- cgit v1.2.3-59-g8ed1b From 3bafac46065eb7ea6f3e3fcbdb8b70b66f6c71b4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 12 Apr 2019 11:05:19 -0500 Subject: ASoC: SOF: Add utils Helpers to set-up back-ends, create platform devices and common IO/block read/write operations Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/utils.c | 112 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 112 insertions(+) create mode 100644 sound/soc/sof/utils.c (limited to 'sound') diff --git a/sound/soc/sof/utils.c b/sound/soc/sof/utils.c new file mode 100644 index 000000000000..2ac4c3da0320 --- /dev/null +++ b/sound/soc/sof/utils.c @@ -0,0 +1,112 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Keyon Jie +// + +#include +#include +#include +#include +#include "sof-priv.h" + +/* + * Register IO + * + * The sof_io_xyz() wrappers are typically referenced in snd_sof_dsp_ops + * structures and cannot be inlined. + */ + +void sof_io_write(struct snd_sof_dev *sdev, void __iomem *addr, u32 value) +{ + writel(value, addr); +} +EXPORT_SYMBOL(sof_io_write); + +u32 sof_io_read(struct snd_sof_dev *sdev, void __iomem *addr) +{ + return readl(addr); +} +EXPORT_SYMBOL(sof_io_read); + +void sof_io_write64(struct snd_sof_dev *sdev, void __iomem *addr, u64 value) +{ + writeq(value, addr); +} +EXPORT_SYMBOL(sof_io_write64); + +u64 sof_io_read64(struct snd_sof_dev *sdev, void __iomem *addr) +{ + return readq(addr); +} +EXPORT_SYMBOL(sof_io_read64); + +/* + * IPC Mailbox IO + */ + +void sof_mailbox_write(struct snd_sof_dev *sdev, u32 offset, + void *message, size_t bytes) +{ + void __iomem *dest = sdev->bar[sdev->mailbox_bar] + offset; + + memcpy_toio(dest, message, bytes); +} +EXPORT_SYMBOL(sof_mailbox_write); + +void sof_mailbox_read(struct snd_sof_dev *sdev, u32 offset, + void *message, size_t bytes) +{ + void __iomem *src = sdev->bar[sdev->mailbox_bar] + offset; + + memcpy_fromio(message, src, bytes); +} +EXPORT_SYMBOL(sof_mailbox_read); + +/* + * Memory copy. + */ + +void sof_block_write(struct snd_sof_dev *sdev, u32 bar, u32 offset, void *src, + size_t size) +{ + void __iomem *dest = sdev->bar[bar] + offset; + const u8 *src_byte = src; + u32 affected_mask; + u32 tmp; + int m, n; + + m = size / 4; + n = size % 4; + + /* __iowrite32_copy use 32bit size values so divide by 4 */ + __iowrite32_copy(dest, src, m); + + if (n) { + affected_mask = (1 << (8 * n)) - 1; + + /* first read the 32bit data of dest, then change affected + * bytes, and write back to dest. For unaffected bytes, it + * should not be changed + */ + tmp = ioread32(dest + m * 4); + tmp &= ~affected_mask; + + tmp |= *(u32 *)(src_byte + m * 4) & affected_mask; + iowrite32(tmp, dest + m * 4); + } +} +EXPORT_SYMBOL(sof_block_write); + +void sof_block_read(struct snd_sof_dev *sdev, u32 bar, u32 offset, void *dest, + size_t size) +{ + void __iomem *src = sdev->bar[bar] + offset; + + memcpy_fromio(dest, src, size); +} +EXPORT_SYMBOL(sof_block_read); -- cgit v1.2.3-59-g8ed1b From 9e42c5ca4a276a668b11116704f5f0d66ab80608 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:44 -0500 Subject: ASoC: SOF: Intel: Add BYT, CHT and BSW DSP HW support. Add support for the audio DSP hardware found on Intel Baytrail, Cherrytrail and Braswell based devices. Signed-off-by: Rander Wang Signed-off-by: Pan Xiuli Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/byt.c | 874 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/shim.h | 183 ++++++++++ 2 files changed, 1057 insertions(+) create mode 100644 sound/soc/sof/intel/byt.c create mode 100644 sound/soc/sof/intel/shim.h (limited to 'sound') diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c new file mode 100644 index 000000000000..7bf9143d3106 --- /dev/null +++ b/sound/soc/sof/intel/byt.c @@ -0,0 +1,874 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +/* + * Hardware interface for audio DSP on Baytrail, Braswell and Cherrytrail. + */ + +#include +#include +#include +#include "../ops.h" +#include "shim.h" + +/* DSP memories */ +#define IRAM_OFFSET 0x0C0000 +#define IRAM_SIZE (80 * 1024) +#define DRAM_OFFSET 0x100000 +#define DRAM_SIZE (160 * 1024) +#define SHIM_OFFSET 0x140000 +#define SHIM_SIZE 0x100 +#define MBOX_OFFSET 0x144000 +#define MBOX_SIZE 0x1000 +#define EXCEPT_OFFSET 0x800 + +/* DSP peripherals */ +#define DMAC0_OFFSET 0x098000 +#define DMAC1_OFFSET 0x09c000 +#define DMAC2_OFFSET 0x094000 +#define DMAC_SIZE 0x420 +#define SSP0_OFFSET 0x0a0000 +#define SSP1_OFFSET 0x0a1000 +#define SSP2_OFFSET 0x0a2000 +#define SSP3_OFFSET 0x0a4000 +#define SSP4_OFFSET 0x0a5000 +#define SSP5_OFFSET 0x0a6000 +#define SSP_SIZE 0x100 + +#define BYT_STACK_DUMP_SIZE 32 + +#define BYT_PCI_BAR_SIZE 0x200000 + +#define BYT_PANIC_OFFSET(x) (((x) & GENMASK_ULL(47, 32)) >> 32) + +/* + * Debug + */ + +#define MBOX_DUMP_SIZE 0x30 + +/* BARs */ +#define BYT_DSP_BAR 0 +#define BYT_PCI_BAR 1 +#define BYT_IMR_BAR 2 + +static const struct snd_sof_debugfs_map byt_debugfs[] = { + {"dmac0", BYT_DSP_BAR, DMAC0_OFFSET, DMAC_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"dmac1", BYT_DSP_BAR, DMAC1_OFFSET, DMAC_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp0", BYT_DSP_BAR, SSP0_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp1", BYT_DSP_BAR, SSP1_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp2", BYT_DSP_BAR, SSP2_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"iram", BYT_DSP_BAR, IRAM_OFFSET, IRAM_SIZE, + SOF_DEBUGFS_ACCESS_D0_ONLY}, + {"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE, + SOF_DEBUGFS_ACCESS_D0_ONLY}, + {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, +}; + +static const struct snd_sof_debugfs_map cht_debugfs[] = { + {"dmac0", BYT_DSP_BAR, DMAC0_OFFSET, DMAC_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"dmac1", BYT_DSP_BAR, DMAC1_OFFSET, DMAC_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"dmac2", BYT_DSP_BAR, DMAC2_OFFSET, DMAC_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp0", BYT_DSP_BAR, SSP0_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp1", BYT_DSP_BAR, SSP1_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp2", BYT_DSP_BAR, SSP2_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp3", BYT_DSP_BAR, SSP3_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp4", BYT_DSP_BAR, SSP4_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp5", BYT_DSP_BAR, SSP5_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"iram", BYT_DSP_BAR, IRAM_OFFSET, IRAM_SIZE, + SOF_DEBUGFS_ACCESS_D0_ONLY}, + {"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE, + SOF_DEBUGFS_ACCESS_D0_ONLY}, + {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, +}; + +static void byt_host_done(struct snd_sof_dev *sdev); +static void byt_dsp_done(struct snd_sof_dev *sdev); +static void byt_get_reply(struct snd_sof_dev *sdev); + +/* + * IPC Firmware ready. + */ +static void byt_get_windows(struct snd_sof_dev *sdev) +{ + struct sof_ipc_window_elem *elem; + u32 outbox_offset = 0; + u32 stream_offset = 0; + u32 inbox_offset = 0; + u32 outbox_size = 0; + u32 stream_size = 0; + u32 inbox_size = 0; + int i; + + if (!sdev->info_window) { + dev_err(sdev->dev, "error: have no window info\n"); + return; + } + + for (i = 0; i < sdev->info_window->num_windows; i++) { + elem = &sdev->info_window->window[i]; + + switch (elem->type) { + case SOF_IPC_REGION_UPBOX: + inbox_offset = elem->offset + MBOX_OFFSET; + inbox_size = elem->size; + snd_sof_debugfs_io_item(sdev, + sdev->bar[BYT_DSP_BAR] + + inbox_offset, + elem->size, "inbox", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_DOWNBOX: + outbox_offset = elem->offset + MBOX_OFFSET; + outbox_size = elem->size; + snd_sof_debugfs_io_item(sdev, + sdev->bar[BYT_DSP_BAR] + + outbox_offset, + elem->size, "outbox", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_TRACE: + snd_sof_debugfs_io_item(sdev, + sdev->bar[BYT_DSP_BAR] + + elem->offset + + MBOX_OFFSET, + elem->size, "etrace", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_DEBUG: + snd_sof_debugfs_io_item(sdev, + sdev->bar[BYT_DSP_BAR] + + elem->offset + + MBOX_OFFSET, + elem->size, "debug", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_STREAM: + stream_offset = elem->offset + MBOX_OFFSET; + stream_size = elem->size; + snd_sof_debugfs_io_item(sdev, + sdev->bar[BYT_DSP_BAR] + + stream_offset, + elem->size, "stream", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_REGS: + snd_sof_debugfs_io_item(sdev, + sdev->bar[BYT_DSP_BAR] + + elem->offset + + MBOX_OFFSET, + elem->size, "regs", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_EXCEPTION: + sdev->dsp_oops_offset = elem->offset + MBOX_OFFSET; + snd_sof_debugfs_io_item(sdev, + sdev->bar[BYT_DSP_BAR] + + elem->offset + + MBOX_OFFSET, + elem->size, "exception", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + default: + dev_err(sdev->dev, "error: get illegal window info\n"); + return; + } + } + + if (outbox_size == 0 || inbox_size == 0) { + dev_err(sdev->dev, "error: get illegal mailbox window\n"); + return; + } + + snd_sof_dsp_mailbox_init(sdev, inbox_offset, inbox_size, + outbox_offset, outbox_size); + sdev->stream_box.offset = stream_offset; + sdev->stream_box.size = stream_size; + + dev_dbg(sdev->dev, " mailbox upstream 0x%x - size 0x%x\n", + inbox_offset, inbox_size); + dev_dbg(sdev->dev, " mailbox downstream 0x%x - size 0x%x\n", + outbox_offset, outbox_size); + dev_dbg(sdev->dev, " stream region 0x%x - size 0x%x\n", + stream_offset, stream_size); +} + +/* check for ABI compatibility and create memory windows on first boot */ +static int byt_fw_ready(struct snd_sof_dev *sdev, u32 msg_id) +{ + struct sof_ipc_fw_ready *fw_ready = &sdev->fw_ready; + u32 offset; + int ret; + + /* mailbox must be on 4k boundary */ + offset = MBOX_OFFSET; + + dev_dbg(sdev->dev, "ipc: DSP is ready 0x%8.8x offset 0x%x\n", + msg_id, offset); + + /* no need to re-check version/ABI for subsequent boots */ + if (!sdev->first_boot) + return 0; + + /* copy data from the DSP FW ready offset */ + sof_block_read(sdev, sdev->mmio_bar, offset, fw_ready, + sizeof(*fw_ready)); + + snd_sof_dsp_mailbox_init(sdev, fw_ready->dspbox_offset, + fw_ready->dspbox_size, + fw_ready->hostbox_offset, + fw_ready->hostbox_size); + + /* make sure ABI version is compatible */ + ret = snd_sof_ipc_valid(sdev); + if (ret < 0) + return ret; + + /* now check for extended data */ + snd_sof_fw_parse_ext_data(sdev, sdev->mmio_bar, MBOX_OFFSET + + sizeof(struct sof_ipc_fw_ready)); + + byt_get_windows(sdev); + + return 0; +} + +/* + * Debug + */ + +static void byt_get_registers(struct snd_sof_dev *sdev, + struct sof_ipc_dsp_oops_xtensa *xoops, + struct sof_ipc_panic_info *panic_info, + u32 *stack, size_t stack_words) +{ + /* first read regsisters */ + sof_mailbox_read(sdev, sdev->dsp_oops_offset, xoops, sizeof(*xoops)); + + /* then get panic info */ + sof_mailbox_read(sdev, sdev->dsp_oops_offset + sizeof(*xoops), + panic_info, sizeof(*panic_info)); + + /* then get the stack */ + sof_mailbox_read(sdev, sdev->dsp_oops_offset + sizeof(*xoops) + + sizeof(*panic_info), stack, + stack_words * sizeof(u32)); +} + +static void byt_dump(struct snd_sof_dev *sdev, u32 flags) +{ + struct sof_ipc_dsp_oops_xtensa xoops; + struct sof_ipc_panic_info panic_info; + u32 stack[BYT_STACK_DUMP_SIZE]; + u32 status, panic; + + /* now try generic SOF status messages */ + status = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCD); + panic = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCX); + byt_get_registers(sdev, &xoops, &panic_info, stack, + BYT_STACK_DUMP_SIZE); + snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack, + BYT_STACK_DUMP_SIZE); +} + +/* + * IPC Doorbell IRQ handler and thread. + */ + +static irqreturn_t byt_irq_handler(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + u64 isr; + int ret = IRQ_NONE; + + /* Interrupt arrived, check src */ + isr = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_ISRX); + if (isr & (SHIM_ISRX_DONE | SHIM_ISRX_BUSY)) + ret = IRQ_WAKE_THREAD; + + return ret; +} + +static irqreturn_t byt_irq_thread(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + u64 ipcx, ipcd; + u64 imrx; + + imrx = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IMRX); + ipcx = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IPCX); + + /* reply message from DSP */ + if (ipcx & SHIM_BYT_IPCX_DONE && + !(imrx & SHIM_IMRX_DONE)) { + /* Mask Done interrupt before first */ + snd_sof_dsp_update_bits64_unlocked(sdev, BYT_DSP_BAR, + SHIM_IMRX, + SHIM_IMRX_DONE, + SHIM_IMRX_DONE); + /* + * handle immediate reply from DSP core. If the msg is + * found, set done bit in cmd_done which is called at the + * end of message processing function, else set it here + * because the done bit can't be set in cmd_done function + * which is triggered by msg + */ + byt_get_reply(sdev); + snd_sof_ipc_reply(sdev, ipcx); + + byt_dsp_done(sdev); + } + + /* new message from DSP */ + ipcd = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IPCD); + if (ipcd & SHIM_BYT_IPCD_BUSY && + !(imrx & SHIM_IMRX_BUSY)) { + /* Mask Busy interrupt before return */ + snd_sof_dsp_update_bits64_unlocked(sdev, BYT_DSP_BAR, + SHIM_IMRX, + SHIM_IMRX_BUSY, + SHIM_IMRX_BUSY); + + /* Handle messages from DSP Core */ + if ((ipcd & SOF_IPC_PANIC_MAGIC_MASK) == SOF_IPC_PANIC_MAGIC) { + snd_sof_dsp_panic(sdev, BYT_PANIC_OFFSET(ipcd) + + MBOX_OFFSET); + } else { + snd_sof_ipc_msgs_rx(sdev); + } + + byt_host_done(sdev); + } + + return IRQ_HANDLED; +} + +static int byt_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) +{ + u64 cmd = msg->header; + + /* send the message */ + sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data, + msg->msg_size); + snd_sof_dsp_write64(sdev, BYT_DSP_BAR, SHIM_IPCX, + cmd | SHIM_BYT_IPCX_BUSY); + + return 0; +} + +static void byt_get_reply(struct snd_sof_dev *sdev) +{ + struct snd_sof_ipc_msg *msg = sdev->msg; + struct sof_ipc_reply reply; + unsigned long flags; + int ret = 0; + + /* + * Sometimes, there is unexpected reply ipc arriving. The reply + * ipc belongs to none of the ipcs sent from driver. + * In this case, the driver must ignore the ipc. + */ + if (!msg) { + dev_warn(sdev->dev, "unexpected ipc interrupt raised!\n"); + return; + } + + /* get reply */ + sof_mailbox_read(sdev, sdev->host_box.offset, &reply, sizeof(reply)); + + spin_lock_irqsave(&sdev->ipc_lock, flags); + + if (reply.error < 0) { + memcpy(msg->reply_data, &reply, sizeof(reply)); + ret = reply.error; + } else { + /* reply correct size ? */ + if (reply.hdr.size != msg->reply_size) { + dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", + msg->reply_size, reply.hdr.size); + ret = -EINVAL; + } + + /* read the message */ + if (msg->reply_size > 0) + sof_mailbox_read(sdev, sdev->host_box.offset, + msg->reply_data, msg->reply_size); + } + + msg->reply_error = ret; + + spin_unlock_irqrestore(&sdev->ipc_lock, flags); +} + +static void byt_host_done(struct snd_sof_dev *sdev) +{ + /* clear BUSY bit and set DONE bit - accept new messages */ + snd_sof_dsp_update_bits64_unlocked(sdev, BYT_DSP_BAR, SHIM_IPCD, + SHIM_BYT_IPCD_BUSY | + SHIM_BYT_IPCD_DONE, + SHIM_BYT_IPCD_DONE); + + /* unmask busy interrupt */ + snd_sof_dsp_update_bits64_unlocked(sdev, BYT_DSP_BAR, SHIM_IMRX, + SHIM_IMRX_BUSY, 0); +} + +static void byt_dsp_done(struct snd_sof_dev *sdev) +{ + /* clear DONE bit - tell DSP we have completed */ + snd_sof_dsp_update_bits64_unlocked(sdev, BYT_DSP_BAR, SHIM_IPCX, + SHIM_BYT_IPCX_DONE, 0); + + /* unmask Done interrupt */ + snd_sof_dsp_update_bits64_unlocked(sdev, BYT_DSP_BAR, SHIM_IMRX, + SHIM_IMRX_DONE, 0); +} + +/* + * DSP control. + */ + +static int byt_run(struct snd_sof_dev *sdev) +{ + int tries = 10; + + /* release stall and wait to unstall */ + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_CSR, + SHIM_BYT_CSR_STALL, 0x0); + while (tries--) { + if (!(snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_CSR) & + SHIM_BYT_CSR_PWAITMODE)) + break; + msleep(100); + } + if (tries < 0) { + dev_err(sdev->dev, "error: unable to run DSP firmware\n"); + byt_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX); + return -ENODEV; + } + + /* return init core mask */ + return 1; +} + +static int byt_reset(struct snd_sof_dev *sdev) +{ + /* put DSP into reset, set reset vector and stall */ + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_CSR, + SHIM_BYT_CSR_RST | SHIM_BYT_CSR_VECTOR_SEL | + SHIM_BYT_CSR_STALL, + SHIM_BYT_CSR_RST | SHIM_BYT_CSR_VECTOR_SEL | + SHIM_BYT_CSR_STALL); + + usleep_range(10, 15); + + /* take DSP out of reset and keep stalled for FW loading */ + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_CSR, + SHIM_BYT_CSR_RST, 0); + + return 0; +} + +/* Baytrail DAIs */ +static struct snd_soc_dai_driver byt_dai[] = { +{ + .name = "ssp0-port", +}, +{ + .name = "ssp1-port", +}, +{ + .name = "ssp2-port", +}, +{ + .name = "ssp3-port", +}, +{ + .name = "ssp4-port", +}, +{ + .name = "ssp5-port", +}, +}; + +/* + * Probe and remove. + */ + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) + +static int tangier_pci_probe(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *pdata = sdev->pdata; + const struct sof_dev_desc *desc = pdata->desc; + struct pci_dev *pci = to_pci_dev(sdev->dev); + u32 base, size; + int ret; + + /* DSP DMA can only access low 31 bits of host memory */ + ret = dma_coerce_mask_and_coherent(&pci->dev, DMA_BIT_MASK(31)); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to set DMA mask %d\n", ret); + return ret; + } + + /* LPE base */ + base = pci_resource_start(pci, desc->resindex_lpe_base) - IRAM_OFFSET; + size = BYT_PCI_BAR_SIZE; + + dev_dbg(sdev->dev, "LPE PHY base at 0x%x size 0x%x", base, size); + sdev->bar[BYT_DSP_BAR] = devm_ioremap(sdev->dev, base, size); + if (!sdev->bar[BYT_DSP_BAR]) { + dev_err(sdev->dev, "error: failed to ioremap LPE base 0x%x size 0x%x\n", + base, size); + return -ENODEV; + } + dev_dbg(sdev->dev, "LPE VADDR %p\n", sdev->bar[BYT_DSP_BAR]); + + /* IMR base - optional */ + if (desc->resindex_imr_base == -1) + goto irq; + + base = pci_resource_start(pci, desc->resindex_imr_base); + size = pci_resource_len(pci, desc->resindex_imr_base); + + /* some BIOSes don't map IMR */ + if (base == 0x55aa55aa || base == 0x0) { + dev_info(sdev->dev, "IMR not set by BIOS. Ignoring\n"); + goto irq; + } + + dev_dbg(sdev->dev, "IMR base at 0x%x size 0x%x", base, size); + sdev->bar[BYT_IMR_BAR] = devm_ioremap(sdev->dev, base, size); + if (!sdev->bar[BYT_IMR_BAR]) { + dev_err(sdev->dev, "error: failed to ioremap IMR base 0x%x size 0x%x\n", + base, size); + return -ENODEV; + } + dev_dbg(sdev->dev, "IMR VADDR %p\n", sdev->bar[BYT_IMR_BAR]); + +irq: + /* register our IRQ */ + sdev->ipc_irq = pci->irq; + dev_dbg(sdev->dev, "using IRQ %d\n", sdev->ipc_irq); + ret = devm_request_threaded_irq(sdev->dev, sdev->ipc_irq, + byt_irq_handler, byt_irq_thread, + 0, "AudioDSP", sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to register IRQ %d\n", + sdev->ipc_irq); + return ret; + } + + /* enable Interrupt from both sides */ + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_IMRX, 0x3, 0x0); + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_IMRD, 0x3, 0x0); + + /* set default mailbox offset for FW ready message */ + sdev->dsp_box.offset = MBOX_OFFSET; + + return ret; +} + +const struct snd_sof_dsp_ops sof_tng_ops = { + /* device init */ + .probe = tangier_pci_probe, + + /* DSP core boot / reset */ + .run = byt_run, + .reset = byt_reset, + + /* Register IO */ + .write = sof_io_write, + .read = sof_io_read, + .write64 = sof_io_write64, + .read64 = sof_io_read64, + + /* Block IO */ + .block_read = sof_block_read, + .block_write = sof_block_write, + + /* doorbell */ + .irq_handler = byt_irq_handler, + .irq_thread = byt_irq_thread, + + /* ipc */ + .send_msg = byt_send_msg, + .fw_ready = byt_fw_ready, + + .ipc_msg_data = intel_ipc_msg_data, + .ipc_pcm_params = intel_ipc_pcm_params, + + /* debug */ + .debug_map = byt_debugfs, + .debug_map_count = ARRAY_SIZE(byt_debugfs), + .dbg_dump = byt_dump, + + /* stream callbacks */ + .pcm_open = intel_pcm_open, + .pcm_close = intel_pcm_close, + + /* module loading */ + .load_module = snd_sof_parse_module_memcpy, + + /*Firmware loading */ + .load_firmware = snd_sof_load_firmware_memcpy, + + /* DAI drivers */ + .drv = byt_dai, + .num_drv = 3, /* we have only 3 SSPs on byt*/ +}; +EXPORT_SYMBOL(sof_tng_ops); + +const struct sof_intel_dsp_desc tng_chip_info = { + .cores_num = 1, + .cores_mask = 1, +}; +EXPORT_SYMBOL(tng_chip_info); + +#endif /* CONFIG_SND_SOC_SOF_MERRIFIELD */ + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) + +static int byt_acpi_probe(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *pdata = sdev->pdata; + const struct sof_dev_desc *desc = pdata->desc; + struct platform_device *pdev = + container_of(sdev->dev, struct platform_device, dev); + struct resource *mmio; + u32 base, size; + int ret; + + /* DSP DMA can only access low 31 bits of host memory */ + ret = dma_coerce_mask_and_coherent(sdev->dev, DMA_BIT_MASK(31)); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to set DMA mask %d\n", ret); + return ret; + } + + /* LPE base */ + mmio = platform_get_resource(pdev, IORESOURCE_MEM, + desc->resindex_lpe_base); + if (mmio) { + base = mmio->start; + size = resource_size(mmio); + } else { + dev_err(sdev->dev, "error: failed to get LPE base at idx %d\n", + desc->resindex_lpe_base); + return -EINVAL; + } + + dev_dbg(sdev->dev, "LPE PHY base at 0x%x size 0x%x", base, size); + sdev->bar[BYT_DSP_BAR] = devm_ioremap(sdev->dev, base, size); + if (!sdev->bar[BYT_DSP_BAR]) { + dev_err(sdev->dev, "error: failed to ioremap LPE base 0x%x size 0x%x\n", + base, size); + return -ENODEV; + } + dev_dbg(sdev->dev, "LPE VADDR %p\n", sdev->bar[BYT_DSP_BAR]); + + /* TODO: add offsets */ + sdev->mmio_bar = BYT_DSP_BAR; + sdev->mailbox_bar = BYT_DSP_BAR; + + /* IMR base - optional */ + if (desc->resindex_imr_base == -1) + goto irq; + + mmio = platform_get_resource(pdev, IORESOURCE_MEM, + desc->resindex_imr_base); + if (mmio) { + base = mmio->start; + size = resource_size(mmio); + } else { + dev_err(sdev->dev, "error: failed to get IMR base at idx %d\n", + desc->resindex_imr_base); + return -ENODEV; + } + + /* some BIOSes don't map IMR */ + if (base == 0x55aa55aa || base == 0x0) { + dev_info(sdev->dev, "IMR not set by BIOS. Ignoring\n"); + goto irq; + } + + dev_dbg(sdev->dev, "IMR base at 0x%x size 0x%x", base, size); + sdev->bar[BYT_IMR_BAR] = devm_ioremap(sdev->dev, base, size); + if (!sdev->bar[BYT_IMR_BAR]) { + dev_err(sdev->dev, "error: failed to ioremap IMR base 0x%x size 0x%x\n", + base, size); + return -ENODEV; + } + dev_dbg(sdev->dev, "IMR VADDR %p\n", sdev->bar[BYT_IMR_BAR]); + +irq: + /* register our IRQ */ + sdev->ipc_irq = platform_get_irq(pdev, desc->irqindex_host_ipc); + if (sdev->ipc_irq < 0) { + dev_err(sdev->dev, "error: failed to get IRQ at index %d\n", + desc->irqindex_host_ipc); + return sdev->ipc_irq; + } + + dev_dbg(sdev->dev, "using IRQ %d\n", sdev->ipc_irq); + ret = devm_request_threaded_irq(sdev->dev, sdev->ipc_irq, + byt_irq_handler, byt_irq_thread, + IRQF_SHARED, "AudioDSP", sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to register IRQ %d\n", + sdev->ipc_irq); + return ret; + } + + /* enable Interrupt from both sides */ + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_IMRX, 0x3, 0x0); + snd_sof_dsp_update_bits64(sdev, BYT_DSP_BAR, SHIM_IMRD, 0x3, 0x0); + + /* set default mailbox offset for FW ready message */ + sdev->dsp_box.offset = MBOX_OFFSET; + + return ret; +} + +/* baytrail ops */ +const struct snd_sof_dsp_ops sof_byt_ops = { + /* device init */ + .probe = byt_acpi_probe, + + /* DSP core boot / reset */ + .run = byt_run, + .reset = byt_reset, + + /* Register IO */ + .write = sof_io_write, + .read = sof_io_read, + .write64 = sof_io_write64, + .read64 = sof_io_read64, + + /* Block IO */ + .block_read = sof_block_read, + .block_write = sof_block_write, + + /* doorbell */ + .irq_handler = byt_irq_handler, + .irq_thread = byt_irq_thread, + + /* ipc */ + .send_msg = byt_send_msg, + .fw_ready = byt_fw_ready, + + .ipc_msg_data = intel_ipc_msg_data, + .ipc_pcm_params = intel_ipc_pcm_params, + + /* debug */ + .debug_map = byt_debugfs, + .debug_map_count = ARRAY_SIZE(byt_debugfs), + .dbg_dump = byt_dump, + + /* stream callbacks */ + .pcm_open = intel_pcm_open, + .pcm_close = intel_pcm_close, + + /* module loading */ + .load_module = snd_sof_parse_module_memcpy, + + /*Firmware loading */ + .load_firmware = snd_sof_load_firmware_memcpy, + + /* DAI drivers */ + .drv = byt_dai, + .num_drv = 3, /* we have only 3 SSPs on byt*/ +}; +EXPORT_SYMBOL(sof_byt_ops); + +const struct sof_intel_dsp_desc byt_chip_info = { + .cores_num = 1, + .cores_mask = 1, +}; +EXPORT_SYMBOL(byt_chip_info); + +/* cherrytrail and braswell ops */ +const struct snd_sof_dsp_ops sof_cht_ops = { + /* device init */ + .probe = byt_acpi_probe, + + /* DSP core boot / reset */ + .run = byt_run, + .reset = byt_reset, + + /* Register IO */ + .write = sof_io_write, + .read = sof_io_read, + .write64 = sof_io_write64, + .read64 = sof_io_read64, + + /* Block IO */ + .block_read = sof_block_read, + .block_write = sof_block_write, + + /* doorbell */ + .irq_handler = byt_irq_handler, + .irq_thread = byt_irq_thread, + + /* ipc */ + .send_msg = byt_send_msg, + .fw_ready = byt_fw_ready, + + .ipc_msg_data = intel_ipc_msg_data, + .ipc_pcm_params = intel_ipc_pcm_params, + + /* debug */ + .debug_map = cht_debugfs, + .debug_map_count = ARRAY_SIZE(cht_debugfs), + .dbg_dump = byt_dump, + + /* stream callbacks */ + .pcm_open = intel_pcm_open, + .pcm_close = intel_pcm_close, + + /* module loading */ + .load_module = snd_sof_parse_module_memcpy, + + /*Firmware loading */ + .load_firmware = snd_sof_load_firmware_memcpy, + + /* DAI drivers */ + .drv = byt_dai, + /* all 6 SSPs may be available for cherrytrail */ + .num_drv = ARRAY_SIZE(byt_dai), +}; +EXPORT_SYMBOL(sof_cht_ops); + +const struct sof_intel_dsp_desc cht_chip_info = { + .cores_num = 1, + .cores_mask = 1, +}; +EXPORT_SYMBOL(cht_chip_info); + +#endif /* CONFIG_SND_SOC_SOF_BAYTRAIL */ + +MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h new file mode 100644 index 000000000000..11fd77cb4e6d --- /dev/null +++ b/sound/soc/sof/intel/shim.h @@ -0,0 +1,183 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2017 Intel Corporation. All rights reserved. + * + * Author: Liam Girdwood + */ + +#ifndef __SOF_INTEL_SHIM_H +#define __SOF_INTEL_SHIM_H + +/* + * SHIM registers for BYT, BSW, CHT, HSW, BDW + */ + +#define SHIM_CSR (SHIM_OFFSET + 0x00) +#define SHIM_PISR (SHIM_OFFSET + 0x08) +#define SHIM_PIMR (SHIM_OFFSET + 0x10) +#define SHIM_ISRX (SHIM_OFFSET + 0x18) +#define SHIM_ISRD (SHIM_OFFSET + 0x20) +#define SHIM_IMRX (SHIM_OFFSET + 0x28) +#define SHIM_IMRD (SHIM_OFFSET + 0x30) +#define SHIM_IPCX (SHIM_OFFSET + 0x38) +#define SHIM_IPCD (SHIM_OFFSET + 0x40) +#define SHIM_ISRSC (SHIM_OFFSET + 0x48) +#define SHIM_ISRLPESC (SHIM_OFFSET + 0x50) +#define SHIM_IMRSC (SHIM_OFFSET + 0x58) +#define SHIM_IMRLPESC (SHIM_OFFSET + 0x60) +#define SHIM_IPCSC (SHIM_OFFSET + 0x68) +#define SHIM_IPCLPESC (SHIM_OFFSET + 0x70) +#define SHIM_CLKCTL (SHIM_OFFSET + 0x78) +#define SHIM_CSR2 (SHIM_OFFSET + 0x80) +#define SHIM_LTRC (SHIM_OFFSET + 0xE0) +#define SHIM_HMDC (SHIM_OFFSET + 0xE8) + +#define SHIM_PWMCTRL 0x1000 + +/* + * SST SHIM register bits for BYT, BSW, CHT HSW, BDW + * Register bit naming and functionaility can differ between devices. + */ + +/* CSR / CS */ +#define SHIM_CSR_RST BIT(1) +#define SHIM_CSR_SBCS0 BIT(2) +#define SHIM_CSR_SBCS1 BIT(3) +#define SHIM_CSR_DCS(x) ((x) << 4) +#define SHIM_CSR_DCS_MASK (0x7 << 4) +#define SHIM_CSR_STALL BIT(10) +#define SHIM_CSR_S0IOCS BIT(21) +#define SHIM_CSR_S1IOCS BIT(23) +#define SHIM_CSR_LPCS BIT(31) +#define SHIM_CSR_24MHZ_LPCS \ + (SHIM_CSR_SBCS0 | SHIM_CSR_SBCS1 | SHIM_CSR_LPCS) +#define SHIM_CSR_24MHZ_NO_LPCS (SHIM_CSR_SBCS0 | SHIM_CSR_SBCS1) +#define SHIM_BYT_CSR_RST BIT(0) +#define SHIM_BYT_CSR_VECTOR_SEL BIT(1) +#define SHIM_BYT_CSR_STALL BIT(2) +#define SHIM_BYT_CSR_PWAITMODE BIT(3) + +/* ISRX / ISC */ +#define SHIM_ISRX_BUSY BIT(1) +#define SHIM_ISRX_DONE BIT(0) +#define SHIM_BYT_ISRX_REQUEST BIT(1) + +/* ISRD / ISD */ +#define SHIM_ISRD_BUSY BIT(1) +#define SHIM_ISRD_DONE BIT(0) + +/* IMRX / IMC */ +#define SHIM_IMRX_BUSY BIT(1) +#define SHIM_IMRX_DONE BIT(0) +#define SHIM_BYT_IMRX_REQUEST BIT(1) + +/* IMRD / IMD */ +#define SHIM_IMRD_DONE BIT(0) +#define SHIM_IMRD_BUSY BIT(1) +#define SHIM_IMRD_SSP0 BIT(16) +#define SHIM_IMRD_DMAC0 BIT(21) +#define SHIM_IMRD_DMAC1 BIT(22) +#define SHIM_IMRD_DMAC (SHIM_IMRD_DMAC0 | SHIM_IMRD_DMAC1) + +/* IPCX / IPCC */ +#define SHIM_IPCX_DONE BIT(30) +#define SHIM_IPCX_BUSY BIT(31) +#define SHIM_BYT_IPCX_DONE BIT_ULL(62) +#define SHIM_BYT_IPCX_BUSY BIT_ULL(63) + +/* IPCD */ +#define SHIM_IPCD_DONE BIT(30) +#define SHIM_IPCD_BUSY BIT(31) +#define SHIM_BYT_IPCD_DONE BIT_ULL(62) +#define SHIM_BYT_IPCD_BUSY BIT_ULL(63) + +/* CLKCTL */ +#define SHIM_CLKCTL_SMOS(x) ((x) << 24) +#define SHIM_CLKCTL_MASK (3 << 24) +#define SHIM_CLKCTL_DCPLCG BIT(18) +#define SHIM_CLKCTL_SCOE1 BIT(17) +#define SHIM_CLKCTL_SCOE0 BIT(16) + +/* CSR2 / CS2 */ +#define SHIM_CSR2_SDFD_SSP0 BIT(1) +#define SHIM_CSR2_SDFD_SSP1 BIT(2) + +/* LTRC */ +#define SHIM_LTRC_VAL(x) ((x) << 0) + +/* HMDC */ +#define SHIM_HMDC_HDDA0(x) ((x) << 0) +#define SHIM_HMDC_HDDA1(x) ((x) << 7) +#define SHIM_HMDC_HDDA_E0_CH0 1 +#define SHIM_HMDC_HDDA_E0_CH1 2 +#define SHIM_HMDC_HDDA_E0_CH2 4 +#define SHIM_HMDC_HDDA_E0_CH3 8 +#define SHIM_HMDC_HDDA_E1_CH0 SHIM_HMDC_HDDA1(SHIM_HMDC_HDDA_E0_CH0) +#define SHIM_HMDC_HDDA_E1_CH1 SHIM_HMDC_HDDA1(SHIM_HMDC_HDDA_E0_CH1) +#define SHIM_HMDC_HDDA_E1_CH2 SHIM_HMDC_HDDA1(SHIM_HMDC_HDDA_E0_CH2) +#define SHIM_HMDC_HDDA_E1_CH3 SHIM_HMDC_HDDA1(SHIM_HMDC_HDDA_E0_CH3) +#define SHIM_HMDC_HDDA_E0_ALLCH \ + (SHIM_HMDC_HDDA_E0_CH0 | SHIM_HMDC_HDDA_E0_CH1 | \ + SHIM_HMDC_HDDA_E0_CH2 | SHIM_HMDC_HDDA_E0_CH3) +#define SHIM_HMDC_HDDA_E1_ALLCH \ + (SHIM_HMDC_HDDA_E1_CH0 | SHIM_HMDC_HDDA_E1_CH1 | \ + SHIM_HMDC_HDDA_E1_CH2 | SHIM_HMDC_HDDA_E1_CH3) + +/* Audio DSP PCI registers */ +#define PCI_VDRTCTL0 0xa0 +#define PCI_VDRTCTL1 0xa4 +#define PCI_VDRTCTL2 0xa8 +#define PCI_VDRTCTL3 0xaC + +/* VDRTCTL0 */ +#define PCI_VDRTCL0_D3PGD BIT(0) +#define PCI_VDRTCL0_D3SRAMPGD BIT(1) +#define PCI_VDRTCL0_DSRAMPGE_SHIFT 12 +#define PCI_VDRTCL0_DSRAMPGE_MASK GENMASK(PCI_VDRTCL0_DSRAMPGE_SHIFT + 19,\ + PCI_VDRTCL0_DSRAMPGE_SHIFT) +#define PCI_VDRTCL0_ISRAMPGE_SHIFT 2 +#define PCI_VDRTCL0_ISRAMPGE_MASK GENMASK(PCI_VDRTCL0_ISRAMPGE_SHIFT + 9,\ + PCI_VDRTCL0_ISRAMPGE_SHIFT) + +/* VDRTCTL2 */ +#define PCI_VDRTCL2_DCLCGE BIT(1) +#define PCI_VDRTCL2_DTCGE BIT(10) +#define PCI_VDRTCL2_APLLSE_MASK BIT(31) + +/* PMCS */ +#define PCI_PMCS 0x84 +#define PCI_PMCS_PS_MASK 0x3 + +/* DSP hardware descriptor */ +struct sof_intel_dsp_desc { + int cores_num; + int cores_mask; + int init_core_mask; /* cores available after fw boot */ + int ipc_req; + int ipc_req_mask; + int ipc_ack; + int ipc_ack_mask; + int ipc_ctl; + int rom_init_timeout; +}; + +extern const struct snd_sof_dsp_ops sof_tng_ops; +extern const struct snd_sof_dsp_ops sof_byt_ops; +extern const struct snd_sof_dsp_ops sof_cht_ops; +extern const struct snd_sof_dsp_ops sof_hsw_ops; +extern const struct snd_sof_dsp_ops sof_bdw_ops; + +extern const struct sof_intel_dsp_desc byt_chip_info; +extern const struct sof_intel_dsp_desc cht_chip_info; +extern const struct sof_intel_dsp_desc bdw_chip_info; +extern const struct sof_intel_dsp_desc hsw_chip_info; +extern const struct sof_intel_dsp_desc tng_chip_info; + +struct sof_intel_stream { + size_t posn_offset; +}; + +#endif -- cgit v1.2.3-59-g8ed1b From 458bc7296184729b5462f704fb0792b6f1676f0d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:45 -0500 Subject: ASoC: SOF: Intel: Add BDW HW DSP support Add SOF support for Intel Broadwell based devices. Signed-off-by: Rander Wang Signed-off-by: Pan Xiuli Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/bdw.c | 713 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 713 insertions(+) create mode 100644 sound/soc/sof/intel/bdw.c (limited to 'sound') diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c new file mode 100644 index 000000000000..065cb868bdfa --- /dev/null +++ b/sound/soc/sof/intel/bdw.c @@ -0,0 +1,713 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +/* + * Hardware interface for audio DSP on Broadwell + */ + +#include +#include +#include +#include "../ops.h" +#include "shim.h" + +/* BARs */ +#define BDW_DSP_BAR 0 +#define BDW_PCI_BAR 1 + +/* + * Debug + */ + +/* DSP memories for BDW */ +#define IRAM_OFFSET 0xA0000 +#define BDW_IRAM_SIZE (10 * 32 * 1024) +#define DRAM_OFFSET 0x00000 +#define BDW_DRAM_SIZE (20 * 32 * 1024) +#define SHIM_OFFSET 0xFB000 +#define SHIM_SIZE 0x100 +#define MBOX_OFFSET 0x9E000 +#define MBOX_SIZE 0x1000 +#define MBOX_DUMP_SIZE 0x30 +#define EXCEPT_OFFSET 0x800 + +/* DSP peripherals */ +#define DMAC0_OFFSET 0xFE000 +#define DMAC1_OFFSET 0xFF000 +#define DMAC_SIZE 0x420 +#define SSP0_OFFSET 0xFC000 +#define SSP1_OFFSET 0xFD000 +#define SSP_SIZE 0x100 + +#define BDW_STACK_DUMP_SIZE 32 + +#define BDW_PANIC_OFFSET(x) ((x) & 0xFFFF) + +static const struct snd_sof_debugfs_map bdw_debugfs[] = { + {"dmac0", BDW_DSP_BAR, DMAC0_OFFSET, DMAC_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"dmac1", BDW_DSP_BAR, DMAC1_OFFSET, DMAC_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp0", BDW_DSP_BAR, SSP0_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"ssp1", BDW_DSP_BAR, SSP1_OFFSET, SSP_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, + {"iram", BDW_DSP_BAR, IRAM_OFFSET, BDW_IRAM_SIZE, + SOF_DEBUGFS_ACCESS_D0_ONLY}, + {"dram", BDW_DSP_BAR, DRAM_OFFSET, BDW_DRAM_SIZE, + SOF_DEBUGFS_ACCESS_D0_ONLY}, + {"shim", BDW_DSP_BAR, SHIM_OFFSET, SHIM_SIZE, + SOF_DEBUGFS_ACCESS_ALWAYS}, +}; + +static void bdw_host_done(struct snd_sof_dev *sdev); +static void bdw_dsp_done(struct snd_sof_dev *sdev); +static void bdw_get_reply(struct snd_sof_dev *sdev); + +/* + * DSP Control. + */ + +static int bdw_run(struct snd_sof_dev *sdev) +{ + /* set opportunistic mode on engine 0,1 for all channels */ + snd_sof_dsp_update_bits(sdev, BDW_DSP_BAR, SHIM_HMDC, + SHIM_HMDC_HDDA_E0_ALLCH | + SHIM_HMDC_HDDA_E1_ALLCH, 0); + + /* set DSP to RUN */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, SHIM_CSR, + SHIM_CSR_STALL, 0x0); + + /* return init core mask */ + return 1; +} + +static int bdw_reset(struct snd_sof_dev *sdev) +{ + /* put DSP into reset and stall */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, SHIM_CSR, + SHIM_CSR_RST | SHIM_CSR_STALL, + SHIM_CSR_RST | SHIM_CSR_STALL); + + /* keep in reset for 10ms */ + mdelay(10); + + /* take DSP out of reset and keep stalled for FW loading */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, SHIM_CSR, + SHIM_CSR_RST | SHIM_CSR_STALL, + SHIM_CSR_STALL); + + return 0; +} + +static int bdw_set_dsp_D0(struct snd_sof_dev *sdev) +{ + int tries = 10; + u32 reg; + + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_PCI_BAR, PCI_VDRTCTL2, + PCI_VDRTCL2_DCLCGE | + PCI_VDRTCL2_DTCGE, 0); + + /* Disable D3PG (VDRTCTL0.D3PGD = 1) */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_PCI_BAR, PCI_VDRTCTL0, + PCI_VDRTCL0_D3PGD, PCI_VDRTCL0_D3PGD); + + /* Set D0 state */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_PCI_BAR, PCI_PMCS, + PCI_PMCS_PS_MASK, 0); + + /* check that ADSP shim is enabled */ + while (tries--) { + reg = readl(sdev->bar[BDW_PCI_BAR] + PCI_PMCS) + & PCI_PMCS_PS_MASK; + if (reg == 0) + goto finish; + + msleep(20); + } + + return -ENODEV; + +finish: + /* + * select SSP1 19.2MHz base clock, SSP clock 0, + * turn off Low Power Clock + */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, SHIM_CSR, + SHIM_CSR_S1IOCS | SHIM_CSR_SBCS1 | + SHIM_CSR_LPCS, 0x0); + + /* stall DSP core, set clk to 192/96Mhz */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, + SHIM_CSR, SHIM_CSR_STALL | + SHIM_CSR_DCS_MASK, + SHIM_CSR_STALL | + SHIM_CSR_DCS(4)); + + /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, SHIM_CLKCTL, + SHIM_CLKCTL_MASK | + SHIM_CLKCTL_DCPLCG | + SHIM_CLKCTL_SCOE0, + SHIM_CLKCTL_MASK | + SHIM_CLKCTL_DCPLCG | + SHIM_CLKCTL_SCOE0); + + /* Stall and reset core, set CSR */ + bdw_reset(sdev); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_PCI_BAR, PCI_VDRTCTL2, + PCI_VDRTCL2_DCLCGE | + PCI_VDRTCL2_DTCGE, + PCI_VDRTCL2_DCLCGE | + PCI_VDRTCL2_DTCGE); + + usleep_range(50, 55); + + /* switch on audio PLL */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_PCI_BAR, PCI_VDRTCTL2, + PCI_VDRTCL2_APLLSE_MASK, 0); + + /* + * set default power gating control, enable power gating control for + * all blocks. that is, can't be accessed, please enable each block + * before accessing. + */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_PCI_BAR, PCI_VDRTCTL0, + 0xfffffffC, 0x0); + + /* disable DMA finish function for SSP0 & SSP1 */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, SHIM_CSR2, + SHIM_CSR2_SDFD_SSP1, + SHIM_CSR2_SDFD_SSP1); + + /* set on-demond mode on engine 0,1 for all channels */ + snd_sof_dsp_update_bits(sdev, BDW_DSP_BAR, SHIM_HMDC, + SHIM_HMDC_HDDA_E0_ALLCH | + SHIM_HMDC_HDDA_E1_ALLCH, + SHIM_HMDC_HDDA_E0_ALLCH | + SHIM_HMDC_HDDA_E1_ALLCH); + + /* Enable Interrupt from both sides */ + snd_sof_dsp_update_bits(sdev, BDW_DSP_BAR, SHIM_IMRX, + (SHIM_IMRX_BUSY | SHIM_IMRX_DONE), 0x0); + snd_sof_dsp_update_bits(sdev, BDW_DSP_BAR, SHIM_IMRD, + (SHIM_IMRD_DONE | SHIM_IMRD_BUSY | + SHIM_IMRD_SSP0 | SHIM_IMRD_DMAC), 0x0); + + /* clear IPC registers */ + snd_sof_dsp_write(sdev, BDW_DSP_BAR, SHIM_IPCX, 0x0); + snd_sof_dsp_write(sdev, BDW_DSP_BAR, SHIM_IPCD, 0x0); + snd_sof_dsp_write(sdev, BDW_DSP_BAR, 0x80, 0x6); + snd_sof_dsp_write(sdev, BDW_DSP_BAR, 0xe0, 0x300a); + + return 0; +} + +static void bdw_get_registers(struct snd_sof_dev *sdev, + struct sof_ipc_dsp_oops_xtensa *xoops, + struct sof_ipc_panic_info *panic_info, + u32 *stack, size_t stack_words) +{ + /* first read regsisters */ + sof_mailbox_read(sdev, sdev->dsp_oops_offset, xoops, sizeof(*xoops)); + + /* then get panic info */ + sof_mailbox_read(sdev, sdev->dsp_oops_offset + sizeof(*xoops), + panic_info, sizeof(*panic_info)); + + /* then get the stack */ + sof_mailbox_read(sdev, sdev->dsp_oops_offset + sizeof(*xoops) + + sizeof(*panic_info), stack, + stack_words * sizeof(u32)); +} + +static void bdw_dump(struct snd_sof_dev *sdev, u32 flags) +{ + struct sof_ipc_dsp_oops_xtensa xoops; + struct sof_ipc_panic_info panic_info; + u32 stack[BDW_STACK_DUMP_SIZE]; + u32 status, panic; + + /* now try generic SOF status messages */ + status = snd_sof_dsp_read(sdev, BDW_DSP_BAR, SHIM_IPCD); + panic = snd_sof_dsp_read(sdev, BDW_DSP_BAR, SHIM_IPCX); + bdw_get_registers(sdev, &xoops, &panic_info, stack, + BDW_STACK_DUMP_SIZE); + snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack, + BDW_STACK_DUMP_SIZE); +} + +/* + * IPC Doorbell IRQ handler and thread. + */ + +static irqreturn_t bdw_irq_handler(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + u32 isr; + int ret = IRQ_NONE; + + /* Interrupt arrived, check src */ + isr = snd_sof_dsp_read(sdev, BDW_DSP_BAR, SHIM_ISRX); + if (isr & (SHIM_ISRX_DONE | SHIM_ISRX_BUSY)) + ret = IRQ_WAKE_THREAD; + + return ret; +} + +static irqreturn_t bdw_irq_thread(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + u32 ipcx, ipcd, imrx; + + imrx = snd_sof_dsp_read64(sdev, BDW_DSP_BAR, SHIM_IMRX); + ipcx = snd_sof_dsp_read(sdev, BDW_DSP_BAR, SHIM_IPCX); + + /* reply message from DSP */ + if (ipcx & SHIM_IPCX_DONE && + !(imrx & SHIM_IMRX_DONE)) { + /* Mask Done interrupt before return */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, + SHIM_IMRX, SHIM_IMRX_DONE, + SHIM_IMRX_DONE); + + /* + * handle immediate reply from DSP core. If the msg is + * found, set done bit in cmd_done which is called at the + * end of message processing function, else set it here + * because the done bit can't be set in cmd_done function + * which is triggered by msg + */ + bdw_get_reply(sdev); + snd_sof_ipc_reply(sdev, ipcx); + + bdw_dsp_done(sdev); + } + + ipcd = snd_sof_dsp_read(sdev, BDW_DSP_BAR, SHIM_IPCD); + + /* new message from DSP */ + if (ipcd & SHIM_IPCD_BUSY && + !(imrx & SHIM_IMRX_BUSY)) { + /* Mask Busy interrupt before return */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, + SHIM_IMRX, SHIM_IMRX_BUSY, + SHIM_IMRX_BUSY); + + /* Handle messages from DSP Core */ + if ((ipcd & SOF_IPC_PANIC_MAGIC_MASK) == SOF_IPC_PANIC_MAGIC) { + snd_sof_dsp_panic(sdev, BDW_PANIC_OFFSET(ipcx) + + MBOX_OFFSET); + } else { + snd_sof_ipc_msgs_rx(sdev); + } + + bdw_host_done(sdev); + } + + return IRQ_HANDLED; +} + +/* + * IPC Firmware ready. + */ +static void bdw_get_windows(struct snd_sof_dev *sdev) +{ + struct sof_ipc_window_elem *elem; + u32 outbox_offset = 0; + u32 stream_offset = 0; + u32 inbox_offset = 0; + u32 outbox_size = 0; + u32 stream_size = 0; + u32 inbox_size = 0; + int i; + + if (!sdev->info_window) { + dev_err(sdev->dev, "error: have no window info\n"); + return; + } + + for (i = 0; i < sdev->info_window->num_windows; i++) { + elem = &sdev->info_window->window[i]; + + switch (elem->type) { + case SOF_IPC_REGION_UPBOX: + inbox_offset = elem->offset + MBOX_OFFSET; + inbox_size = elem->size; + snd_sof_debugfs_io_item(sdev, + sdev->bar[BDW_DSP_BAR] + + inbox_offset, + elem->size, "inbox", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_DOWNBOX: + outbox_offset = elem->offset + MBOX_OFFSET; + outbox_size = elem->size; + snd_sof_debugfs_io_item(sdev, + sdev->bar[BDW_DSP_BAR] + + outbox_offset, + elem->size, "outbox", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_TRACE: + snd_sof_debugfs_io_item(sdev, + sdev->bar[BDW_DSP_BAR] + + elem->offset + + MBOX_OFFSET, + elem->size, "etrace", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_DEBUG: + snd_sof_debugfs_io_item(sdev, + sdev->bar[BDW_DSP_BAR] + + elem->offset + + MBOX_OFFSET, + elem->size, "debug", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_STREAM: + stream_offset = elem->offset + MBOX_OFFSET; + stream_size = elem->size; + snd_sof_debugfs_io_item(sdev, + sdev->bar[BDW_DSP_BAR] + + stream_offset, + elem->size, "stream", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_REGS: + snd_sof_debugfs_io_item(sdev, + sdev->bar[BDW_DSP_BAR] + + elem->offset + + MBOX_OFFSET, + elem->size, "regs", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_EXCEPTION: + sdev->dsp_oops_offset = elem->offset + MBOX_OFFSET; + snd_sof_debugfs_io_item(sdev, + sdev->bar[BDW_DSP_BAR] + + elem->offset + + MBOX_OFFSET, + elem->size, "exception", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + default: + dev_err(sdev->dev, "error: get illegal window info\n"); + return; + } + } + + if (outbox_size == 0 || inbox_size == 0) { + dev_err(sdev->dev, "error: get illegal mailbox window\n"); + return; + } + + snd_sof_dsp_mailbox_init(sdev, inbox_offset, inbox_size, + outbox_offset, outbox_size); + sdev->stream_box.offset = stream_offset; + sdev->stream_box.size = stream_size; + + dev_dbg(sdev->dev, " mailbox upstream 0x%x - size 0x%x\n", + inbox_offset, inbox_size); + dev_dbg(sdev->dev, " mailbox downstream 0x%x - size 0x%x\n", + outbox_offset, outbox_size); + dev_dbg(sdev->dev, " stream region 0x%x - size 0x%x\n", + stream_offset, stream_size); +} + +/* check for ABI compatibility and create memory windows on first boot */ +static int bdw_fw_ready(struct snd_sof_dev *sdev, u32 msg_id) +{ + struct sof_ipc_fw_ready *fw_ready = &sdev->fw_ready; + u32 offset; + int ret; + + /* mailbox must be on 4k boundary */ + offset = MBOX_OFFSET; + + dev_dbg(sdev->dev, "ipc: DSP is ready 0x%8.8x offset %d\n", + msg_id, offset); + + /* no need to re-check version/ABI for subsequent boots */ + if (!sdev->first_boot) + return 0; + + /* copy data from the DSP FW ready offset */ + sof_block_read(sdev, sdev->mmio_bar, offset, fw_ready, + sizeof(*fw_ready)); + + snd_sof_dsp_mailbox_init(sdev, fw_ready->dspbox_offset, + fw_ready->dspbox_size, + fw_ready->hostbox_offset, + fw_ready->hostbox_size); + + /* make sure ABI version is compatible */ + ret = snd_sof_ipc_valid(sdev); + if (ret < 0) + return ret; + + /* now check for extended data */ + snd_sof_fw_parse_ext_data(sdev, sdev->mmio_bar, MBOX_OFFSET + + sizeof(struct sof_ipc_fw_ready)); + + bdw_get_windows(sdev); + + return 0; +} + +/* + * IPC Mailbox IO + */ + +static int bdw_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) +{ + /* send the message */ + sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data, + msg->msg_size); + snd_sof_dsp_write(sdev, BDW_DSP_BAR, SHIM_IPCX, SHIM_IPCX_BUSY); + + return 0; +} + +static void bdw_get_reply(struct snd_sof_dev *sdev) +{ + struct snd_sof_ipc_msg *msg = sdev->msg; + struct sof_ipc_reply reply; + unsigned long flags; + int ret = 0; + + /* + * Sometimes, there is unexpected reply ipc arriving. The reply + * ipc belongs to none of the ipcs sent from driver. + * In this case, the driver must ignore the ipc. + */ + if (!msg) { + dev_warn(sdev->dev, "unexpected ipc interrupt raised!\n"); + return; + } + + /* get reply */ + sof_mailbox_read(sdev, sdev->host_box.offset, &reply, sizeof(reply)); + + spin_lock_irqsave(&sdev->ipc_lock, flags); + + if (reply.error < 0) { + memcpy(msg->reply_data, &reply, sizeof(reply)); + ret = reply.error; + } else { + /* reply correct size ? */ + if (reply.hdr.size != msg->reply_size) { + dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", + msg->reply_size, reply.hdr.size); + ret = -EINVAL; + } + + /* read the message */ + if (msg->reply_size > 0) + sof_mailbox_read(sdev, sdev->host_box.offset, + msg->reply_data, msg->reply_size); + } + + msg->reply_error = ret; + + spin_unlock_irqrestore(&sdev->ipc_lock, flags); +} + +static void bdw_host_done(struct snd_sof_dev *sdev) +{ + /* clear BUSY bit and set DONE bit - accept new messages */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, SHIM_IPCD, + SHIM_IPCD_BUSY | SHIM_IPCD_DONE, + SHIM_IPCD_DONE); + + /* unmask busy interrupt */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, SHIM_IMRX, + SHIM_IMRX_BUSY, 0); +} + +static void bdw_dsp_done(struct snd_sof_dev *sdev) +{ + /* clear DONE bit - tell DSP we have completed */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, SHIM_IPCX, + SHIM_IPCX_DONE, 0); + + /* unmask Done interrupt */ + snd_sof_dsp_update_bits_unlocked(sdev, BDW_DSP_BAR, SHIM_IMRX, + SHIM_IMRX_DONE, 0); +} + +/* + * Probe and remove. + */ +static int bdw_probe(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *pdata = sdev->pdata; + const struct sof_dev_desc *desc = pdata->desc; + struct platform_device *pdev = + container_of(sdev->dev, struct platform_device, dev); + struct resource *mmio; + u32 base, size; + int ret; + + /* LPE base */ + mmio = platform_get_resource(pdev, IORESOURCE_MEM, + desc->resindex_lpe_base); + if (mmio) { + base = mmio->start; + size = resource_size(mmio); + } else { + dev_err(sdev->dev, "error: failed to get LPE base at idx %d\n", + desc->resindex_lpe_base); + return -EINVAL; + } + + dev_dbg(sdev->dev, "LPE PHY base at 0x%x size 0x%x", base, size); + sdev->bar[BDW_DSP_BAR] = devm_ioremap(sdev->dev, base, size); + if (!sdev->bar[BDW_DSP_BAR]) { + dev_err(sdev->dev, + "error: failed to ioremap LPE base 0x%x size 0x%x\n", + base, size); + return -ENODEV; + } + dev_dbg(sdev->dev, "LPE VADDR %p\n", sdev->bar[BDW_DSP_BAR]); + + /* TODO: add offsets */ + sdev->mmio_bar = BDW_DSP_BAR; + sdev->mailbox_bar = BDW_DSP_BAR; + + /* PCI base */ + mmio = platform_get_resource(pdev, IORESOURCE_MEM, + desc->resindex_pcicfg_base); + if (mmio) { + base = mmio->start; + size = resource_size(mmio); + } else { + dev_err(sdev->dev, "error: failed to get PCI base at idx %d\n", + desc->resindex_pcicfg_base); + return -ENODEV; + } + + dev_dbg(sdev->dev, "PCI base at 0x%x size 0x%x", base, size); + sdev->bar[BDW_PCI_BAR] = devm_ioremap(sdev->dev, base, size); + if (!sdev->bar[BDW_PCI_BAR]) { + dev_err(sdev->dev, + "error: failed to ioremap PCI base 0x%x size 0x%x\n", + base, size); + return -ENODEV; + } + dev_dbg(sdev->dev, "PCI VADDR %p\n", sdev->bar[BDW_PCI_BAR]); + + /* register our IRQ */ + sdev->ipc_irq = platform_get_irq(pdev, desc->irqindex_host_ipc); + if (sdev->ipc_irq < 0) { + dev_err(sdev->dev, "error: failed to get IRQ at index %d\n", + desc->irqindex_host_ipc); + return sdev->ipc_irq; + } + + dev_dbg(sdev->dev, "using IRQ %d\n", sdev->ipc_irq); + ret = devm_request_threaded_irq(sdev->dev, sdev->ipc_irq, + bdw_irq_handler, bdw_irq_thread, + IRQF_SHARED, "AudioDSP", sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to register IRQ %d\n", + sdev->ipc_irq); + return ret; + } + + /* enable the DSP SHIM */ + ret = bdw_set_dsp_D0(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to set DSP D0\n"); + return ret; + } + + /* DSP DMA can only access low 31 bits of host memory */ + ret = dma_coerce_mask_and_coherent(sdev->dev, DMA_BIT_MASK(31)); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to set DMA mask %d\n", ret); + return ret; + } + + /* set default mailbox */ + snd_sof_dsp_mailbox_init(sdev, MBOX_OFFSET, MBOX_SIZE, 0, 0); + + return ret; +} + +/* Broadwell DAIs */ +static struct snd_soc_dai_driver bdw_dai[] = { +{ + .name = "ssp0-port", +}, +{ + .name = "ssp1-port", +}, +}; + +/* broadwell ops */ +const struct snd_sof_dsp_ops sof_bdw_ops = { + /*Device init */ + .probe = bdw_probe, + + /* DSP Core Control */ + .run = bdw_run, + .reset = bdw_reset, + + /* Register IO */ + .write = sof_io_write, + .read = sof_io_read, + .write64 = sof_io_write64, + .read64 = sof_io_read64, + + /* Block IO */ + .block_read = sof_block_read, + .block_write = sof_block_write, + + /* ipc */ + .send_msg = bdw_send_msg, + .fw_ready = bdw_fw_ready, + + .ipc_msg_data = intel_ipc_msg_data, + .ipc_pcm_params = intel_ipc_pcm_params, + + /* debug */ + .debug_map = bdw_debugfs, + .debug_map_count = ARRAY_SIZE(bdw_debugfs), + .dbg_dump = bdw_dump, + + /* stream callbacks */ + .pcm_open = intel_pcm_open, + .pcm_close = intel_pcm_close, + + /* Module loading */ + .load_module = snd_sof_parse_module_memcpy, + + /*Firmware loading */ + .load_firmware = snd_sof_load_firmware_memcpy, + + /* DAI drivers */ + .drv = bdw_dai, + .num_drv = ARRAY_SIZE(bdw_dai) +}; +EXPORT_SYMBOL(sof_bdw_ops); + +const struct sof_intel_dsp_desc bdw_chip_info = { + .cores_num = 1, + .cores_mask = 1, +}; +EXPORT_SYMBOL(bdw_chip_info); + +MODULE_LICENSE("Dual BSD/GPL"); -- cgit v1.2.3-59-g8ed1b From 351d1174fef03adb4defed7ef14f1ede0251aae9 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Fri, 12 Apr 2019 11:08:46 -0500 Subject: ASoC: SOF: Intel: Add legacy IPC support Add IPC support required for devices introduced before Skylake (Merrifield, baytrail, CherryTrail, Haswell, Broadwell) Signed-off-by: Guennadi Liakhovetski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/intel-ipc.c | 92 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 92 insertions(+) create mode 100644 sound/soc/sof/intel/intel-ipc.c (limited to 'sound') diff --git a/sound/soc/sof/intel/intel-ipc.c b/sound/soc/sof/intel/intel-ipc.c new file mode 100644 index 000000000000..4edd92151fd5 --- /dev/null +++ b/sound/soc/sof/intel/intel-ipc.c @@ -0,0 +1,92 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019 Intel Corporation. All rights reserved. +// +// Authors: Guennadi Liakhovetski + +/* Intel-specific SOF IPC code */ + +#include +#include +#include +#include + +#include +#include + +#include "../ops.h" +#include "../sof-priv.h" + +struct intel_stream { + size_t posn_offset; +}; + +/* Mailbox-based Intel IPC implementation */ +void intel_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz) +{ + if (!substream || !sdev->stream_box.size) { + sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz); + } else { + struct intel_stream *stream = substream->runtime->private_data; + + /* The stream might already be closed */ + if (stream) + sof_mailbox_read(sdev, stream->posn_offset, p, sz); + } +} +EXPORT_SYMBOL(intel_ipc_msg_data); + +int intel_ipc_pcm_params(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + const struct sof_ipc_pcm_params_reply *reply) +{ + struct intel_stream *stream = substream->runtime->private_data; + size_t posn_offset = reply->posn_offset; + + /* check if offset is overflow or it is not aligned */ + if (posn_offset > sdev->stream_box.size || + posn_offset % sizeof(struct sof_ipc_stream_posn) != 0) + return -EINVAL; + + stream->posn_offset = sdev->stream_box.offset + posn_offset; + + dev_dbg(sdev->dev, "pcm: stream dir %d, posn mailbox offset is %zu", + substream->stream, stream->posn_offset); + + return 0; +} +EXPORT_SYMBOL(intel_ipc_pcm_params); + +int intel_pcm_open(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream) +{ + struct intel_stream *stream = kmalloc(sizeof(*stream), GFP_KERNEL); + + if (!stream) + return -ENOMEM; + + /* binding pcm substream to hda stream */ + substream->runtime->private_data = stream; + + return 0; +} +EXPORT_SYMBOL(intel_pcm_open); + +int intel_pcm_close(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream) +{ + struct intel_stream *stream = substream->runtime->private_data; + + substream->runtime->private_data = NULL; + kfree(stream); + + return 0; +} +EXPORT_SYMBOL(intel_pcm_close); + +MODULE_LICENSE("Dual BSD/GPL"); -- cgit v1.2.3-59-g8ed1b From dd96daca6c83ecaf37f38ff49d8d174bbff576b4 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:47 -0500 Subject: ASoC: SOF: Intel: Add APL/CNL HW DSP support Add SOF hardware DSP support for Intel Apollolake and Cannonlake based devices. Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 671 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 560 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 1231 insertions(+) create mode 100644 sound/soc/sof/intel/hda.c create mode 100644 sound/soc/sof/intel/hda.h (limited to 'sound') diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c new file mode 100644 index 000000000000..b8fc19790f3b --- /dev/null +++ b/sound/soc/sof/intel/hda.c @@ -0,0 +1,671 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Liam Girdwood +// Ranjani Sridharan +// Rander Wang +// Keyon Jie +// + +/* + * Hardware interface for generic Intel audio DSP HDA IP + */ + +#include +#include +#include +#include +#include "../ops.h" +#include "hda.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) +#include "../../codecs/hdac_hda.h" +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) +#include +#endif + +/* platform specific devices */ +#include "shim.h" + +/* + * Debug + */ + +struct hda_dsp_msg_code { + u32 code; + const char *msg; +}; + +static const struct hda_dsp_msg_code hda_dsp_rom_msg[] = { + {HDA_DSP_ROM_FW_MANIFEST_LOADED, "status: manifest loaded"}, + {HDA_DSP_ROM_FW_FW_LOADED, "status: fw loaded"}, + {HDA_DSP_ROM_FW_ENTERED, "status: fw entered"}, + {HDA_DSP_ROM_CSE_ERROR, "error: cse error"}, + {HDA_DSP_ROM_CSE_WRONG_RESPONSE, "error: cse wrong response"}, + {HDA_DSP_ROM_IMR_TO_SMALL, "error: IMR too small"}, + {HDA_DSP_ROM_BASE_FW_NOT_FOUND, "error: base fw not found"}, + {HDA_DSP_ROM_CSE_VALIDATION_FAILED, "error: signature verification failed"}, + {HDA_DSP_ROM_IPC_FATAL_ERROR, "error: ipc fatal error"}, + {HDA_DSP_ROM_L2_CACHE_ERROR, "error: L2 cache error"}, + {HDA_DSP_ROM_LOAD_OFFSET_TO_SMALL, "error: load offset too small"}, + {HDA_DSP_ROM_API_PTR_INVALID, "error: API ptr invalid"}, + {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatble"}, + {HDA_DSP_ROM_UNHANDLED_INTERRUPT, "error: unhandled interrupt"}, + {HDA_DSP_ROM_MEMORY_HOLE_ECC, "error: ECC memory hole"}, + {HDA_DSP_ROM_KERNEL_EXCEPTION, "error: kernel exception"}, + {HDA_DSP_ROM_USER_EXCEPTION, "error: user exception"}, + {HDA_DSP_ROM_UNEXPECTED_RESET, "error: unexpected reset"}, + {HDA_DSP_ROM_NULL_FW_ENTRY, "error: null FW entry point"}, +}; + +static void hda_dsp_get_status_skl(struct snd_sof_dev *sdev) +{ + u32 status; + int i; + + status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_ADSP_FW_STATUS_SKL); + + for (i = 0; i < ARRAY_SIZE(hda_dsp_rom_msg); i++) { + if (status == hda_dsp_rom_msg[i].code) { + dev_err(sdev->dev, "%s - code %8.8x\n", + hda_dsp_rom_msg[i].msg, status); + return; + } + } + + /* not for us, must be generic sof message */ + dev_dbg(sdev->dev, "unknown ROM status value %8.8x\n", status); +} + +static void hda_dsp_get_status(struct snd_sof_dev *sdev) +{ + u32 status; + int i; + + status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_SRAM_REG_ROM_STATUS); + + for (i = 0; i < ARRAY_SIZE(hda_dsp_rom_msg); i++) { + if (status == hda_dsp_rom_msg[i].code) { + dev_err(sdev->dev, "%s - code %8.8x\n", + hda_dsp_rom_msg[i].msg, status); + return; + } + } + + /* not for us, must be generic sof message */ + dev_dbg(sdev->dev, "unknown ROM status value %8.8x\n", status); +} + +static void hda_dsp_get_registers(struct snd_sof_dev *sdev, + struct sof_ipc_dsp_oops_xtensa *xoops, + struct sof_ipc_panic_info *panic_info, + u32 *stack, size_t stack_words) +{ + /* first read registers */ + sof_block_read(sdev, sdev->mmio_bar, sdev->dsp_oops_offset, xoops, + sizeof(*xoops)); + + /* then get panic info */ + sof_block_read(sdev, sdev->mmio_bar, sdev->dsp_oops_offset + + sizeof(*xoops), panic_info, sizeof(*panic_info)); + + /* then get the stack */ + sof_block_read(sdev, sdev->mmio_bar, sdev->dsp_oops_offset + + sizeof(*xoops) + sizeof(*panic_info), stack, + stack_words * sizeof(u32)); +} + +void hda_dsp_dump_skl(struct snd_sof_dev *sdev, u32 flags) +{ + struct sof_ipc_dsp_oops_xtensa xoops; + struct sof_ipc_panic_info panic_info; + u32 stack[HDA_DSP_STACK_DUMP_SIZE]; + u32 status, panic; + + /* try APL specific status message types first */ + hda_dsp_get_status_skl(sdev); + + /* now try generic SOF status messages */ + status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_ADSP_ERROR_CODE_SKL); + + /*TODO: Check: there is no define in spec, but it is used in the code*/ + panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_ADSP_ERROR_CODE_SKL + 0x4); + + if (sdev->boot_complete) { + hda_dsp_get_registers(sdev, &xoops, &panic_info, stack, + HDA_DSP_STACK_DUMP_SIZE); + snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, + stack, HDA_DSP_STACK_DUMP_SIZE); + } else { + dev_err(sdev->dev, "error: status = 0x%8.8x panic = 0x%8.8x\n", + status, panic); + hda_dsp_get_status_skl(sdev); + } +} + +void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags) +{ + struct sof_ipc_dsp_oops_xtensa xoops; + struct sof_ipc_panic_info panic_info; + u32 stack[HDA_DSP_STACK_DUMP_SIZE]; + u32 status, panic; + + /* try APL specific status message types first */ + hda_dsp_get_status(sdev); + + /* now try generic SOF status messages */ + status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_SRAM_REG_FW_STATUS); + panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP); + + if (sdev->boot_complete) { + hda_dsp_get_registers(sdev, &xoops, &panic_info, stack, + HDA_DSP_STACK_DUMP_SIZE); + snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, + stack, HDA_DSP_STACK_DUMP_SIZE); + } else { + dev_err(sdev->dev, "error: status = 0x%8.8x panic = 0x%8.8x\n", + status, panic); + hda_dsp_get_status(sdev); + } +} + +static int hda_init(struct snd_sof_dev *sdev) +{ + struct hda_bus *hbus; + struct hdac_bus *bus; + struct hdac_ext_bus_ops *ext_ops = NULL; + struct pci_dev *pci = to_pci_dev(sdev->dev); + int ret; + + hbus = sof_to_hbus(sdev); + bus = sof_to_bus(sdev); + + /* HDA bus init */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + ext_ops = snd_soc_hdac_hda_get_ops(); +#endif + sof_hda_bus_init(bus, &pci->dev, ext_ops); + bus->use_posbuf = 1; + bus->bdl_pos_adj = 0; + + mutex_init(&hbus->prepare_mutex); + hbus->pci = pci; + hbus->mixer_assigned = -1; + hbus->modelname = "sofbus"; + + /* initialise hdac bus */ + bus->addr = pci_resource_start(pci, 0); + bus->remap_addr = pci_ioremap_bar(pci, 0); + if (!bus->remap_addr) { + dev_err(bus->dev, "error: ioremap error\n"); + return -ENXIO; + } + + /* HDA base */ + sdev->bar[HDA_DSP_HDA_BAR] = bus->remap_addr; + + /* get controller capabilities */ + ret = hda_dsp_ctrl_get_caps(sdev); + if (ret < 0) + dev_err(sdev->dev, "error: get caps error\n"); + + return ret; +} + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + +static const char *fixup_tplg_name(struct snd_sof_dev *sdev, + const char *sof_tplg_filename) +{ + const char *tplg_filename = NULL; + char *filename; + char *split_ext; + + filename = devm_kstrdup(sdev->dev, sof_tplg_filename, GFP_KERNEL); + if (!filename) + return NULL; + + /* this assumes a .tplg extension */ + split_ext = strsep(&filename, "."); + if (split_ext) { + tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, + "%s-idisp.tplg", split_ext); + if (!tplg_filename) + return NULL; + } + return tplg_filename; +} + +static int hda_init_caps(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct hdac_ext_link *hlink; + struct snd_soc_acpi_mach_params *mach_params; + struct snd_soc_acpi_mach *hda_mach; + struct snd_sof_pdata *pdata = sdev->pdata; + struct snd_soc_acpi_mach *mach; + const char *tplg_filename; + int codec_num = 0; + int ret = 0; + int i; + + device_disable_async_suspend(bus->dev); + + /* check if dsp is there */ + if (bus->ppcap) + dev_dbg(sdev->dev, "PP capability, will probe DSP later.\n"); + + if (bus->mlcap) + snd_hdac_ext_bus_get_ml_capabilities(bus); + + /* init i915 and HDMI codecs */ + ret = hda_codec_i915_init(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: no HDMI audio devices found\n"); + return ret; + } + + ret = hda_dsp_ctrl_init_chip(sdev, true); + if (ret < 0) { + dev_err(bus->dev, "error: init chip failed with ret: %d\n", ret); + goto out; + } + + /* codec detection */ + if (!bus->codec_mask) { + dev_info(bus->dev, "no hda codecs found!\n"); + } else { + dev_info(bus->dev, "hda codecs found, mask %lx\n", + bus->codec_mask); + + for (i = 0; i < HDA_MAX_CODECS; i++) { + if (bus->codec_mask & (1 << i)) + codec_num++; + } + + /* + * If no machine driver is found, then: + * + * hda machine driver is used if : + * 1. there is one HDMI codec and one external HDAudio codec + * 2. only HDMI codec + */ + if (!pdata->machine && codec_num <= 2 && + HDA_IDISP_CODEC(bus->codec_mask)) { + hda_mach = snd_soc_acpi_intel_hda_machines; + pdata->machine = hda_mach; + + /* topology: use the info from hda_machines */ + pdata->tplg_filename = + hda_mach->sof_tplg_filename; + + /* firmware: pick the first in machine list */ + mach = pdata->desc->machines; + pdata->fw_filename = mach->sof_fw_filename; + + dev_info(bus->dev, "using HDA machine driver %s now\n", + hda_mach->drv_name); + + /* fixup topology file for HDMI only platforms */ + if (codec_num == 1) { + /* use local variable for readability */ + tplg_filename = pdata->tplg_filename; + tplg_filename = fixup_tplg_name(sdev, tplg_filename); + if (!tplg_filename) + goto out; + pdata->tplg_filename = tplg_filename; + } + } + } + + /* used by hda machine driver to create dai links */ + if (pdata->machine) { + mach_params = (struct snd_soc_acpi_mach_params *) + &pdata->machine->mach_params; + mach_params->codec_mask = bus->codec_mask; + mach_params->platform = dev_name(sdev->dev); + } + + /* create codec instances */ + hda_codec_probe_bus(sdev); + + hda_codec_i915_put(sdev); + + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &bus->hlink_list, list) + snd_hdac_ext_bus_link_put(bus, hlink); + + return 0; + +out: + hda_codec_i915_exit(sdev); + return ret; +} + +#else + +static int hda_init_caps(struct snd_sof_dev *sdev) +{ + /* + * set CGCTL.MISCBDCGE to 0 during reset and set back to 1 + * when reset finished. + * TODO: maybe no need for init_caps? + */ + hda_dsp_ctrl_misc_clock_gating(sdev, 0); + + /* clear WAKESTS */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_WAKESTS, + SOF_HDA_WAKESTS_INT_MASK, + SOF_HDA_WAKESTS_INT_MASK); + + return 0; +} + +#endif + +static const struct sof_intel_dsp_desc + *get_chip_info(struct snd_sof_pdata *pdata) +{ + const struct sof_dev_desc *desc = pdata->desc; + const struct sof_intel_dsp_desc *chip_info; + + chip_info = desc->chip_info; + + return chip_info; +} + +int hda_dsp_probe(struct snd_sof_dev *sdev) +{ + struct pci_dev *pci = to_pci_dev(sdev->dev); + struct sof_intel_hda_dev *hdev; + struct hdac_bus *bus; + struct hdac_stream *stream; + const struct sof_intel_dsp_desc *chip; + int sd_offset, ret = 0; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, legacy driver is required + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: either of DSP or legacy mode works + */ + if (pci->class == 0x040300) { + dev_err(sdev->dev, "error: the DSP is not enabled on this platform, aborting probe\n"); + return -ENODEV; + } else if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(sdev->dev, "error: unknown PCI class/subclass/prog-if 0x%06x found, aborting probe\n", pci->class); + return -ENODEV; + } + dev_info(sdev->dev, "DSP detected with PCI class/subclass/prog-if 0x%06x\n", pci->class); + + chip = get_chip_info(sdev->pdata); + if (!chip) { + dev_err(sdev->dev, "error: no such device supported, chip id:%x\n", + pci->device); + ret = -EIO; + goto err; + } + + hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL); + if (!hdev) + return -ENOMEM; + sdev->pdata->hw_pdata = hdev; + hdev->desc = chip; + + hdev->dmic_dev = platform_device_register_data(sdev->dev, "dmic-codec", + PLATFORM_DEVID_NONE, + NULL, 0); + if (IS_ERR(hdev->dmic_dev)) { + dev_err(sdev->dev, "error: failed to create DMIC device\n"); + return PTR_ERR(hdev->dmic_dev); + } + + /* + * use position update IPC if either it is forced + * or we don't have other choice + */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_FORCE_IPC_POSITION) + hdev->no_ipc_position = 0; +#else + hdev->no_ipc_position = sof_ops(sdev)->pcm_pointer ? 1 : 0; +#endif + + /* set up HDA base */ + bus = sof_to_bus(sdev); + ret = hda_init(sdev); + if (ret < 0) + goto hdac_bus_unmap; + + /* DSP base */ + sdev->bar[HDA_DSP_BAR] = pci_ioremap_bar(pci, HDA_DSP_BAR); + if (!sdev->bar[HDA_DSP_BAR]) { + dev_err(sdev->dev, "error: ioremap error\n"); + ret = -ENXIO; + goto hdac_bus_unmap; + } + + sdev->mmio_bar = HDA_DSP_BAR; + sdev->mailbox_bar = HDA_DSP_BAR; + + /* allow 64bit DMA address if supported by H/W */ + if (!dma_set_mask(&pci->dev, DMA_BIT_MASK(64))) { + dev_dbg(sdev->dev, "DMA mask is 64 bit\n"); + dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(64)); + } else { + dev_dbg(sdev->dev, "DMA mask is 32 bit\n"); + dma_set_mask(&pci->dev, DMA_BIT_MASK(32)); + dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(32)); + } + + /* init streams */ + ret = hda_dsp_stream_init(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to init streams\n"); + /* + * not all errors are due to memory issues, but trying + * to free everything does not harm + */ + goto free_streams; + } + + /* + * register our IRQ + * let's try to enable msi firstly + * if it fails, use legacy interrupt mode + * TODO: support interrupt mode selection with kernel parameter + * support msi multiple vectors + */ + ret = pci_alloc_irq_vectors(pci, 1, 1, PCI_IRQ_MSI); + if (ret < 0) { + dev_info(sdev->dev, "use legacy interrupt mode\n"); + /* + * in IO-APIC mode, hda->irq and ipc_irq are using the same + * irq number of pci->irq + */ + hdev->irq = pci->irq; + sdev->ipc_irq = pci->irq; + sdev->msi_enabled = 0; + } else { + dev_info(sdev->dev, "use msi interrupt mode\n"); + hdev->irq = pci_irq_vector(pci, 0); + /* ipc irq number is the same of hda irq */ + sdev->ipc_irq = hdev->irq; + sdev->msi_enabled = 1; + } + + dev_dbg(sdev->dev, "using HDA IRQ %d\n", hdev->irq); + ret = request_threaded_irq(hdev->irq, hda_dsp_stream_interrupt, + hda_dsp_stream_threaded_handler, + IRQF_SHARED, "AudioHDA", bus); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to register HDA IRQ %d\n", + hdev->irq); + goto free_irq_vector; + } + + dev_dbg(sdev->dev, "using IPC IRQ %d\n", sdev->ipc_irq); + ret = request_threaded_irq(sdev->ipc_irq, hda_dsp_ipc_irq_handler, + sof_ops(sdev)->irq_thread, IRQF_SHARED, + "AudioDSP", sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to register IPC IRQ %d\n", + sdev->ipc_irq); + goto free_hda_irq; + } + + pci_set_master(pci); + synchronize_irq(pci->irq); + + /* + * clear TCSEL to clear playback on some HD Audio + * codecs. PCI TCSEL is defined in the Intel manuals. + */ + snd_sof_pci_update_bits(sdev, PCI_TCSEL, 0x07, 0); + + /* init HDA capabilities */ + ret = hda_init_caps(sdev); + if (ret < 0) + goto free_ipc_irq; + + /* reset HDA controller */ + ret = hda_dsp_ctrl_link_reset(sdev, true); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to reset HDA controller\n"); + goto free_ipc_irq; + } + + /* exit HDA controller reset */ + ret = hda_dsp_ctrl_link_reset(sdev, false); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to exit HDA controller reset\n"); + goto free_ipc_irq; + } + + /* clear stream status */ + list_for_each_entry(stream, &bus->stream_list, list) { + sd_offset = SOF_STREAM_SD_OFFSET(stream); + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + sd_offset + + SOF_HDA_ADSP_REG_CL_SD_STS, + SOF_HDA_CL_DMA_SD_INT_MASK, + SOF_HDA_CL_DMA_SD_INT_MASK); + } + + /* clear WAKESTS */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_WAKESTS, + SOF_HDA_WAKESTS_INT_MASK, + SOF_HDA_WAKESTS_INT_MASK); + + /* clear interrupt status register */ + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS, + SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_ALL_STREAM); + + /* enable CIE and GIE interrupts */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, + SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, + SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN); + + /* re-enable CGCTL.MISCBDCGE after reset */ + hda_dsp_ctrl_misc_clock_gating(sdev, true); + + device_disable_async_suspend(&pci->dev); + + /* enable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + SOF_HDA_PPCTL_GPROCEN, SOF_HDA_PPCTL_GPROCEN); + + /* enable DSP IRQ */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + SOF_HDA_PPCTL_PIE, SOF_HDA_PPCTL_PIE); + + /* initialize waitq for code loading */ + init_waitqueue_head(&sdev->waitq); + + /* set default mailbox offset for FW ready message */ + sdev->dsp_box.offset = HDA_DSP_MBOX_UPLINK_OFFSET; + + return 0; + +free_ipc_irq: + free_irq(sdev->ipc_irq, sdev); +free_hda_irq: + free_irq(hdev->irq, bus); +free_irq_vector: + if (sdev->msi_enabled) + pci_free_irq_vectors(pci); +free_streams: + hda_dsp_stream_free(sdev); +/* dsp_unmap: not currently used */ + iounmap(sdev->bar[HDA_DSP_BAR]); +hdac_bus_unmap: + iounmap(bus->remap_addr); +err: + return ret; +} + +int hda_dsp_remove(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + struct hdac_bus *bus = sof_to_bus(sdev); + struct pci_dev *pci = to_pci_dev(sdev->dev); + const struct sof_intel_dsp_desc *chip = hda->desc; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* codec removal, invoke bus_device_remove */ + snd_hdac_ext_bus_device_remove(bus); +#endif + + if (!IS_ERR_OR_NULL(hda->dmic_dev)) + platform_device_unregister(hda->dmic_dev); + + /* disable DSP IRQ */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + SOF_HDA_PPCTL_PIE, 0); + + /* disable CIE and GIE interrupts */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, + SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, 0); + + /* disable cores */ + if (chip) + hda_dsp_core_reset_power_down(sdev, chip->cores_mask); + + /* disable DSP */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + SOF_HDA_PPCTL_GPROCEN, 0); + + free_irq(sdev->ipc_irq, sdev); + free_irq(hda->irq, bus); + if (sdev->msi_enabled) + pci_free_irq_vectors(pci); + + hda_dsp_stream_free(sdev); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + snd_hdac_link_free_all(bus); +#endif + + iounmap(sdev->bar[HDA_DSP_BAR]); + iounmap(bus->remap_addr); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + snd_hdac_ext_bus_exit(bus); +#endif + hda_codec_i915_exit(sdev); + + return 0; +} + +MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h new file mode 100644 index 000000000000..4efcf85af6b5 --- /dev/null +++ b/sound/soc/sof/intel/hda.h @@ -0,0 +1,560 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2017 Intel Corporation. All rights reserved. + * + * Author: Liam Girdwood + */ + +#ifndef __SOF_INTEL_HDA_H +#define __SOF_INTEL_HDA_H + +#include +#include +#include "shim.h" + +/* PCI registers */ +#define PCI_TCSEL 0x44 +#define PCI_PGCTL PCI_TCSEL +#define PCI_CGCTL 0x48 + +/* PCI_PGCTL bits */ +#define PCI_PGCTL_ADSPPGD BIT(2) +#define PCI_PGCTL_LSRMD_MASK BIT(4) + +/* PCI_CGCTL bits */ +#define PCI_CGCTL_MISCBDCGE_MASK BIT(6) +#define PCI_CGCTL_ADSPDCGE BIT(1) + +/* Legacy HDA registers and bits used - widths are variable */ +#define SOF_HDA_GCAP 0x0 +#define SOF_HDA_GCTL 0x8 +/* accept unsol. response enable */ +#define SOF_HDA_GCTL_UNSOL BIT(8) +#define SOF_HDA_LLCH 0x14 +#define SOF_HDA_INTCTL 0x20 +#define SOF_HDA_INTSTS 0x24 +#define SOF_HDA_WAKESTS 0x0E +#define SOF_HDA_WAKESTS_INT_MASK ((1 << 8) - 1) +#define SOF_HDA_RIRBSTS 0x5d +#define SOF_HDA_VS_EM2_L1SEN BIT(13) + +/* SOF_HDA_GCTL register bist */ +#define SOF_HDA_GCTL_RESET BIT(0) + +/* SOF_HDA_INCTL and SOF_HDA_INTSTS regs */ +#define SOF_HDA_INT_GLOBAL_EN BIT(31) +#define SOF_HDA_INT_CTRL_EN BIT(30) +#define SOF_HDA_INT_ALL_STREAM 0xff + +#define SOF_HDA_MAX_CAPS 10 +#define SOF_HDA_CAP_ID_OFF 16 +#define SOF_HDA_CAP_ID_MASK GENMASK(SOF_HDA_CAP_ID_OFF + 11,\ + SOF_HDA_CAP_ID_OFF) +#define SOF_HDA_CAP_NEXT_MASK 0xFFFF + +#define SOF_HDA_GTS_CAP_ID 0x1 +#define SOF_HDA_ML_CAP_ID 0x2 + +#define SOF_HDA_PP_CAP_ID 0x3 +#define SOF_HDA_REG_PP_PPCH 0x10 +#define SOF_HDA_REG_PP_PPCTL 0x04 +#define SOF_HDA_PPCTL_PIE BIT(31) +#define SOF_HDA_PPCTL_GPROCEN BIT(30) + +/* DPIB entry size: 8 Bytes = 2 DWords */ +#define SOF_HDA_DPIB_ENTRY_SIZE 0x8 + +#define SOF_HDA_SPIB_CAP_ID 0x4 +#define SOF_HDA_DRSM_CAP_ID 0x5 + +#define SOF_HDA_SPIB_BASE 0x08 +#define SOF_HDA_SPIB_INTERVAL 0x08 +#define SOF_HDA_SPIB_SPIB 0x00 +#define SOF_HDA_SPIB_MAXFIFO 0x04 + +#define SOF_HDA_PPHC_BASE 0x10 +#define SOF_HDA_PPHC_INTERVAL 0x10 + +#define SOF_HDA_PPLC_BASE 0x10 +#define SOF_HDA_PPLC_MULTI 0x10 +#define SOF_HDA_PPLC_INTERVAL 0x10 + +#define SOF_HDA_DRSM_BASE 0x08 +#define SOF_HDA_DRSM_INTERVAL 0x08 + +/* Descriptor error interrupt */ +#define SOF_HDA_CL_DMA_SD_INT_DESC_ERR 0x10 + +/* FIFO error interrupt */ +#define SOF_HDA_CL_DMA_SD_INT_FIFO_ERR 0x08 + +/* Buffer completion interrupt */ +#define SOF_HDA_CL_DMA_SD_INT_COMPLETE 0x04 + +#define SOF_HDA_CL_DMA_SD_INT_MASK \ + (SOF_HDA_CL_DMA_SD_INT_DESC_ERR | \ + SOF_HDA_CL_DMA_SD_INT_FIFO_ERR | \ + SOF_HDA_CL_DMA_SD_INT_COMPLETE) +#define SOF_HDA_SD_CTL_DMA_START 0x02 /* Stream DMA start bit */ + +/* Intel HD Audio Code Loader DMA Registers */ +#define SOF_HDA_ADSP_LOADER_BASE 0x80 +#define SOF_HDA_ADSP_DPLBASE 0x70 +#define SOF_HDA_ADSP_DPUBASE 0x74 +#define SOF_HDA_ADSP_DPLBASE_ENABLE 0x01 + +/* Stream Registers */ +#define SOF_HDA_ADSP_REG_CL_SD_CTL 0x00 +#define SOF_HDA_ADSP_REG_CL_SD_STS 0x03 +#define SOF_HDA_ADSP_REG_CL_SD_LPIB 0x04 +#define SOF_HDA_ADSP_REG_CL_SD_CBL 0x08 +#define SOF_HDA_ADSP_REG_CL_SD_LVI 0x0C +#define SOF_HDA_ADSP_REG_CL_SD_FIFOW 0x0E +#define SOF_HDA_ADSP_REG_CL_SD_FIFOSIZE 0x10 +#define SOF_HDA_ADSP_REG_CL_SD_FORMAT 0x12 +#define SOF_HDA_ADSP_REG_CL_SD_FIFOL 0x14 +#define SOF_HDA_ADSP_REG_CL_SD_BDLPL 0x18 +#define SOF_HDA_ADSP_REG_CL_SD_BDLPU 0x1C +#define SOF_HDA_ADSP_SD_ENTRY_SIZE 0x20 + +/* CL: Software Position Based FIFO Capability Registers */ +#define SOF_DSP_REG_CL_SPBFIFO \ + (SOF_HDA_ADSP_LOADER_BASE + 0x20) +#define SOF_HDA_ADSP_REG_CL_SPBFIFO_SPBFCH 0x0 +#define SOF_HDA_ADSP_REG_CL_SPBFIFO_SPBFCCTL 0x4 +#define SOF_HDA_ADSP_REG_CL_SPBFIFO_SPIB 0x8 +#define SOF_HDA_ADSP_REG_CL_SPBFIFO_MAXFIFOS 0xc + +/* Stream Number */ +#define SOF_HDA_CL_SD_CTL_STREAM_TAG_SHIFT 20 +#define SOF_HDA_CL_SD_CTL_STREAM_TAG_MASK \ + GENMASK(SOF_HDA_CL_SD_CTL_STREAM_TAG_SHIFT + 3,\ + SOF_HDA_CL_SD_CTL_STREAM_TAG_SHIFT) + +#define HDA_DSP_HDA_BAR 0 +#define HDA_DSP_PP_BAR 1 +#define HDA_DSP_SPIB_BAR 2 +#define HDA_DSP_DRSM_BAR 3 +#define HDA_DSP_BAR 4 + +#define SRAM_WINDOW_OFFSET(x) (0x80000 + (x) * 0x20000) + +#define HDA_DSP_MBOX_OFFSET SRAM_WINDOW_OFFSET(0) + +#define HDA_DSP_PANIC_OFFSET(x) \ + (((x) & 0xFFFFFF) + HDA_DSP_MBOX_OFFSET) + +/* SRAM window 0 FW "registers" */ +#define HDA_DSP_SRAM_REG_ROM_STATUS (HDA_DSP_MBOX_OFFSET + 0x0) +#define HDA_DSP_SRAM_REG_ROM_ERROR (HDA_DSP_MBOX_OFFSET + 0x4) +/* FW and ROM share offset 4 */ +#define HDA_DSP_SRAM_REG_FW_STATUS (HDA_DSP_MBOX_OFFSET + 0x4) +#define HDA_DSP_SRAM_REG_FW_TRACEP (HDA_DSP_MBOX_OFFSET + 0x8) +#define HDA_DSP_SRAM_REG_FW_END (HDA_DSP_MBOX_OFFSET + 0xc) + +#define HDA_DSP_MBOX_UPLINK_OFFSET 0x81000 + +#define HDA_DSP_STREAM_RESET_TIMEOUT 300 +#define HDA_DSP_CL_TRIGGER_TIMEOUT 300 + +#define HDA_DSP_SPIB_ENABLE 1 +#define HDA_DSP_SPIB_DISABLE 0 + +#define SOF_HDA_MAX_BUFFER_SIZE (32 * PAGE_SIZE) + +#define HDA_DSP_STACK_DUMP_SIZE 32 + +/* ROM status/error values */ +#define HDA_DSP_ROM_STS_MASK 0xf +#define HDA_DSP_ROM_INIT 0x1 +#define HDA_DSP_ROM_FW_MANIFEST_LOADED 0x3 +#define HDA_DSP_ROM_FW_FW_LOADED 0x4 +#define HDA_DSP_ROM_FW_ENTERED 0x5 +#define HDA_DSP_ROM_RFW_START 0xf +#define HDA_DSP_ROM_CSE_ERROR 40 +#define HDA_DSP_ROM_CSE_WRONG_RESPONSE 41 +#define HDA_DSP_ROM_IMR_TO_SMALL 42 +#define HDA_DSP_ROM_BASE_FW_NOT_FOUND 43 +#define HDA_DSP_ROM_CSE_VALIDATION_FAILED 44 +#define HDA_DSP_ROM_IPC_FATAL_ERROR 45 +#define HDA_DSP_ROM_L2_CACHE_ERROR 46 +#define HDA_DSP_ROM_LOAD_OFFSET_TO_SMALL 47 +#define HDA_DSP_ROM_API_PTR_INVALID 50 +#define HDA_DSP_ROM_BASEFW_INCOMPAT 51 +#define HDA_DSP_ROM_UNHANDLED_INTERRUPT 0xBEE00000 +#define HDA_DSP_ROM_MEMORY_HOLE_ECC 0xECC00000 +#define HDA_DSP_ROM_KERNEL_EXCEPTION 0xCAFE0000 +#define HDA_DSP_ROM_USER_EXCEPTION 0xBEEF0000 +#define HDA_DSP_ROM_UNEXPECTED_RESET 0xDECAF000 +#define HDA_DSP_ROM_NULL_FW_ENTRY 0x4c4c4e55 +#define HDA_DSP_IPC_PURGE_FW 0x01004000 + +/* various timeout values */ +#define HDA_DSP_PU_TIMEOUT 50 +#define HDA_DSP_PD_TIMEOUT 50 +#define HDA_DSP_RESET_TIMEOUT_US 50000 +#define HDA_DSP_BASEFW_TIMEOUT_US 3000000 +#define HDA_DSP_INIT_TIMEOUT_US 500000 +#define HDA_DSP_CTRL_RESET_TIMEOUT 100 +#define HDA_DSP_WAIT_TIMEOUT 500 /* 500 msec */ +#define HDA_DSP_REG_POLL_INTERVAL_US 500 /* 0.5 msec */ + +#define HDA_DSP_ADSPIC_IPC 1 +#define HDA_DSP_ADSPIS_IPC 1 + +/* Intel HD Audio General DSP Registers */ +#define HDA_DSP_GEN_BASE 0x0 +#define HDA_DSP_REG_ADSPCS (HDA_DSP_GEN_BASE + 0x04) +#define HDA_DSP_REG_ADSPIC (HDA_DSP_GEN_BASE + 0x08) +#define HDA_DSP_REG_ADSPIS (HDA_DSP_GEN_BASE + 0x0C) +#define HDA_DSP_REG_ADSPIC2 (HDA_DSP_GEN_BASE + 0x10) +#define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14) + +/* Intel HD Audio Inter-Processor Communication Registers */ +#define HDA_DSP_IPC_BASE 0x40 +#define HDA_DSP_REG_HIPCT (HDA_DSP_IPC_BASE + 0x00) +#define HDA_DSP_REG_HIPCTE (HDA_DSP_IPC_BASE + 0x04) +#define HDA_DSP_REG_HIPCI (HDA_DSP_IPC_BASE + 0x08) +#define HDA_DSP_REG_HIPCIE (HDA_DSP_IPC_BASE + 0x0C) +#define HDA_DSP_REG_HIPCCTL (HDA_DSP_IPC_BASE + 0x10) + +/* HIPCI */ +#define HDA_DSP_REG_HIPCI_BUSY BIT(31) +#define HDA_DSP_REG_HIPCI_MSG_MASK 0x7FFFFFFF + +/* HIPCIE */ +#define HDA_DSP_REG_HIPCIE_DONE BIT(30) +#define HDA_DSP_REG_HIPCIE_MSG_MASK 0x3FFFFFFF + +/* HIPCCTL */ +#define HDA_DSP_REG_HIPCCTL_DONE BIT(1) +#define HDA_DSP_REG_HIPCCTL_BUSY BIT(0) + +/* HIPCT */ +#define HDA_DSP_REG_HIPCT_BUSY BIT(31) +#define HDA_DSP_REG_HIPCT_MSG_MASK 0x7FFFFFFF + +/* HIPCTE */ +#define HDA_DSP_REG_HIPCTE_MSG_MASK 0x3FFFFFFF + +#define HDA_DSP_ADSPIC_CL_DMA 0x2 +#define HDA_DSP_ADSPIS_CL_DMA 0x2 + +/* Delay before scheduling D0i3 entry */ +#define BXT_D0I3_DELAY 5000 + +#define FW_CL_STREAM_NUMBER 0x1 + +/* ADSPCS - Audio DSP Control & Status */ + +/* + * Core Reset - asserted high + * CRST Mask for a given core mask pattern, cm + */ +#define HDA_DSP_ADSPCS_CRST_SHIFT 0 +#define HDA_DSP_ADSPCS_CRST_MASK(cm) ((cm) << HDA_DSP_ADSPCS_CRST_SHIFT) + +/* + * Core run/stall - when set to '1' core is stalled + * CSTALL Mask for a given core mask pattern, cm + */ +#define HDA_DSP_ADSPCS_CSTALL_SHIFT 8 +#define HDA_DSP_ADSPCS_CSTALL_MASK(cm) ((cm) << HDA_DSP_ADSPCS_CSTALL_SHIFT) + +/* + * Set Power Active - when set to '1' turn cores on + * SPA Mask for a given core mask pattern, cm + */ +#define HDA_DSP_ADSPCS_SPA_SHIFT 16 +#define HDA_DSP_ADSPCS_SPA_MASK(cm) ((cm) << HDA_DSP_ADSPCS_SPA_SHIFT) + +/* + * Current Power Active - power status of cores, set by hardware + * CPA Mask for a given core mask pattern, cm + */ +#define HDA_DSP_ADSPCS_CPA_SHIFT 24 +#define HDA_DSP_ADSPCS_CPA_MASK(cm) ((cm) << HDA_DSP_ADSPCS_CPA_SHIFT) + +/* Mask for a given core index, c = 0.. number of supported cores - 1 */ +#define HDA_DSP_CORE_MASK(c) BIT(c) + +/* + * Mask for a given number of cores + * nc = number of supported cores + */ +#define SOF_DSP_CORES_MASK(nc) GENMASK(((nc) - 1), 0) + +/* Intel HD Audio Inter-Processor Communication Registers for Cannonlake*/ +#define CNL_DSP_IPC_BASE 0xc0 +#define CNL_DSP_REG_HIPCTDR (CNL_DSP_IPC_BASE + 0x00) +#define CNL_DSP_REG_HIPCTDA (CNL_DSP_IPC_BASE + 0x04) +#define CNL_DSP_REG_HIPCTDD (CNL_DSP_IPC_BASE + 0x08) +#define CNL_DSP_REG_HIPCIDR (CNL_DSP_IPC_BASE + 0x10) +#define CNL_DSP_REG_HIPCIDA (CNL_DSP_IPC_BASE + 0x14) +#define CNL_DSP_REG_HIPCCTL (CNL_DSP_IPC_BASE + 0x28) + +/* HIPCI */ +#define CNL_DSP_REG_HIPCIDR_BUSY BIT(31) +#define CNL_DSP_REG_HIPCIDR_MSG_MASK 0x7FFFFFFF + +/* HIPCIE */ +#define CNL_DSP_REG_HIPCIDA_DONE BIT(31) +#define CNL_DSP_REG_HIPCIDA_MSG_MASK 0x7FFFFFFF + +/* HIPCCTL */ +#define CNL_DSP_REG_HIPCCTL_DONE BIT(1) +#define CNL_DSP_REG_HIPCCTL_BUSY BIT(0) + +/* HIPCT */ +#define CNL_DSP_REG_HIPCTDR_BUSY BIT(31) +#define CNL_DSP_REG_HIPCTDR_MSG_MASK 0x7FFFFFFF + +/* HIPCTDA */ +#define CNL_DSP_REG_HIPCTDA_DONE BIT(31) +#define CNL_DSP_REG_HIPCTDA_MSG_MASK 0x7FFFFFFF + +/* HIPCTDD */ +#define CNL_DSP_REG_HIPCTDD_MSG_MASK 0x7FFFFFFF + +/* BDL */ +#define HDA_DSP_BDL_SIZE 4096 +#define HDA_DSP_MAX_BDL_ENTRIES \ + (HDA_DSP_BDL_SIZE / sizeof(struct sof_intel_dsp_bdl)) + +/* Number of DAIs */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) +#define SOF_SKL_NUM_DAIS 14 +#else +#define SOF_SKL_NUM_DAIS 8 +#endif + +/* Intel HD Audio SRAM Window 0*/ +#define HDA_ADSP_SRAM0_BASE_SKL 0x8000 + +/* Firmware status window */ +#define HDA_ADSP_FW_STATUS_SKL HDA_ADSP_SRAM0_BASE_SKL +#define HDA_ADSP_ERROR_CODE_SKL (HDA_ADSP_FW_STATUS_SKL + 0x4) + +#define HDA_IDISP_CODEC(x) ((x) & BIT(2)) + +struct sof_intel_dsp_bdl { + __le32 addr_l; + __le32 addr_h; + __le32 size; + __le32 ioc; +} __attribute((packed)); + +#define SOF_HDA_PLAYBACK_STREAMS 16 +#define SOF_HDA_CAPTURE_STREAMS 16 +#define SOF_HDA_PLAYBACK 0 +#define SOF_HDA_CAPTURE 1 + +/* represents DSP HDA controller frontend - i.e. host facing control */ +struct sof_intel_hda_dev { + + struct hda_bus hbus; + + /* hw config */ + const struct sof_intel_dsp_desc *desc; + + /* trace */ + struct hdac_ext_stream *dtrace_stream; + + /* if position update IPC needed */ + u32 no_ipc_position; + + int irq; + + /* DMIC device */ + struct platform_device *dmic_dev; +}; + +static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s) +{ + struct sof_intel_hda_dev *hda = s->pdata->hw_pdata; + + return &hda->hbus.core; +} + +static inline struct hda_bus *sof_to_hbus(struct snd_sof_dev *s) +{ + struct sof_intel_hda_dev *hda = s->pdata->hw_pdata; + + return &hda->hbus; +} + +struct sof_intel_hda_stream { + struct hdac_ext_stream hda_stream; + struct sof_intel_stream stream; +}; + +#define bus_to_sof_hda(bus) \ + container_of(bus, struct sof_intel_hda_dev, hbus.core) + +#define SOF_STREAM_SD_OFFSET(s) \ + (SOF_HDA_ADSP_SD_ENTRY_SIZE * ((s)->index) \ + + SOF_HDA_ADSP_LOADER_BASE) + +/* + * DSP Core services. + */ +int hda_dsp_probe(struct snd_sof_dev *sdev); +int hda_dsp_remove(struct snd_sof_dev *sdev); +int hda_dsp_core_reset_enter(struct snd_sof_dev *sdev, + unsigned int core_mask); +int hda_dsp_core_reset_leave(struct snd_sof_dev *sdev, + unsigned int core_mask); +int hda_dsp_core_stall_reset(struct snd_sof_dev *sdev, unsigned int core_mask); +int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask); +int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask); +int hda_dsp_enable_core(struct snd_sof_dev *sdev, unsigned int core_mask); +int hda_dsp_core_power_down(struct snd_sof_dev *sdev, unsigned int core_mask); +bool hda_dsp_core_is_enabled(struct snd_sof_dev *sdev, + unsigned int core_mask); +int hda_dsp_core_reset_power_down(struct snd_sof_dev *sdev, + unsigned int core_mask); +void hda_dsp_ipc_int_enable(struct snd_sof_dev *sdev); +void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev); + +int hda_dsp_suspend(struct snd_sof_dev *sdev, int state); +int hda_dsp_resume(struct snd_sof_dev *sdev); +int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev, int state); +int hda_dsp_runtime_resume(struct snd_sof_dev *sdev); +void hda_dsp_dump_skl(struct snd_sof_dev *sdev, u32 flags); +void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags); + +/* + * DSP PCM Operations. + */ +int hda_dsp_pcm_open(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream); +int hda_dsp_pcm_close(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream); +int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct sof_ipc_stream_params *ipc_params); +int hda_dsp_pcm_trigger(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, int cmd); +snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream); + +/* + * DSP Stream Operations. + */ + +int hda_dsp_stream_init(struct snd_sof_dev *sdev); +void hda_dsp_stream_free(struct snd_sof_dev *sdev); +int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, + struct hdac_ext_stream *stream, + struct snd_dma_buffer *dmab, + struct snd_pcm_hw_params *params); +int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, + struct hdac_ext_stream *stream, int cmd); +irqreturn_t hda_dsp_stream_interrupt(int irq, void *context); +irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context); +int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, + struct snd_dma_buffer *dmab, + struct hdac_stream *stream); + +struct hdac_ext_stream * + hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction); +int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag); +int hda_dsp_stream_spib_config(struct snd_sof_dev *sdev, + struct hdac_ext_stream *stream, + int enable, u32 size); + +void hda_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz); +int hda_ipc_pcm_params(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + const struct sof_ipc_pcm_params_reply *reply); + +/* + * DSP IPC Operations. + */ +int hda_dsp_ipc_send_msg(struct snd_sof_dev *sdev, + struct snd_sof_ipc_msg *msg); +void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev); +int hda_dsp_ipc_fw_ready(struct snd_sof_dev *sdev, u32 msg_id); +irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context); +irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context); +int hda_dsp_ipc_cmd_done(struct snd_sof_dev *sdev, int dir); + +/* + * DSP Code loader. + */ +int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev); +int hda_dsp_cl_boot_firmware_skl(struct snd_sof_dev *sdev); + +/* pre and post fw run ops */ +int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev); +int hda_dsp_post_fw_run(struct snd_sof_dev *sdev); + +/* + * HDA Controller Operations. + */ +int hda_dsp_ctrl_get_caps(struct snd_sof_dev *sdev); +void hda_dsp_ctrl_ppcap_enable(struct snd_sof_dev *sdev, bool enable); +void hda_dsp_ctrl_ppcap_int_enable(struct snd_sof_dev *sdev, bool enable); +int hda_dsp_ctrl_link_reset(struct snd_sof_dev *sdev, bool reset); +void hda_dsp_ctrl_misc_clock_gating(struct snd_sof_dev *sdev, bool enable); +int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable); +int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset); + +/* + * HDA bus operations. + */ +void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev, + const struct hdac_ext_bus_ops *ext_ops); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) +/* + * HDA Codec operations. + */ +int hda_codec_probe_bus(struct snd_sof_dev *sdev); + +#endif /* CONFIG_SND_SOC_SOF_HDA */ + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) && IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI) + +void hda_codec_i915_get(struct snd_sof_dev *sdev); +void hda_codec_i915_put(struct snd_sof_dev *sdev); +int hda_codec_i915_init(struct snd_sof_dev *sdev); +int hda_codec_i915_exit(struct snd_sof_dev *sdev); + +#else + +static inline void hda_codec_i915_get(struct snd_sof_dev *sdev) { } +static inline void hda_codec_i915_put(struct snd_sof_dev *sdev) { } +static inline int hda_codec_i915_init(struct snd_sof_dev *sdev) { return 0; } +static inline int hda_codec_i915_exit(struct snd_sof_dev *sdev) { return 0; } + +#endif /* CONFIG_SND_SOC_SOF_HDA && CONFIG_SND_SOC_HDAC_HDMI */ + +/* + * Trace Control. + */ +int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag); +int hda_dsp_trace_release(struct snd_sof_dev *sdev); +int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); + +/* common dai driver */ +extern struct snd_soc_dai_driver skl_dai[]; + +/* + * Platform Specific HW abstraction Ops. + */ +extern const struct snd_sof_dsp_ops sof_apl_ops; +extern const struct snd_sof_dsp_ops sof_cnl_ops; +extern const struct snd_sof_dsp_ops sof_skl_ops; + +extern const struct sof_intel_dsp_desc apl_chip_info; +extern const struct sof_intel_dsp_desc cnl_chip_info; +extern const struct sof_intel_dsp_desc skl_chip_info; + +#endif -- cgit v1.2.3-59-g8ed1b From 8a300c8fb17c7546aaf3f2203a2eeff32debfb8f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:48 -0500 Subject: ASoC: SOF: Intel: Add HDA controller for Intel DSP Support HDA controller operations for DSP and provide space for future DSP HDA FW integration. Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ctrl.c | 181 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 181 insertions(+) create mode 100644 sound/soc/sof/intel/hda-ctrl.c (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c new file mode 100644 index 000000000000..2c3645736e1f --- /dev/null +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -0,0 +1,181 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Liam Girdwood +// Ranjani Sridharan +// Rander Wang +// Keyon Jie +// + +/* + * Hardware interface for generic Intel audio DSP HDA IP + */ + +#include +#include +#include "../ops.h" +#include "hda.h" + +/* + * HDA Operations. + */ + +int hda_dsp_ctrl_link_reset(struct snd_sof_dev *sdev, bool reset) +{ + unsigned long timeout; + u32 gctl = 0; + u32 val; + + /* 0 to enter reset and 1 to exit reset */ + val = reset ? 0 : SOF_HDA_GCTL_RESET; + + /* enter/exit HDA controller reset */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_GCTL, + SOF_HDA_GCTL_RESET, val); + + /* wait to enter/exit reset */ + timeout = jiffies + msecs_to_jiffies(HDA_DSP_CTRL_RESET_TIMEOUT); + while (time_before(jiffies, timeout)) { + gctl = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_GCTL); + if ((gctl & SOF_HDA_GCTL_RESET) == val) + return 0; + usleep_range(500, 1000); + } + + /* enter/exit reset failed */ + dev_err(sdev->dev, "error: failed to %s HDA controller gctl 0x%x\n", + reset ? "reset" : "ready", gctl); + return -EIO; +} + +int hda_dsp_ctrl_get_caps(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + u32 cap, offset, feature; + int count = 0; + + offset = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_LLCH); + + do { + cap = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, offset); + + dev_dbg(sdev->dev, "checking for capabilities at offset 0x%x\n", + offset & SOF_HDA_CAP_NEXT_MASK); + + feature = (cap & SOF_HDA_CAP_ID_MASK) >> SOF_HDA_CAP_ID_OFF; + + switch (feature) { + case SOF_HDA_PP_CAP_ID: + dev_dbg(sdev->dev, "found DSP capability at 0x%x\n", + offset); + bus->ppcap = bus->remap_addr + offset; + sdev->bar[HDA_DSP_PP_BAR] = bus->ppcap; + break; + case SOF_HDA_SPIB_CAP_ID: + dev_dbg(sdev->dev, "found SPIB capability at 0x%x\n", + offset); + bus->spbcap = bus->remap_addr + offset; + sdev->bar[HDA_DSP_SPIB_BAR] = bus->spbcap; + break; + case SOF_HDA_DRSM_CAP_ID: + dev_dbg(sdev->dev, "found DRSM capability at 0x%x\n", + offset); + bus->drsmcap = bus->remap_addr + offset; + sdev->bar[HDA_DSP_DRSM_BAR] = bus->drsmcap; + break; + case SOF_HDA_GTS_CAP_ID: + dev_dbg(sdev->dev, "found GTS capability at 0x%x\n", + offset); + bus->gtscap = bus->remap_addr + offset; + break; + case SOF_HDA_ML_CAP_ID: + dev_dbg(sdev->dev, "found ML capability at 0x%x\n", + offset); + bus->mlcap = bus->remap_addr + offset; + break; + default: + dev_vdbg(sdev->dev, "found capability %d at 0x%x\n", + feature, offset); + break; + } + + offset = cap & SOF_HDA_CAP_NEXT_MASK; + } while (count++ <= SOF_HDA_MAX_CAPS && offset); + + return 0; +} + +void hda_dsp_ctrl_ppcap_enable(struct snd_sof_dev *sdev, bool enable) +{ + u32 val = enable ? SOF_HDA_PPCTL_GPROCEN : 0; + + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + SOF_HDA_PPCTL_GPROCEN, val); +} + +void hda_dsp_ctrl_ppcap_int_enable(struct snd_sof_dev *sdev, bool enable) +{ + u32 val = enable ? SOF_HDA_PPCTL_PIE : 0; + + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + SOF_HDA_PPCTL_PIE, val); +} + +void hda_dsp_ctrl_misc_clock_gating(struct snd_sof_dev *sdev, bool enable) +{ + u32 val = enable ? PCI_CGCTL_MISCBDCGE_MASK : 0; + + snd_sof_pci_update_bits(sdev, PCI_CGCTL, PCI_CGCTL_MISCBDCGE_MASK, val); +} + +/* + * enable/disable audio dsp clock gating and power gating bits. + * This allows the HW to opportunistically power and clock gate + * the audio dsp when it is idle + */ +int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable) +{ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + struct hdac_bus *bus = sof_to_bus(sdev); +#endif + u32 val; + + /* enable/disable audio dsp clock gating */ + val = enable ? PCI_CGCTL_ADSPDCGE : 0; + snd_sof_pci_update_bits(sdev, PCI_CGCTL, PCI_CGCTL_ADSPDCGE, val); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* enable/disable L1 support */ + val = enable ? SOF_HDA_VS_EM2_L1SEN : 0; + snd_hdac_chip_updatel(bus, VS_EM2, SOF_HDA_VS_EM2_L1SEN, val); +#endif + + /* enable/disable audio dsp power gating */ + val = enable ? 0 : PCI_PGCTL_ADSPPGD; + snd_sof_pci_update_bits(sdev, PCI_PGCTL, PCI_PGCTL_ADSPPGD, val); + + return 0; +} + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) +/* + * While performing reset, controller may not come back properly and causing + * issues, so recommendation is to set CGCTL.MISCBDCGE to 0 then do reset + * (init chip) and then again set CGCTL.MISCBDCGE to 1 + */ +int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + int ret; + + hda_dsp_ctrl_misc_clock_gating(sdev, false); + ret = snd_hdac_bus_init_chip(bus, full_reset); + hda_dsp_ctrl_misc_clock_gating(sdev, true); + + return ret; +} +#endif -- cgit v1.2.3-59-g8ed1b From 747503b1813a3e6e4c52d9c0b4bd462b64940940 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:49 -0500 Subject: ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations Add support for various PM and core reset/run state transitions. Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 455 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 455 insertions(+) create mode 100644 sound/soc/sof/intel/hda-dsp.c (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c new file mode 100644 index 000000000000..311fed502e09 --- /dev/null +++ b/sound/soc/sof/intel/hda-dsp.c @@ -0,0 +1,455 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Liam Girdwood +// Ranjani Sridharan +// Rander Wang +// Keyon Jie +// + +/* + * Hardware interface for generic Intel audio DSP HDA IP + */ + +#include +#include +#include "../ops.h" +#include "hda.h" + +/* + * DSP Core control. + */ + +int hda_dsp_core_reset_enter(struct snd_sof_dev *sdev, unsigned int core_mask) +{ + u32 adspcs; + u32 reset; + int ret; + + /* set reset bits for cores */ + reset = HDA_DSP_ADSPCS_CRST_MASK(core_mask); + snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS, + reset, reset), + + /* poll with timeout to check if operation successful */ + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS, adspcs, + ((adspcs & reset) == reset), + HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_RESET_TIMEOUT_US); + + /* has core entered reset ? */ + adspcs = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS); + if ((adspcs & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) != + HDA_DSP_ADSPCS_CRST_MASK(core_mask)) { + dev_err(sdev->dev, + "error: reset enter failed: core_mask %x adspcs 0x%x\n", + core_mask, adspcs); + ret = -EIO; + } + + return ret; +} + +int hda_dsp_core_reset_leave(struct snd_sof_dev *sdev, unsigned int core_mask) +{ + unsigned int crst; + u32 adspcs; + int ret; + + /* clear reset bits for cores */ + snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS, + HDA_DSP_ADSPCS_CRST_MASK(core_mask), + 0); + + /* poll with timeout to check if operation successful */ + crst = HDA_DSP_ADSPCS_CRST_MASK(core_mask); + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS, adspcs, + !(adspcs & crst), + HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_RESET_TIMEOUT_US); + + /* has core left reset ? */ + adspcs = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS); + if ((adspcs & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) != 0) { + dev_err(sdev->dev, + "error: reset leave failed: core_mask %x adspcs 0x%x\n", + core_mask, adspcs); + ret = -EIO; + } + + return ret; +} + +int hda_dsp_core_stall_reset(struct snd_sof_dev *sdev, unsigned int core_mask) +{ + /* stall core */ + snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS, + HDA_DSP_ADSPCS_CSTALL_MASK(core_mask), + HDA_DSP_ADSPCS_CSTALL_MASK(core_mask)); + + /* set reset state */ + return hda_dsp_core_reset_enter(sdev, core_mask); +} + +int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask) +{ + int ret; + + /* leave reset state */ + ret = hda_dsp_core_reset_leave(sdev, core_mask); + if (ret < 0) + return ret; + + /* run core */ + dev_dbg(sdev->dev, "unstall/run core: core_mask = %x\n", core_mask); + snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS, + HDA_DSP_ADSPCS_CSTALL_MASK(core_mask), + 0); + + /* is core now running ? */ + if (!hda_dsp_core_is_enabled(sdev, core_mask)) { + hda_dsp_core_stall_reset(sdev, core_mask); + dev_err(sdev->dev, "error: DSP start core failed: core_mask %x\n", + core_mask); + ret = -EIO; + } + + return ret; +} + +/* + * Power Management. + */ + +int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask) +{ + unsigned int cpa; + u32 adspcs; + int ret; + + /* update bits */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS, + HDA_DSP_ADSPCS_SPA_MASK(core_mask), + HDA_DSP_ADSPCS_SPA_MASK(core_mask)); + + /* poll with timeout to check if operation successful */ + cpa = HDA_DSP_ADSPCS_CPA_MASK(core_mask); + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS, adspcs, + (adspcs & cpa) == cpa, + HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_RESET_TIMEOUT_US); + if (ret < 0) + dev_err(sdev->dev, "error: timeout on core powerup\n"); + + /* did core power up ? */ + adspcs = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS); + if ((adspcs & HDA_DSP_ADSPCS_CPA_MASK(core_mask)) != + HDA_DSP_ADSPCS_CPA_MASK(core_mask)) { + dev_err(sdev->dev, + "error: power up core failed core_mask %xadspcs 0x%x\n", + core_mask, adspcs); + ret = -EIO; + } + + return ret; +} + +int hda_dsp_core_power_down(struct snd_sof_dev *sdev, unsigned int core_mask) +{ + u32 adspcs; + + /* update bits */ + snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS, + HDA_DSP_ADSPCS_SPA_MASK(core_mask), 0); + + return snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPCS, adspcs, + !(adspcs & HDA_DSP_ADSPCS_SPA_MASK(core_mask)), + HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_PD_TIMEOUT * USEC_PER_MSEC); +} + +bool hda_dsp_core_is_enabled(struct snd_sof_dev *sdev, + unsigned int core_mask) +{ + int val; + bool is_enable; + + val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS); + + is_enable = ((val & HDA_DSP_ADSPCS_CPA_MASK(core_mask)) && + (val & HDA_DSP_ADSPCS_SPA_MASK(core_mask)) && + !(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) && + !(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask))); + + dev_dbg(sdev->dev, "DSP core(s) enabled? %d : core_mask %x\n", + is_enable, core_mask); + + return is_enable; +} + +int hda_dsp_enable_core(struct snd_sof_dev *sdev, unsigned int core_mask) +{ + int ret; + + /* return if core is already enabled */ + if (hda_dsp_core_is_enabled(sdev, core_mask)) + return 0; + + /* power up */ + ret = hda_dsp_core_power_up(sdev, core_mask); + if (ret < 0) { + dev_err(sdev->dev, "error: dsp core power up failed: core_mask %x\n", + core_mask); + return ret; + } + + return hda_dsp_core_run(sdev, core_mask); +} + +int hda_dsp_core_reset_power_down(struct snd_sof_dev *sdev, + unsigned int core_mask) +{ + int ret; + + /* place core in reset prior to power down */ + ret = hda_dsp_core_stall_reset(sdev, core_mask); + if (ret < 0) { + dev_err(sdev->dev, "error: dsp core reset failed: core_mask %x\n", + core_mask); + return ret; + } + + /* power down core */ + ret = hda_dsp_core_power_down(sdev, core_mask); + if (ret < 0) { + dev_err(sdev->dev, "error: dsp core power down fail mask %x: %d\n", + core_mask, ret); + return ret; + } + + /* make sure we are in OFF state */ + if (hda_dsp_core_is_enabled(sdev, core_mask)) { + dev_err(sdev->dev, "error: dsp core disable fail mask %x: %d\n", + core_mask, ret); + ret = -EIO; + } + + return ret; +} + +void hda_dsp_ipc_int_enable(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; + + /* enable IPC DONE and BUSY interrupts */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, chip->ipc_ctl, + HDA_DSP_REG_HIPCCTL_DONE | HDA_DSP_REG_HIPCCTL_BUSY, + HDA_DSP_REG_HIPCCTL_DONE | HDA_DSP_REG_HIPCCTL_BUSY); + + /* enable IPC interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, + HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC); +} + +void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; + + /* disable IPC interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, + HDA_DSP_ADSPIC_IPC, 0); + + /* disable IPC BUSY and DONE interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, chip->ipc_ctl, + HDA_DSP_REG_HIPCCTL_BUSY | HDA_DSP_REG_HIPCCTL_DONE, 0); +} + +static int hda_suspend(struct snd_sof_dev *sdev, int state) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + struct hdac_bus *bus = sof_to_bus(sdev); +#endif + int ret; + + /* disable IPC interrupts */ + hda_dsp_ipc_int_disable(sdev); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* power down all hda link */ + snd_hdac_ext_bus_link_power_down_all(bus); +#endif + + /* power down DSP */ + ret = hda_dsp_core_reset_power_down(sdev, chip->cores_mask); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to power down core during suspend\n"); + return ret; + } + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* disable ppcap interrupt */ + snd_hdac_ext_bus_ppcap_int_enable(bus, false); + snd_hdac_ext_bus_ppcap_enable(bus, false); + + /* disable hda bus irq and i/o */ + snd_hdac_bus_stop_chip(bus); +#else + /* disable ppcap interrupt */ + hda_dsp_ctrl_ppcap_enable(sdev, false); + hda_dsp_ctrl_ppcap_int_enable(sdev, false); + + /* disable hda bus irq */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, + SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, + 0); +#endif + + /* disable LP retention mode */ + snd_sof_pci_update_bits(sdev, PCI_PGCTL, + PCI_PGCTL_LSRMD_MASK, PCI_PGCTL_LSRMD_MASK); + + /* reset controller */ + ret = hda_dsp_ctrl_link_reset(sdev, true); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to reset controller during suspend\n"); + return ret; + } + + return 0; +} + +static int hda_resume(struct snd_sof_dev *sdev) +{ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + struct hdac_bus *bus = sof_to_bus(sdev); + struct hdac_ext_link *hlink = NULL; +#endif + int ret; + + /* + * clear TCSEL to clear playback on some HD Audio + * codecs. PCI TCSEL is defined in the Intel manuals. + */ + snd_sof_pci_update_bits(sdev, PCI_TCSEL, 0x07, 0); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* reset and start hda controller */ + ret = hda_dsp_ctrl_init_chip(sdev, true); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to start controller after resume\n"); + return ret; + } + + hda_dsp_ctrl_misc_clock_gating(sdev, false); + + /* Reset stream-to-link mapping */ + list_for_each_entry(hlink, &bus->hlink_list, list) + bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); + + hda_dsp_ctrl_misc_clock_gating(sdev, true); + + /* enable ppcap interrupt */ + snd_hdac_ext_bus_ppcap_enable(bus, true); + snd_hdac_ext_bus_ppcap_int_enable(bus, true); +#else + + hda_dsp_ctrl_misc_clock_gating(sdev, false); + + /* reset controller */ + ret = hda_dsp_ctrl_link_reset(sdev, true); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to reset controller during resume\n"); + return ret; + } + + /* take controller out of reset */ + ret = hda_dsp_ctrl_link_reset(sdev, false); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to ready controller during resume\n"); + return ret; + } + + /* enable hda bus irq */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, + SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, + SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN); + + hda_dsp_ctrl_misc_clock_gating(sdev, true); + + /* enable ppcap interrupt */ + hda_dsp_ctrl_ppcap_enable(sdev, true); + hda_dsp_ctrl_ppcap_int_enable(sdev, true); +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* turn off the links that were off before suspend */ + list_for_each_entry(hlink, &bus->hlink_list, list) { + if (!hlink->ref_count) + snd_hdac_ext_bus_link_power_down(hlink); + } + + /* check dma status and clean up CORB/RIRB buffers */ + if (!bus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(bus); +#endif + + return 0; +} + +int hda_dsp_resume(struct snd_sof_dev *sdev) +{ + /* init hda controller. DSP cores will be powered up during fw boot */ + return hda_resume(sdev); +} + +int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) +{ + /* init hda controller. DSP cores will be powered up during fw boot */ + return hda_resume(sdev); +} + +int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev, int state) +{ + /* stop hda controller and power dsp off */ + return hda_suspend(sdev, state); +} + +int hda_dsp_suspend(struct snd_sof_dev *sdev, int state) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + int ret; + + /* stop hda controller and power dsp off */ + ret = hda_suspend(sdev, state); + if (ret < 0) { + dev_err(bus->dev, "error: suspending dsp\n"); + return ret; + } + + return 0; +} -- cgit v1.2.3-59-g8ed1b From 6e9cde974863dc9d9c6cdb178f625e410c5be3d0 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:50 -0500 Subject: ASoC: SOF: Intel: Add Intel specific HDA IPC mechanisms. Add HDA specific IPC mechanism for Intel DSP HW. Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ipc.c | 458 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 458 insertions(+) create mode 100644 sound/soc/sof/intel/hda-ipc.c (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c new file mode 100644 index 000000000000..a938f568dbb1 --- /dev/null +++ b/sound/soc/sof/intel/hda-ipc.c @@ -0,0 +1,458 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Liam Girdwood +// Ranjani Sridharan +// Rander Wang +// Keyon Jie +// + +/* + * Hardware interface for generic Intel audio DSP HDA IP + */ + +#include "../ops.h" +#include "hda.h" + +static void hda_dsp_ipc_host_done(struct snd_sof_dev *sdev) +{ + /* + * tell DSP cmd is done - clear busy + * interrupt and send reply msg to dsp + */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, + HDA_DSP_REG_HIPCT, + HDA_DSP_REG_HIPCT_BUSY, + HDA_DSP_REG_HIPCT_BUSY); + + /* unmask BUSY interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, + HDA_DSP_REG_HIPCCTL, + HDA_DSP_REG_HIPCCTL_BUSY, + HDA_DSP_REG_HIPCCTL_BUSY); +} + +static void hda_dsp_ipc_dsp_done(struct snd_sof_dev *sdev) +{ + /* + * set DONE bit - tell DSP we have received the reply msg + * from DSP, and processed it, don't send more reply to host + */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, + HDA_DSP_REG_HIPCIE, + HDA_DSP_REG_HIPCIE_DONE, + HDA_DSP_REG_HIPCIE_DONE); + + /* unmask Done interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, + HDA_DSP_REG_HIPCCTL, + HDA_DSP_REG_HIPCCTL_DONE, + HDA_DSP_REG_HIPCCTL_DONE); +} + +int hda_dsp_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) +{ + u32 cmd = msg->header; + + /* send IPC message to DSP */ + sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data, + msg->msg_size); + snd_sof_dsp_write(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCI, + cmd | HDA_DSP_REG_HIPCI_BUSY); + + return 0; +} + +void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) +{ + struct snd_sof_ipc_msg *msg = sdev->msg; + struct sof_ipc_reply reply; + struct sof_ipc_cmd_hdr *hdr; + unsigned long flags; + int ret = 0; + + /* + * Sometimes, there is unexpected reply ipc arriving. The reply + * ipc belongs to none of the ipcs sent from driver. + * In this case, the driver must ignore the ipc. + */ + if (!msg) { + dev_warn(sdev->dev, "unexpected ipc interrupt raised!\n"); + return; + } + spin_lock_irqsave(&sdev->ipc_lock, flags); + + hdr = msg->msg_data; + if (hdr->cmd == (SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CTX_SAVE)) { + /* + * memory windows are powered off before sending IPC reply, + * so we can't read the mailbox for CTX_SAVE reply. + */ + reply.error = 0; + reply.hdr.cmd = SOF_IPC_GLB_REPLY; + reply.hdr.size = sizeof(reply); + memcpy(msg->reply_data, &reply, sizeof(reply)); + goto out; + } + + /* get IPC reply from DSP in the mailbox */ + sof_mailbox_read(sdev, sdev->host_box.offset, &reply, + sizeof(reply)); + + if (reply.error < 0) { + memcpy(msg->reply_data, &reply, sizeof(reply)); + ret = reply.error; + } else { + /* reply correct size ? */ + if (reply.hdr.size != msg->reply_size) { + dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", + msg->reply_size, reply.hdr.size); + ret = -EINVAL; + } + + /* read the message */ + if (msg->reply_size > 0) + sof_mailbox_read(sdev, sdev->host_box.offset, + msg->reply_data, msg->reply_size); + } + +out: + msg->reply_error = ret; + + spin_unlock_irqrestore(&sdev->ipc_lock, flags); +} + +static bool hda_dsp_ipc_is_sof(uint32_t msg) +{ + return (msg & (HDA_DSP_IPC_PURGE_FW | 0xf << 9)) != msg || + (msg & HDA_DSP_IPC_PURGE_FW) != HDA_DSP_IPC_PURGE_FW; +} + +/* IPC handler thread */ +irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + irqreturn_t ret = IRQ_NONE; + u32 hipci; + u32 hipcie; + u32 hipct; + u32 hipcte; + u32 hipcctl; + u32 msg; + u32 msg_ext; + + /* here we handle IPC interrupts only */ + if (!(sdev->irq_status & HDA_DSP_ADSPIS_IPC)) + return IRQ_NONE; + + /* read IPC status */ + hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_REG_HIPCIE); + hipct = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCT); + hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCCTL); + + /* reenable IPC interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, + HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC); + + /* is this a reply message from the DSP */ + if (hipcie & HDA_DSP_REG_HIPCIE_DONE && + hipcctl & HDA_DSP_REG_HIPCCTL_DONE) { + hipci = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_REG_HIPCI); + msg = hipci & HDA_DSP_REG_HIPCI_MSG_MASK; + msg_ext = hipcie & HDA_DSP_REG_HIPCIE_MSG_MASK; + + dev_vdbg(sdev->dev, + "ipc: firmware response, msg:0x%x, msg_ext:0x%x\n", + msg, msg_ext); + + /* mask Done interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, + HDA_DSP_REG_HIPCCTL, + HDA_DSP_REG_HIPCCTL_DONE, 0); + + /* handle immediate reply from DSP core - ignore ROM messages */ + if (hda_dsp_ipc_is_sof(msg)) { + hda_dsp_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, msg); + } + + /* wake up sleeper if we are loading code */ + if (sdev->code_loading) { + sdev->code_loading = 0; + wake_up(&sdev->waitq); + } + + /* set the done bit */ + hda_dsp_ipc_dsp_done(sdev); + + ret = IRQ_HANDLED; + } + + /* is this a new message from DSP */ + if (hipct & HDA_DSP_REG_HIPCT_BUSY && + hipcctl & HDA_DSP_REG_HIPCCTL_BUSY) { + + hipcte = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_REG_HIPCTE); + msg = hipct & HDA_DSP_REG_HIPCT_MSG_MASK; + msg_ext = hipcte & HDA_DSP_REG_HIPCTE_MSG_MASK; + + dev_vdbg(sdev->dev, + "ipc: firmware initiated, msg:0x%x, msg_ext:0x%x\n", + msg, msg_ext); + + /* mask BUSY interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, + HDA_DSP_REG_HIPCCTL, + HDA_DSP_REG_HIPCCTL_BUSY, 0); + + /* handle messages from DSP */ + if ((hipct & SOF_IPC_PANIC_MAGIC_MASK) == SOF_IPC_PANIC_MAGIC) { + /* this is a PANIC message !! */ + snd_sof_dsp_panic(sdev, HDA_DSP_PANIC_OFFSET(msg_ext)); + } else { + /* normal message - process normally */ + snd_sof_ipc_msgs_rx(sdev); + } + + hda_dsp_ipc_host_done(sdev); + + ret = IRQ_HANDLED; + } + + return ret; +} + +/* is this IRQ for ADSP ? - we only care about IPC here */ +irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + int ret = IRQ_NONE; + + spin_lock(&sdev->hw_lock); + + /* store status */ + sdev->irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPIS); + + /* invalid message ? */ + if (sdev->irq_status == 0xffffffff) + goto out; + + /* IPC message ? */ + if (sdev->irq_status & HDA_DSP_ADSPIS_IPC) { + /* disable IPC interrupt */ + snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, + HDA_DSP_REG_ADSPIC, + HDA_DSP_ADSPIC_IPC, 0); + ret = IRQ_WAKE_THREAD; + } + +out: + spin_unlock(&sdev->hw_lock); + return ret; +} + +/* IPC Firmware ready */ + +static void ipc_get_windows(struct snd_sof_dev *sdev) +{ + struct sof_ipc_window_elem *elem; + u32 outbox_offset = 0; + u32 stream_offset = 0; + u32 inbox_offset = 0; + u32 outbox_size = 0; + u32 stream_size = 0; + u32 inbox_size = 0; + int i; + + if (!sdev->info_window) { + dev_err(sdev->dev, "error: have no window info\n"); + return; + } + + for (i = 0; i < sdev->info_window->num_windows; i++) { + elem = &sdev->info_window->window[i]; + + switch (elem->type) { + case SOF_IPC_REGION_UPBOX: + inbox_offset = + elem->offset + SRAM_WINDOW_OFFSET(elem->id); + inbox_size = elem->size; + snd_sof_debugfs_io_item(sdev, + sdev->bar[HDA_DSP_BAR] + + inbox_offset, + elem->size, "inbox", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_DOWNBOX: + outbox_offset = + elem->offset + SRAM_WINDOW_OFFSET(elem->id); + outbox_size = elem->size; + snd_sof_debugfs_io_item(sdev, + sdev->bar[HDA_DSP_BAR] + + outbox_offset, + elem->size, "outbox", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_TRACE: + snd_sof_debugfs_io_item(sdev, + sdev->bar[HDA_DSP_BAR] + + elem->offset + + SRAM_WINDOW_OFFSET + (elem->id), + elem->size, "etrace", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_DEBUG: + snd_sof_debugfs_io_item(sdev, + sdev->bar[HDA_DSP_BAR] + + elem->offset + + SRAM_WINDOW_OFFSET + (elem->id), + elem->size, "debug", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_STREAM: + stream_offset = + elem->offset + SRAM_WINDOW_OFFSET(elem->id); + stream_size = elem->size; + snd_sof_debugfs_io_item(sdev, + sdev->bar[HDA_DSP_BAR] + + elem->offset + + SRAM_WINDOW_OFFSET + (elem->id), + elem->size, "stream", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_REGS: + snd_sof_debugfs_io_item(sdev, + sdev->bar[HDA_DSP_BAR] + + elem->offset + + SRAM_WINDOW_OFFSET + (elem->id), + elem->size, "regs", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + case SOF_IPC_REGION_EXCEPTION: + sdev->dsp_oops_offset = elem->offset + + SRAM_WINDOW_OFFSET(elem->id); + snd_sof_debugfs_io_item(sdev, + sdev->bar[HDA_DSP_BAR] + + elem->offset + + SRAM_WINDOW_OFFSET + (elem->id), + elem->size, "exception", + SOF_DEBUGFS_ACCESS_D0_ONLY); + break; + default: + dev_err(sdev->dev, "error: get illegal window info\n"); + return; + } + } + + if (outbox_size == 0 || inbox_size == 0) { + dev_err(sdev->dev, "error: get illegal mailbox window\n"); + return; + } + + snd_sof_dsp_mailbox_init(sdev, inbox_offset, inbox_size, + outbox_offset, outbox_size); + sdev->stream_box.offset = stream_offset; + sdev->stream_box.size = stream_size; + + dev_dbg(sdev->dev, " mailbox upstream 0x%x - size 0x%x\n", + inbox_offset, inbox_size); + dev_dbg(sdev->dev, " mailbox downstream 0x%x - size 0x%x\n", + outbox_offset, outbox_size); + dev_dbg(sdev->dev, " stream region 0x%x - size 0x%x\n", + stream_offset, stream_size); +} + +/* check for ABI compatibility and create memory windows on first boot */ +int hda_dsp_ipc_fw_ready(struct snd_sof_dev *sdev, u32 msg_id) +{ + struct sof_ipc_fw_ready *fw_ready = &sdev->fw_ready; + u32 offset; + int ret; + + /* mailbox must be on 4k boundary */ + offset = HDA_DSP_MBOX_UPLINK_OFFSET; + + dev_dbg(sdev->dev, "ipc: DSP is ready 0x%8.8x offset 0x%x\n", + msg_id, offset); + + /* no need to re-check version/ABI for subsequent boots */ + if (!sdev->first_boot) + return 0; + + /* copy data from the DSP FW ready offset */ + sof_block_read(sdev, sdev->mmio_bar, offset, fw_ready, + sizeof(*fw_ready)); + + /* make sure ABI version is compatible */ + ret = snd_sof_ipc_valid(sdev); + if (ret < 0) + return ret; + + /* now check for extended data */ + snd_sof_fw_parse_ext_data(sdev, sdev->mmio_bar, + HDA_DSP_MBOX_UPLINK_OFFSET + + sizeof(struct sof_ipc_fw_ready)); + + ipc_get_windows(sdev); + + return 0; +} + +void hda_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz) +{ + if (!substream || !sdev->stream_box.size) { + sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz); + } else { + struct hdac_stream *hstream = substream->runtime->private_data; + struct sof_intel_hda_stream *hda_stream; + + hda_stream = container_of(hstream, + struct sof_intel_hda_stream, + hda_stream.hstream); + + /* The stream might already be closed */ + if (hstream) + sof_mailbox_read(sdev, hda_stream->stream.posn_offset, + p, sz); + } +} + +int hda_ipc_pcm_params(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + const struct sof_ipc_pcm_params_reply *reply) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct sof_intel_hda_stream *hda_stream; + /* validate offset */ + size_t posn_offset = reply->posn_offset; + + hda_stream = container_of(hstream, struct sof_intel_hda_stream, + hda_stream.hstream); + + /* check for unaligned offset or overflow */ + if (posn_offset > sdev->stream_box.size || + posn_offset % sizeof(struct sof_ipc_stream_posn) != 0) + return -EINVAL; + + hda_stream->stream.posn_offset = sdev->stream_box.offset + posn_offset; + + dev_dbg(sdev->dev, "pcm: stream dir %d, posn mailbox offset is %zu", + substream->stream, hda_stream->stream.posn_offset); + + return 0; +} -- cgit v1.2.3-59-g8ed1b From cd679898cd323d958b35c014f97c575b4abb8ce2 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:59 -0500 Subject: ASoC: SOF: Add ACPI device support Add support ACPI based SOF DSP devices. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/sof-acpi-dev.c | 312 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 312 insertions(+) create mode 100644 sound/soc/sof/sof-acpi-dev.c (limited to 'sound') diff --git a/sound/soc/sof/sof-acpi-dev.c b/sound/soc/sof/sof-acpi-dev.c new file mode 100644 index 000000000000..e9cf69874b5b --- /dev/null +++ b/sound/soc/sof/sof-acpi-dev.c @@ -0,0 +1,312 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +#include +#include +#include +#include +#include +#include +#include +#ifdef CONFIG_X86 +#include +#endif + +#include "ops.h" + +/* platform specific devices */ +#include "intel/shim.h" + +static char *fw_path; +module_param(fw_path, charp, 0444); +MODULE_PARM_DESC(fw_path, "alternate path for SOF firmware."); + +static char *tplg_path; +module_param(tplg_path, charp, 0444); +MODULE_PARM_DESC(tplg_path, "alternate path for SOF topology."); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HASWELL) +static const struct sof_dev_desc sof_acpi_haswell_desc = { + .machines = snd_soc_acpi_intel_haswell_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_imr_base = -1, + .irqindex_host_ipc = 0, + .chip_info = &hsw_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-hsw.ri", + .nocodec_tplg_filename = "sof-hsw-nocodec.tplg", + .ops = &sof_hsw_ops, + .arch_ops = &sof_xtensa_arch_ops +}; +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) +static const struct sof_dev_desc sof_acpi_broadwell_desc = { + .machines = snd_soc_acpi_intel_broadwell_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_imr_base = -1, + .irqindex_host_ipc = 0, + .chip_info = &bdw_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-bdw.ri", + .nocodec_tplg_filename = "sof-bdw-nocodec.tplg", + .ops = &sof_bdw_ops, + .arch_ops = &sof_xtensa_arch_ops +}; +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) + +/* BYTCR uses different IRQ index */ +static const struct sof_dev_desc sof_acpi_baytrailcr_desc = { + .machines = snd_soc_acpi_intel_baytrail_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_imr_base = 2, + .irqindex_host_ipc = 0, + .chip_info = &byt_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-byt.ri", + .nocodec_tplg_filename = "sof-byt-nocodec.tplg", + .ops = &sof_byt_ops, + .arch_ops = &sof_xtensa_arch_ops +}; + +static const struct sof_dev_desc sof_acpi_baytrail_desc = { + .machines = snd_soc_acpi_intel_baytrail_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_imr_base = 2, + .irqindex_host_ipc = 5, + .chip_info = &byt_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-byt.ri", + .nocodec_tplg_filename = "sof-byt-nocodec.tplg", + .ops = &sof_byt_ops, + .arch_ops = &sof_xtensa_arch_ops +}; + +#ifdef CONFIG_X86 /* TODO: move this to common helper */ + +static bool is_byt_cr(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + int status; + + if (iosf_mbi_available()) { + u32 bios_status; + status = iosf_mbi_read(BT_MBI_UNIT_PMC, /* 0x04 PUNIT */ + MBI_REG_READ, /* 0x10 */ + 0x006, /* BIOS_CONFIG */ + &bios_status); + + if (status) { + dev_err(dev, "could not read PUNIT BIOS_CONFIG\n"); + } else { + /* bits 26:27 mirror PMIC options */ + bios_status = (bios_status >> 26) & 3; + + if (bios_status == 1 || bios_status == 3) { + dev_info(dev, "Detected Baytrail-CR platform\n"); + return true; + } + + dev_info(dev, "BYT-CR not detected\n"); + } + } else { + dev_info(dev, "IOSF_MBI not available, no BYT-CR detection\n"); + } + + if (platform_get_resource(pdev, IORESOURCE_IRQ, 5) == NULL) { + /* + * Some devices detected as BYT-T have only a single IRQ listed, + * causing platform_get_irq with index 5 to return -ENXIO. + * The correct IRQ in this case is at index 0, as on BYT-CR. + */ + dev_info(dev, "Falling back to Baytrail-CR platform\n"); + return true; + } + + return false; +} +#else +static int is_byt_cr(struct platform_device *pdev) +{ + return 0; +} +#endif + +static const struct sof_dev_desc sof_acpi_cherrytrail_desc = { + .machines = snd_soc_acpi_intel_cherrytrail_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_imr_base = 2, + .irqindex_host_ipc = 5, + .chip_info = &cht_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-cht.ri", + .nocodec_tplg_filename = "sof-cht-nocodec.tplg", + .ops = &sof_cht_ops, + .arch_ops = &sof_xtensa_arch_ops +}; + +#endif + +static const struct dev_pm_ops sof_acpi_pm = { + SET_SYSTEM_SLEEP_PM_OPS(snd_sof_suspend, snd_sof_resume) + SET_RUNTIME_PM_OPS(snd_sof_runtime_suspend, snd_sof_runtime_resume, + NULL) +}; + +static void sof_acpi_probe_complete(struct device *dev) +{ + /* allow runtime_pm */ + pm_runtime_set_autosuspend_delay(dev, SND_SOF_SUSPEND_DELAY_MS); + pm_runtime_use_autosuspend(dev); + pm_runtime_enable(dev); +} + +static int sof_acpi_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + const struct sof_dev_desc *desc; + struct snd_soc_acpi_mach *mach; + struct snd_sof_pdata *sof_pdata; + const struct snd_sof_dsp_ops *ops; + int ret; + + dev_dbg(&pdev->dev, "ACPI DSP detected"); + + sof_pdata = devm_kzalloc(dev, sizeof(*sof_pdata), GFP_KERNEL); + if (!sof_pdata) + return -ENOMEM; + + desc = device_get_match_data(dev); + if (!desc) + return -ENODEV; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) + if (desc == &sof_acpi_baytrail_desc && is_byt_cr(pdev)) + desc = &sof_acpi_baytrailcr_desc; +#endif + + /* get ops for platform */ + ops = desc->ops; + if (!ops) { + dev_err(dev, "error: no matching ACPI descriptor ops\n"); + return -ENODEV; + } + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_FORCE_NOCODEC_MODE) + /* force nocodec mode */ + dev_warn(dev, "Force to use nocodec mode\n"); + mach = devm_kzalloc(dev, sizeof(*mach), GFP_KERNEL); + if (!mach) + return -ENOMEM; + ret = sof_nocodec_setup(dev, sof_pdata, mach, desc, ops); + if (ret < 0) + return ret; +#else + /* find machine */ + mach = snd_soc_acpi_find_machine(desc->machines); + if (!mach) { + dev_warn(dev, "warning: No matching ASoC machine driver found\n"); + } else { + sof_pdata->fw_filename = mach->sof_fw_filename; + sof_pdata->tplg_filename = mach->sof_tplg_filename; + } +#endif + + if (mach) { + mach->mach_params.platform = dev_name(dev); + mach->mach_params.acpi_ipc_irq_index = desc->irqindex_host_ipc; + } + + sof_pdata->machine = mach; + sof_pdata->desc = desc; + sof_pdata->dev = &pdev->dev; + sof_pdata->platform = dev_name(dev); + + /* alternate fw and tplg filenames ? */ + if (fw_path) + sof_pdata->fw_filename_prefix = fw_path; + else + sof_pdata->fw_filename_prefix = + sof_pdata->desc->default_fw_path; + + if (tplg_path) + sof_pdata->tplg_filename_prefix = tplg_path; + else + sof_pdata->tplg_filename_prefix = + sof_pdata->desc->default_tplg_path; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE) + /* set callback to enable runtime_pm */ + sof_pdata->sof_probe_complete = sof_acpi_probe_complete; +#endif + /* call sof helper for DSP hardware probe */ + ret = snd_sof_device_probe(dev, sof_pdata); + if (ret) { + dev_err(dev, "error: failed to probe DSP hardware!\n"); + return ret; + } + +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE) + sof_acpi_probe_complete(dev); +#endif + + return ret; +} + +static int sof_acpi_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + + /* call sof helper for DSP hardware remove */ + snd_sof_device_remove(&pdev->dev); + + return 0; +} + +static const struct acpi_device_id sof_acpi_match[] = { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HASWELL) + { "INT33C8", (unsigned long)&sof_acpi_haswell_desc }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) + { "INT3438", (unsigned long)&sof_acpi_broadwell_desc }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) + { "80860F28", (unsigned long)&sof_acpi_baytrail_desc }, + { "808622A8", (unsigned long)&sof_acpi_cherrytrail_desc }, +#endif + { } +}; +MODULE_DEVICE_TABLE(acpi, sof_acpi_match); + +/* acpi_driver definition */ +static struct platform_driver snd_sof_acpi_driver = { + .probe = sof_acpi_probe, + .remove = sof_acpi_remove, + .driver = { + .name = "sof-audio-acpi", + .pm = &sof_acpi_pm, + .acpi_match_table = ACPI_PTR(sof_acpi_match), + }, +}; +module_platform_driver(snd_sof_acpi_driver); + +MODULE_LICENSE("Dual BSD/GPL"); -- cgit v1.2.3-59-g8ed1b From a226893b85b3c35f6c7c168beb28de738211164a Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:09:00 -0500 Subject: ASoC: SOF: Add PCI device support Add support for PCI based DSP devices. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/sof-pci-dev.c | 373 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 373 insertions(+) create mode 100644 sound/soc/sof/sof-pci-dev.c (limited to 'sound') diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c new file mode 100644 index 000000000000..b778dffb2d25 --- /dev/null +++ b/sound/soc/sof/sof-pci-dev.c @@ -0,0 +1,373 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Author: Liam Girdwood +// + +#include +#include +#include +#include +#include +#include +#include +#include "ops.h" + +/* platform specific devices */ +#include "intel/shim.h" +#include "intel/hda.h" + +static char *fw_path; +module_param(fw_path, charp, 0444); +MODULE_PARM_DESC(fw_path, "alternate path for SOF firmware."); + +static char *tplg_path; +module_param(tplg_path, charp, 0444); +MODULE_PARM_DESC(tplg_path, "alternate path for SOF topology."); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) +static const struct sof_dev_desc bxt_desc = { + .machines = snd_soc_acpi_intel_bxt_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &apl_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-apl.ri", + .nocodec_tplg_filename = "sof-apl-nocodec.tplg", + .ops = &sof_apl_ops, + .arch_ops = &sof_xtensa_arch_ops +}; +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) +static const struct sof_dev_desc glk_desc = { + .machines = snd_soc_acpi_intel_glk_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &apl_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-glk.ri", + .nocodec_tplg_filename = "sof-glk-nocodec.tplg", + .ops = &sof_apl_ops, + .arch_ops = &sof_xtensa_arch_ops +}; +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) +static struct snd_soc_acpi_mach sof_tng_machines[] = { + { + .id = "INT343A", + .drv_name = "edison", + .sof_fw_filename = "sof-byt.ri", + .sof_tplg_filename = "sof-byt.tplg", + }, + {} +}; + +static const struct sof_dev_desc tng_desc = { + .machines = sof_tng_machines, + .resindex_lpe_base = 3, /* IRAM, but subtract IRAM offset */ + .resindex_pcicfg_base = -1, + .resindex_imr_base = 0, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &tng_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-byt.ri", + .nocodec_tplg_filename = "sof-byt.tplg", + .ops = &sof_tng_ops, + .arch_ops = &sof_xtensa_arch_ops +}; +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) +static const struct sof_dev_desc cnl_desc = { + .machines = snd_soc_acpi_intel_cnl_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &cnl_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-cnl.ri", + .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", + .ops = &sof_cnl_ops, + .arch_ops = &sof_xtensa_arch_ops +}; +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) +static const struct sof_dev_desc cfl_desc = { + .machines = snd_soc_acpi_intel_cnl_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &cnl_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-cnl.ri", + .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", + .ops = &sof_cnl_ops, + .arch_ops = &sof_xtensa_arch_ops +}; +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) +static const struct sof_dev_desc icl_desc = { + .machines = snd_soc_acpi_intel_icl_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &cnl_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-icl.ri", + .nocodec_tplg_filename = "sof-icl-nocodec.tplg", + .ops = &sof_cnl_ops, + .arch_ops = &sof_xtensa_arch_ops +}; +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) +static const struct sof_dev_desc skl_desc = { + .machines = snd_soc_acpi_intel_skl_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &skl_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-skl.ri", + .nocodec_tplg_filename = "sof-skl-nocodec.tplg", + .ops = &sof_skl_ops, + .arch_ops = &sof_xtensa_arch_ops +}; +#endif + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) +static const struct sof_dev_desc kbl_desc = { + .machines = snd_soc_acpi_intel_kbl_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &skl_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .nocodec_fw_filename = "sof-kbl.ri", + .nocodec_tplg_filename = "sof-kbl-nocodec.tplg", + .ops = &sof_skl_ops, + .arch_ops = &sof_xtensa_arch_ops +}; +#endif + +static const struct dev_pm_ops sof_pci_pm = { + SET_SYSTEM_SLEEP_PM_OPS(snd_sof_suspend, snd_sof_resume) + SET_RUNTIME_PM_OPS(snd_sof_runtime_suspend, snd_sof_runtime_resume, + NULL) +}; + +static void sof_pci_probe_complete(struct device *dev) +{ + dev_dbg(dev, "Completing SOF PCI probe"); + + /* allow runtime_pm */ + pm_runtime_set_autosuspend_delay(dev, SND_SOF_SUSPEND_DELAY_MS); + pm_runtime_use_autosuspend(dev); + + /* + * runtime pm for pci device is "forbidden" by default. + * so call pm_runtime_allow() to enable it. + */ + pm_runtime_allow(dev); + + /* follow recommendation in pci-driver.c to decrement usage counter */ + pm_runtime_put_noidle(dev); +} + +static int sof_pci_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct device *dev = &pci->dev; + const struct sof_dev_desc *desc = + (const struct sof_dev_desc *)pci_id->driver_data; + struct snd_soc_acpi_mach *mach; + struct snd_sof_pdata *sof_pdata; + const struct snd_sof_dsp_ops *ops; + int ret; + + dev_dbg(&pci->dev, "PCI DSP detected"); + + /* get ops for platform */ + ops = desc->ops; + if (!ops) { + dev_err(dev, "error: no matching PCI descriptor ops\n"); + return -ENODEV; + } + + sof_pdata = devm_kzalloc(dev, sizeof(*sof_pdata), GFP_KERNEL); + if (!sof_pdata) + return -ENOMEM; + + ret = pcim_enable_device(pci); + if (ret < 0) + return ret; + + ret = pci_request_regions(pci, "Audio DSP"); + if (ret < 0) + return ret; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_FORCE_NOCODEC_MODE) + /* force nocodec mode */ + dev_warn(dev, "Force to use nocodec mode\n"); + mach = devm_kzalloc(dev, sizeof(*mach), GFP_KERNEL); + if (!mach) { + ret = -ENOMEM; + goto release_regions; + } + ret = sof_nocodec_setup(dev, sof_pdata, mach, desc, ops); + if (ret < 0) + goto release_regions; + +#else + /* find machine */ + mach = snd_soc_acpi_find_machine(desc->machines); + if (!mach) { + dev_warn(dev, "warning: No matching ASoC machine driver found\n"); + } else { + mach->mach_params.platform = dev_name(dev); + sof_pdata->fw_filename = mach->sof_fw_filename; + sof_pdata->tplg_filename = mach->sof_tplg_filename; + } +#endif /* CONFIG_SND_SOC_SOF_FORCE_NOCODEC_MODE */ + + sof_pdata->name = pci_name(pci); + sof_pdata->machine = mach; + sof_pdata->desc = (struct sof_dev_desc *)pci_id->driver_data; + sof_pdata->dev = dev; + sof_pdata->platform = dev_name(dev); + + /* alternate fw and tplg filenames ? */ + if (fw_path) + sof_pdata->fw_filename_prefix = fw_path; + else + sof_pdata->fw_filename_prefix = + sof_pdata->desc->default_fw_path; + + if (tplg_path) + sof_pdata->tplg_filename_prefix = tplg_path; + else + sof_pdata->tplg_filename_prefix = + sof_pdata->desc->default_tplg_path; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE) + /* set callback to enable runtime_pm */ + sof_pdata->sof_probe_complete = sof_pci_probe_complete; +#endif + /* call sof helper for DSP hardware probe */ + ret = snd_sof_device_probe(dev, sof_pdata); + if (ret) { + dev_err(dev, "error: failed to probe DSP hardware!\n"); + goto release_regions; + } + +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE) + sof_pci_probe_complete(dev); +#endif + + return ret; + +release_regions: + pci_release_regions(pci); + + return ret; +} + +static void sof_pci_remove(struct pci_dev *pci) +{ + /* call sof helper for DSP hardware remove */ + snd_sof_device_remove(&pci->dev); + + /* follow recommendation in pci-driver.c to increment usage counter */ + pm_runtime_get_noresume(&pci->dev); + + /* release pci regions and disable device */ + pci_release_regions(pci); +} + +/* PCI IDs */ +static const struct pci_device_id sof_pci_ids[] = { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) + { PCI_DEVICE(0x8086, 0x119a), + .driver_data = (unsigned long)&tng_desc}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + /* BXT-P & Apollolake */ + { PCI_DEVICE(0x8086, 0x5a98), + .driver_data = (unsigned long)&bxt_desc}, + { PCI_DEVICE(0x8086, 0x1a98), + .driver_data = (unsigned long)&bxt_desc}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + { PCI_DEVICE(0x8086, 0x3198), + .driver_data = (unsigned long)&glk_desc}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + { PCI_DEVICE(0x8086, 0x9dc8), + .driver_data = (unsigned long)&cnl_desc}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + { PCI_DEVICE(0x8086, 0xa348), + .driver_data = (unsigned long)&cfl_desc}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) + { PCI_DEVICE(0x8086, 0x9d71), + .driver_data = (unsigned long)&kbl_desc}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) + { PCI_DEVICE(0x8086, 0x9d70), + .driver_data = (unsigned long)&skl_desc}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) + { PCI_DEVICE(0x8086, 0x34C8), + .driver_data = (unsigned long)&icl_desc}, +#endif + { 0, } +}; +MODULE_DEVICE_TABLE(pci, sof_pci_ids); + +/* pci_driver definition */ +static struct pci_driver snd_sof_pci_driver = { + .name = "sof-audio-pci", + .id_table = sof_pci_ids, + .probe = sof_pci_probe, + .remove = sof_pci_remove, + .driver = { + .pm = &sof_pci_pm, + }, +}; +module_pci_driver(snd_sof_pci_driver); + +MODULE_LICENSE("Dual BSD/GPL"); -- cgit v1.2.3-59-g8ed1b From d16046ffa6de040bf580a64d5f4d0aa18258a854 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:51 -0500 Subject: ASoC: SOF: Intel: Add Intel specific HDA firmware loader Add support for loading DSP firmware on Intel HDA based platforms. Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 371 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 371 insertions(+) create mode 100644 sound/soc/sof/intel/hda-loader.c (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c new file mode 100644 index 000000000000..6a44bc349e44 --- /dev/null +++ b/sound/soc/sof/intel/hda-loader.c @@ -0,0 +1,371 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Liam Girdwood +// Ranjani Sridharan +// Rander Wang +// Keyon Jie +// + +/* + * Hardware interface for HDA DSP code loader + */ + +#include +#include +#include +#include "../ops.h" +#include "hda.h" + +#define HDA_FW_BOOT_ATTEMPTS 3 + +static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, + unsigned int size, struct snd_dma_buffer *dmab, + int direction) +{ + struct hdac_ext_stream *dsp_stream; + struct hdac_stream *hstream; + struct pci_dev *pci = to_pci_dev(sdev->dev); + int ret; + + if (direction != SNDRV_PCM_STREAM_PLAYBACK) { + dev_err(sdev->dev, "error: code loading DMA is playback only\n"); + return -EINVAL; + } + + dsp_stream = hda_dsp_stream_get(sdev, direction); + + if (!dsp_stream) { + dev_err(sdev->dev, "error: no stream available\n"); + return -ENODEV; + } + hstream = &dsp_stream->hstream; + + /* allocate DMA buffer */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab); + if (ret < 0) { + dev_err(sdev->dev, "error: memory alloc failed: %x\n", ret); + goto error; + } + + hstream->period_bytes = 0;/* initialize period_bytes */ + hstream->format_val = format; + hstream->bufsize = size; + + ret = hda_dsp_stream_hw_params(sdev, dsp_stream, dmab, NULL); + if (ret < 0) { + dev_err(sdev->dev, "error: hdac prepare failed: %x\n", ret); + goto error; + } + + hda_dsp_stream_spib_config(sdev, dsp_stream, HDA_DSP_SPIB_ENABLE, size); + + return hstream->stream_tag; + +error: + hda_dsp_stream_put(sdev, direction, hstream->stream_tag); + snd_dma_free_pages(dmab); + return ret; +} + +/* + * first boot sequence has some extra steps. core 0 waits for power + * status on core 1, so power up core 1 also momentarily, keep it in + * reset/stall and then turn it off + */ +static int cl_dsp_init(struct snd_sof_dev *sdev, const void *fwdata, + u32 fwsize, int stream_tag) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; + unsigned int status; + int ret; + + /* step 1: power up corex */ + ret = hda_dsp_core_power_up(sdev, chip->cores_mask); + if (ret < 0) { + dev_err(sdev->dev, "error: dsp core 0/1 power up failed\n"); + goto err; + } + + /* step 2: purge FW request */ + snd_sof_dsp_write(sdev, HDA_DSP_BAR, chip->ipc_req, + chip->ipc_req_mask | (HDA_DSP_IPC_PURGE_FW | + ((stream_tag - 1) << 9))); + + /* step 3: unset core 0 reset state & unstall/run core 0 */ + ret = hda_dsp_core_run(sdev, HDA_DSP_CORE_MASK(0)); + if (ret < 0) { + dev_err(sdev->dev, "error: dsp core start failed %d\n", ret); + ret = -EIO; + goto err; + } + + /* step 4: wait for IPC DONE bit from ROM */ + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, + chip->ipc_ack, status, + ((status & chip->ipc_ack_mask) + == chip->ipc_ack_mask), + HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_INIT_TIMEOUT_US); + + if (ret < 0) { + dev_err(sdev->dev, "error: waiting for HIPCIE done\n"); + goto err; + } + + /* step 5: power down corex */ + ret = hda_dsp_core_power_down(sdev, + chip->cores_mask & ~(HDA_DSP_CORE_MASK(0))); + if (ret < 0) { + dev_err(sdev->dev, "error: dsp core x power down failed\n"); + goto err; + } + + /* step 6: enable IPC interrupts */ + hda_dsp_ipc_int_enable(sdev); + + /* step 7: wait for ROM init */ + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, + HDA_DSP_SRAM_REG_ROM_STATUS, status, + ((status & HDA_DSP_ROM_STS_MASK) + == HDA_DSP_ROM_INIT), + HDA_DSP_REG_POLL_INTERVAL_US, + chip->rom_init_timeout * + USEC_PER_MSEC); + if (!ret) + return 0; + +err: + hda_dsp_dump(sdev, SOF_DBG_REGS | SOF_DBG_PCI | SOF_DBG_MBOX); + hda_dsp_core_reset_power_down(sdev, chip->cores_mask); + + return ret; +} + +static int cl_trigger(struct snd_sof_dev *sdev, + struct hdac_ext_stream *stream, int cmd) +{ + struct hdac_stream *hstream = &stream->hstream; + int sd_offset = SOF_STREAM_SD_OFFSET(hstream); + + /* code loader is special case that reuses stream ops */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + wait_event_timeout(sdev->waitq, !sdev->code_loading, + HDA_DSP_CL_TRIGGER_TIMEOUT); + + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, + 1 << hstream->index, + 1 << hstream->index); + + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + sd_offset, + SOF_HDA_SD_CTL_DMA_START | + SOF_HDA_CL_DMA_SD_INT_MASK, + SOF_HDA_SD_CTL_DMA_START | + SOF_HDA_CL_DMA_SD_INT_MASK); + + hstream->running = true; + return 0; + default: + return hda_dsp_stream_trigger(sdev, stream, cmd); + } +} + +static struct hdac_ext_stream *get_stream_with_tag(struct snd_sof_dev *sdev, + int tag) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct hdac_stream *s; + + /* get stream with tag */ + list_for_each_entry(s, &bus->stream_list, list) { + if (s->direction == SNDRV_PCM_STREAM_PLAYBACK && + s->stream_tag == tag) { + return stream_to_hdac_ext_stream(s); + } + } + + return NULL; +} + +static int cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, + struct hdac_ext_stream *stream) +{ + struct hdac_stream *hstream = &stream->hstream; + int sd_offset = SOF_STREAM_SD_OFFSET(hstream); + int ret; + + ret = hda_dsp_stream_spib_config(sdev, stream, HDA_DSP_SPIB_DISABLE, 0); + + hda_dsp_stream_put(sdev, SNDRV_PCM_STREAM_PLAYBACK, + hstream->stream_tag); + hstream->running = 0; + hstream->substream = NULL; + + /* reset BDL address */ + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_BDLPL, 0); + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_BDLPU, 0); + + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, sd_offset, 0); + snd_dma_free_pages(dmab); + dmab->area = NULL; + hstream->bufsize = 0; + hstream->format_val = 0; + + return ret; +} + +static int cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *stream) +{ + unsigned int reg; + int ret, status; + + ret = cl_trigger(sdev, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) { + dev_err(sdev->dev, "error: DMA trigger start failed\n"); + return ret; + } + + status = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, + HDA_DSP_SRAM_REG_ROM_STATUS, reg, + ((reg & HDA_DSP_ROM_STS_MASK) + == HDA_DSP_ROM_FW_ENTERED), + HDA_DSP_REG_POLL_INTERVAL_US, + HDA_DSP_BASEFW_TIMEOUT_US); + + ret = cl_trigger(sdev, stream, SNDRV_PCM_TRIGGER_STOP); + if (ret < 0) { + dev_err(sdev->dev, "error: DMA trigger stop failed\n"); + return ret; + } + + return status; +} + +int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *plat_data = sdev->pdata; + const struct sof_dev_desc *desc = plat_data->desc; + const struct sof_intel_dsp_desc *chip_info; + struct hdac_ext_stream *stream; + struct firmware stripped_firmware; + int ret, ret1, tag, i; + + chip_info = desc->chip_info; + + stripped_firmware.data = plat_data->fw->data; + stripped_firmware.size = plat_data->fw->size; + + /* init for booting wait */ + init_waitqueue_head(&sdev->boot_wait); + sdev->boot_complete = false; + + /* prepare DMA for code loader stream */ + tag = cl_stream_prepare(sdev, 0x40, stripped_firmware.size, + &sdev->dmab, SNDRV_PCM_STREAM_PLAYBACK); + + if (tag < 0) { + dev_err(sdev->dev, "error: dma prepare for fw loading err: %x\n", + tag); + return tag; + } + + /* get stream with tag */ + stream = get_stream_with_tag(sdev, tag); + if (!stream) { + dev_err(sdev->dev, + "error: could not get stream with stream tag %d\n", + tag); + ret = -ENODEV; + goto err; + } + + memcpy(sdev->dmab.area, stripped_firmware.data, + stripped_firmware.size); + + /* try ROM init a few times before giving up */ + for (i = 0; i < HDA_FW_BOOT_ATTEMPTS; i++) { + ret = cl_dsp_init(sdev, stripped_firmware.data, + stripped_firmware.size, tag); + + /* don't retry anymore if successful */ + if (!ret) + break; + + dev_err(sdev->dev, "error: Error code=0x%x: FW status=0x%x\n", + snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_SRAM_REG_ROM_ERROR), + snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_SRAM_REG_ROM_STATUS)); + dev_err(sdev->dev, "error: iteration %d of Core En/ROM load failed: %d\n", + i, ret); + } + + if (i == HDA_FW_BOOT_ATTEMPTS) { + dev_err(sdev->dev, "error: dsp init failed after %d attempts with err: %d\n", + i, ret); + goto cleanup; + } + + /* + * at this point DSP ROM has been initialized and + * should be ready for code loading and firmware boot + */ + ret = cl_copy_fw(sdev, stream); + if (!ret) + dev_dbg(sdev->dev, "Firmware download successful, booting...\n"); + else + dev_err(sdev->dev, "error: load fw failed ret: %d\n", ret); + +cleanup: + /* + * Perform codeloader stream cleanup. + * This should be done even if firmware loading fails. + */ + ret1 = cl_cleanup(sdev, &sdev->dmab, stream); + if (ret1 < 0) { + dev_err(sdev->dev, "error: Code loader DSP cleanup failed\n"); + + /* set return value to indicate cleanup failure */ + ret = ret1; + } + + /* + * return master core id if both fw copy + * and stream clean up are successful + */ + if (!ret) + return chip_info->init_core_mask; + + /* dump dsp registers and disable DSP upon error */ +err: + hda_dsp_dump(sdev, SOF_DBG_REGS | SOF_DBG_PCI | SOF_DBG_MBOX); + + /* disable DSP */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, + SOF_HDA_REG_PP_PPCTL, + SOF_HDA_PPCTL_GPROCEN, 0); + return ret; +} + +/* pre fw run operations */ +int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev) +{ + /* disable clock gating and power gating */ + return hda_dsp_ctrl_clock_power_gating(sdev, false); +} + +/* post fw run operations */ +int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) +{ + /* re-enable clock gating and power gating */ + return hda_dsp_ctrl_clock_power_gating(sdev, true); +} -- cgit v1.2.3-59-g8ed1b From c6be710f5e2a92204f92d872a0c0b4fe29666492 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:52 -0500 Subject: ASoC: SOF: Intel: Add Intel specific HDA PCM operations Add PCM operations for Intel HDA based DSPs. Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-pcm.c | 240 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 240 insertions(+) create mode 100644 sound/soc/sof/intel/hda-pcm.c (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c new file mode 100644 index 000000000000..5714a79fbe1a --- /dev/null +++ b/sound/soc/sof/intel/hda-pcm.c @@ -0,0 +1,240 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Liam Girdwood +// Ranjani Sridharan +// Rander Wang +// Keyon Jie +// + +/* + * Hardware interface for generic Intel audio DSP HDA IP + */ + +#include +#include +#include +#include "../ops.h" +#include "hda.h" + +#define SDnFMT_BASE(x) ((x) << 14) +#define SDnFMT_MULT(x) (((x) - 1) << 11) +#define SDnFMT_DIV(x) (((x) - 1) << 8) +#define SDnFMT_BITS(x) ((x) << 4) +#define SDnFMT_CHAN(x) ((x) << 0) + +static inline u32 get_mult_div(struct snd_sof_dev *sdev, int rate) +{ + switch (rate) { + case 8000: + return SDnFMT_DIV(6); + case 9600: + return SDnFMT_DIV(5); + case 11025: + return SDnFMT_BASE(1) | SDnFMT_DIV(4); + case 16000: + return SDnFMT_DIV(3); + case 22050: + return SDnFMT_BASE(1) | SDnFMT_DIV(2); + case 32000: + return SDnFMT_DIV(3) | SDnFMT_MULT(2); + case 44100: + return SDnFMT_BASE(1); + case 48000: + return 0; + case 88200: + return SDnFMT_BASE(1) | SDnFMT_MULT(2); + case 96000: + return SDnFMT_MULT(2); + case 176400: + return SDnFMT_BASE(1) | SDnFMT_MULT(4); + case 192000: + return SDnFMT_MULT(4); + default: + dev_warn(sdev->dev, "can't find div rate %d using 48kHz\n", + rate); + return 0; /* use 48KHz if not found */ + } +}; + +static inline u32 get_bits(struct snd_sof_dev *sdev, int sample_bits) +{ + switch (sample_bits) { + case 8: + return SDnFMT_BITS(0); + case 16: + return SDnFMT_BITS(1); + case 20: + return SDnFMT_BITS(2); + case 24: + return SDnFMT_BITS(3); + case 32: + return SDnFMT_BITS(4); + default: + dev_warn(sdev->dev, "can't find %d bits using 16bit\n", + sample_bits); + return SDnFMT_BITS(1); /* use 16bits format if not found */ + } +}; + +int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct sof_ipc_stream_params *ipc_params) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *stream = stream_to_hdac_ext_stream(hstream); + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + struct snd_dma_buffer *dmab; + int ret; + u32 size, rate, bits; + + size = params_buffer_bytes(params); + rate = get_mult_div(sdev, params_rate(params)); + bits = get_bits(sdev, params_width(params)); + + hstream->substream = substream; + + dmab = substream->runtime->dma_buffer_p; + + hstream->format_val = rate | bits | (params_channels(params) - 1); + hstream->bufsize = size; + hstream->period_bytes = params_period_bytes(params); + hstream->no_period_wakeup = + (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && + (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); + + ret = hda_dsp_stream_hw_params(sdev, stream, dmab, params); + if (ret < 0) { + dev_err(sdev->dev, "error: hdac prepare failed: %x\n", ret); + return ret; + } + + /* disable SPIB, to enable buffer wrap for stream */ + hda_dsp_stream_spib_config(sdev, stream, HDA_DSP_SPIB_DISABLE, 0); + + /* set host_period_bytes to 0 if no IPC position */ + if (hda && hda->no_ipc_position) + ipc_params->host_period_bytes = 0; + + ipc_params->stream_tag = hstream->stream_tag; + + return 0; +} + +int hda_dsp_pcm_trigger(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, int cmd) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *stream = stream_to_hdac_ext_stream(hstream); + + return hda_dsp_stream_trigger(sdev, stream, cmd); +} + +snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hdac_stream *hstream = substream->runtime->private_data; + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + struct snd_sof_pcm *spcm; + snd_pcm_uframes_t pos; + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) { + dev_warn_ratelimited(sdev->dev, "warn: can't find PCM with DAI ID %d\n", + rtd->dai_link->id); + return 0; + } + + if (hda && !hda->no_ipc_position) { + /* read position from IPC position */ + pos = spcm->stream[substream->stream].posn.host_posn; + goto found; + } + + /* + * DPIB/posbuf position mode: + * For Playback, Use DPIB register from HDA space which + * reflects the actual data transferred. + * For Capture, Use the position buffer for pointer, as DPIB + * is not accurate enough, its update may be completed + * earlier than the data written to DDR. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + } else { + /* + * For capture stream, we need more workaround to fix the + * position incorrect issue: + * + * 1. Wait at least 20us before reading position buffer after + * the interrupt generated(IOC), to make sure position update + * happens on frame boundary i.e. 20.833uSec for 48KHz. + * 2. Perform a dummy Read to DPIB register to flush DMA + * position value. + * 3. Read the DMA Position from posbuf. Now the readback + * value should be >= period boundary. + */ + usleep_range(20, 21); + snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + pos = snd_hdac_stream_get_pos_posbuf(hstream); + } + + if (pos >= hstream->bufsize) + pos = 0; + +found: + pos = bytes_to_frames(substream->runtime, pos); + + dev_vdbg(sdev->dev, "PCM: stream %d dir %d position %lu\n", + hstream->index, substream->stream, pos); + return pos; +} + +int hda_dsp_pcm_open(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream) +{ + struct hdac_ext_stream *dsp_stream; + int direction = substream->stream; + + dsp_stream = hda_dsp_stream_get(sdev, direction); + + if (!dsp_stream) { + dev_err(sdev->dev, "error: no stream available\n"); + return -ENODEV; + } + + /* binding pcm substream to hda stream */ + substream->runtime->private_data = &dsp_stream->hstream; + return 0; +} + +int hda_dsp_pcm_close(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + int direction = substream->stream; + int ret; + + ret = hda_dsp_stream_put(sdev, direction, hstream->stream_tag); + + if (ret) { + dev_dbg(sdev->dev, "stream %s not opened!\n", substream->name); + return -ENODEV; + } + + /* unbinding pcm substream to hda stream */ + substream->runtime->private_data = NULL; + return 0; +} -- cgit v1.2.3-59-g8ed1b From 78ad1f07a49a56fa3714ee36573133a39e32593d Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Fri, 12 Apr 2019 11:08:53 -0500 Subject: ASoC: SOF: Intel: Add hda-bus support and initialization Use hdac_io_ops and configure all required spin_locks/mutex to use hdac_hda_ext library. Keep the code conditional so that the HDA link and audio codec support can be removed. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-bus.c | 108 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 108 insertions(+) create mode 100644 sound/soc/sof/intel/hda-bus.c (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c new file mode 100644 index 000000000000..62cc9921bb55 --- /dev/null +++ b/sound/soc/sof/intel/hda-bus.c @@ -0,0 +1,108 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Keyon Jie + +#include +#include +#include "../sof-priv.h" +#include "hda.h" + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + +static const struct hdac_bus_ops bus_ops = { + .command = snd_hdac_bus_send_cmd, + .get_response = snd_hdac_bus_get_response, +}; + +#endif + +static void sof_hda_writel(u32 value, u32 __iomem *addr) +{ + writel(value, addr); +} + +static u32 sof_hda_readl(u32 __iomem *addr) +{ + return readl(addr); +} + +static void sof_hda_writew(u16 value, u16 __iomem *addr) +{ + writew(value, addr); +} + +static u16 sof_hda_readw(u16 __iomem *addr) +{ + return readw(addr); +} + +static void sof_hda_writeb(u8 value, u8 __iomem *addr) +{ + writeb(value, addr); +} + +static u8 sof_hda_readb(u8 __iomem *addr) +{ + return readb(addr); +} + +static int sof_hda_dma_alloc_pages(struct hdac_bus *bus, int type, + size_t size, struct snd_dma_buffer *buf) +{ + return snd_dma_alloc_pages(type, bus->dev, size, buf); +} + +static void sof_hda_dma_free_pages(struct hdac_bus *bus, + struct snd_dma_buffer *buf) +{ + snd_dma_free_pages(buf); +} + +static const struct hdac_io_ops io_ops = { + .reg_writel = sof_hda_writel, + .reg_readl = sof_hda_readl, + .reg_writew = sof_hda_writew, + .reg_readw = sof_hda_readw, + .reg_writeb = sof_hda_writeb, + .reg_readb = sof_hda_readb, + .dma_alloc_pages = sof_hda_dma_alloc_pages, + .dma_free_pages = sof_hda_dma_free_pages, +}; + +/* + * This can be used for both with/without hda link support. + */ +void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev, + const struct hdac_ext_bus_ops *ext_ops) +{ + static int idx; + + memset(bus, 0, sizeof(*bus)); + bus->dev = dev; + + bus->io_ops = &io_ops; + INIT_LIST_HEAD(&bus->stream_list); + + bus->irq = -1; + bus->ext_ops = ext_ops; + bus->idx = idx++; + + spin_lock_init(&bus->reg_lock); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + INIT_LIST_HEAD(&bus->codec_list); + INIT_LIST_HEAD(&bus->hlink_list); + + mutex_init(&bus->cmd_mutex); + mutex_init(&bus->lock); + bus->ops = &bus_ops; + INIT_WORK(&bus->unsol_work, snd_hdac_bus_process_unsol_events); + bus->cmd_dma_state = true; +#endif + +} -- cgit v1.2.3-59-g8ed1b From a1d1e266b44517e736a456b3771152c33504f13d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:54 -0500 Subject: ASoC: SOF: Intel: Add Intel specific HDA stream operations Add support or HDA DSP stream operations for Intel HDA DSPs. Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 692 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 692 insertions(+) create mode 100644 sound/soc/sof/intel/hda-stream.c (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c new file mode 100644 index 000000000000..6290b2df5e62 --- /dev/null +++ b/sound/soc/sof/intel/hda-stream.c @@ -0,0 +1,692 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Liam Girdwood +// Ranjani Sridharan +// Rander Wang +// Keyon Jie +// + +/* + * Hardware interface for generic Intel audio DSP HDA IP + */ + +#include +#include +#include +#include +#include "../ops.h" +#include "hda.h" + +/* + * set up one of BDL entries for a stream + */ +static int hda_setup_bdle(struct snd_sof_dev *sdev, + struct snd_dma_buffer *dmab, + struct hdac_stream *stream, + struct sof_intel_dsp_bdl **bdlp, + int offset, int size, int ioc) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct sof_intel_dsp_bdl *bdl = *bdlp; + + while (size > 0) { + dma_addr_t addr; + int chunk; + + if (stream->frags >= HDA_DSP_MAX_BDL_ENTRIES) { + dev_err(sdev->dev, "error: stream frags exceeded\n"); + return -EINVAL; + } + + addr = snd_sgbuf_get_addr(dmab, offset); + /* program BDL addr */ + bdl->addr_l = cpu_to_le32(lower_32_bits(addr)); + bdl->addr_h = cpu_to_le32(upper_32_bits(addr)); + /* program BDL size */ + chunk = snd_sgbuf_get_chunk_size(dmab, offset, size); + /* one BDLE should not cross 4K boundary */ + if (bus->align_bdle_4k) { + u32 remain = 0x1000 - (offset & 0xfff); + + if (chunk > remain) + chunk = remain; + } + bdl->size = cpu_to_le32(chunk); + /* only program IOC when the whole segment is processed */ + size -= chunk; + bdl->ioc = (size || !ioc) ? 0 : cpu_to_le32(0x01); + bdl++; + stream->frags++; + offset += chunk; + + dev_vdbg(sdev->dev, "bdl, frags:%d, chunk size:0x%x;\n", + stream->frags, chunk); + } + + *bdlp = bdl; + return offset; +} + +/* + * set up Buffer Descriptor List (BDL) for host memory transfer + * BDL describes the location of the individual buffers and is little endian. + */ +int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, + struct snd_dma_buffer *dmab, + struct hdac_stream *stream) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + struct sof_intel_dsp_bdl *bdl; + int i, offset, period_bytes, periods; + int remain, ioc; + + period_bytes = stream->period_bytes; + dev_dbg(sdev->dev, "period_bytes:0x%x\n", period_bytes); + if (!period_bytes) + period_bytes = stream->bufsize; + + periods = stream->bufsize / period_bytes; + + dev_dbg(sdev->dev, "periods:%d\n", periods); + + remain = stream->bufsize % period_bytes; + if (remain) + periods++; + + /* program the initial BDL entries */ + bdl = (struct sof_intel_dsp_bdl *)stream->bdl.area; + offset = 0; + stream->frags = 0; + + /* + * set IOC if don't use position IPC + * and period_wakeup needed. + */ + ioc = hda->no_ipc_position ? + !stream->no_period_wakeup : 0; + + for (i = 0; i < periods; i++) { + if (i == (periods - 1) && remain) + /* set the last small entry */ + offset = hda_setup_bdle(sdev, dmab, + stream, &bdl, offset, + remain, 0); + else + offset = hda_setup_bdle(sdev, dmab, + stream, &bdl, offset, + period_bytes, ioc); + } + + return offset; +} + +int hda_dsp_stream_spib_config(struct snd_sof_dev *sdev, + struct hdac_ext_stream *stream, + int enable, u32 size) +{ + struct hdac_stream *hstream = &stream->hstream; + u32 mask; + + if (!sdev->bar[HDA_DSP_SPIB_BAR]) { + dev_err(sdev->dev, "error: address of spib capability is NULL\n"); + return -EINVAL; + } + + mask = (1 << hstream->index); + + /* enable/disable SPIB for the stream */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_SPIB_BAR, + SOF_HDA_ADSP_REG_CL_SPBFIFO_SPBFCCTL, mask, + enable << hstream->index); + + /* set the SPIB value */ + sof_io_write(sdev, stream->spib_addr, size); + + return 0; +} + +/* get next unused stream */ +struct hdac_ext_stream * +hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct hdac_ext_stream *stream = NULL; + struct hdac_stream *s; + + spin_lock_irq(&bus->reg_lock); + + /* get an unused stream */ + list_for_each_entry(s, &bus->stream_list, list) { + if (s->direction == direction && !s->opened) { + s->opened = true; + stream = stream_to_hdac_ext_stream(s); + break; + } + } + + spin_unlock_irq(&bus->reg_lock); + + /* stream found ? */ + if (!stream) + dev_err(sdev->dev, "error: no free %s streams\n", + direction == SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"); + + return stream; +} + +/* free a stream */ +int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct hdac_stream *s; + + spin_lock_irq(&bus->reg_lock); + + /* find used stream */ + list_for_each_entry(s, &bus->stream_list, list) { + if (s->direction == direction && + s->opened && s->stream_tag == stream_tag) { + s->opened = false; + spin_unlock_irq(&bus->reg_lock); + return 0; + } + } + + spin_unlock_irq(&bus->reg_lock); + + dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag); + return -ENODEV; +} + +int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, + struct hdac_ext_stream *stream, int cmd) +{ + struct hdac_stream *hstream = &stream->hstream; + int sd_offset = SOF_STREAM_SD_OFFSET(hstream); + + /* cmd must be for audio stream */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_START: + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, + 1 << hstream->index, + 1 << hstream->index); + + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + sd_offset, + SOF_HDA_SD_CTL_DMA_START | + SOF_HDA_CL_DMA_SD_INT_MASK, + SOF_HDA_SD_CTL_DMA_START | + SOF_HDA_CL_DMA_SD_INT_MASK); + + hstream->running = true; + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + sd_offset, + SOF_HDA_SD_CTL_DMA_START | + SOF_HDA_CL_DMA_SD_INT_MASK, 0x0); + + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, sd_offset + + SOF_HDA_ADSP_REG_CL_SD_STS, + SOF_HDA_CL_DMA_SD_INT_MASK); + + hstream->running = false; + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, + 1 << hstream->index, 0x0); + break; + default: + dev_err(sdev->dev, "error: unknown command: %d\n", cmd); + return -EINVAL; + } + + return 0; +} + +/* + * prepare for common hdac registers settings, for both code loader + * and normal stream. + */ +int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, + struct hdac_ext_stream *stream, + struct snd_dma_buffer *dmab, + struct snd_pcm_hw_params *params) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct hdac_stream *hstream = &stream->hstream; + int sd_offset = SOF_STREAM_SD_OFFSET(hstream); + int ret, timeout = HDA_DSP_STREAM_RESET_TIMEOUT; + u32 val, mask; + + if (!stream) { + dev_err(sdev->dev, "error: no stream available\n"); + return -ENODEV; + } + + /* decouple host and link DMA */ + mask = 0x1 << hstream->index; + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, mask); + + if (!dmab) { + dev_err(sdev->dev, "error: no dma buffer allocated!\n"); + return -ENODEV; + } + + /* clear stream status */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, sd_offset, + SOF_HDA_CL_DMA_SD_INT_MASK | + SOF_HDA_SD_CTL_DMA_START, 0); + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_STS, + SOF_HDA_CL_DMA_SD_INT_MASK, + SOF_HDA_CL_DMA_SD_INT_MASK); + + /* stream reset */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, sd_offset, 0x1, + 0x1); + udelay(3); + do { + val = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + sd_offset); + if (val & 0x1) + break; + } while (--timeout); + if (timeout == 0) { + dev_err(sdev->dev, "error: stream reset failed\n"); + return -ETIMEDOUT; + } + + timeout = HDA_DSP_STREAM_RESET_TIMEOUT; + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, sd_offset, 0x1, + 0x0); + + /* wait for hardware to report that stream is out of reset */ + udelay(3); + do { + val = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + sd_offset); + if ((val & 0x1) == 0) + break; + } while (--timeout); + if (timeout == 0) { + dev_err(sdev->dev, "error: timeout waiting for stream reset\n"); + return -ETIMEDOUT; + } + + if (hstream->posbuf) + *hstream->posbuf = 0; + + /* reset BDL address */ + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_BDLPL, + 0x0); + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_BDLPU, + 0x0); + + /* clear stream status */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, sd_offset, + SOF_HDA_CL_DMA_SD_INT_MASK | + SOF_HDA_SD_CTL_DMA_START, 0); + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_STS, + SOF_HDA_CL_DMA_SD_INT_MASK, + SOF_HDA_CL_DMA_SD_INT_MASK); + + hstream->frags = 0; + + ret = hda_dsp_stream_setup_bdl(sdev, dmab, hstream); + if (ret < 0) { + dev_err(sdev->dev, "error: set up of BDL failed\n"); + return ret; + } + + /* program stream tag to set up stream descriptor for DMA */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, sd_offset, + SOF_HDA_CL_SD_CTL_STREAM_TAG_MASK, + hstream->stream_tag << + SOF_HDA_CL_SD_CTL_STREAM_TAG_SHIFT); + + /* program cyclic buffer length */ + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_CBL, + hstream->bufsize); + + /* + * Recommended hardware programming sequence for HDAudio DMA format + * + * 1. Put DMA into coupled mode by clearing PPCTL.PROCEN bit + * for corresponding stream index before the time of writing + * format to SDxFMT register. + * 2. Write SDxFMT + * 3. Set PPCTL.PROCEN bit for corresponding stream index to + * enable decoupled mode + */ + + /* couple host and link DMA, disable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, 0); + + /* program stream format */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + sd_offset + + SOF_HDA_ADSP_REG_CL_SD_FORMAT, + 0xffff, hstream->format_val); + + /* decouple host and link DMA, enable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, mask); + + /* program last valid index */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_LVI, + 0xffff, (hstream->frags - 1)); + + /* program BDL address */ + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_BDLPL, + (u32)hstream->bdl.addr); + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_BDLPU, + upper_32_bits(hstream->bdl.addr)); + + /* enable position buffer */ + if (!(snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_ADSP_DPLBASE) + & SOF_HDA_ADSP_DPLBASE_ENABLE)) { + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_ADSP_DPUBASE, + upper_32_bits(bus->posbuf.addr)); + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_ADSP_DPLBASE, + (u32)bus->posbuf.addr | + SOF_HDA_ADSP_DPLBASE_ENABLE); + } + + /* set interrupt enable bits */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, sd_offset, + SOF_HDA_CL_DMA_SD_INT_MASK, + SOF_HDA_CL_DMA_SD_INT_MASK); + + /* read FIFO size */ + if (hstream->direction == SNDRV_PCM_STREAM_PLAYBACK) { + hstream->fifo_size = + snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + sd_offset + + SOF_HDA_ADSP_REG_CL_SD_FIFOSIZE); + hstream->fifo_size &= 0xffff; + hstream->fifo_size += 1; + } else { + hstream->fifo_size = 0; + } + + return ret; +} + +irqreturn_t hda_dsp_stream_interrupt(int irq, void *context) +{ + struct hdac_bus *bus = context; + u32 status; + + if (!pm_runtime_active(bus->dev)) + return IRQ_NONE; + + spin_lock(&bus->reg_lock); + + status = snd_hdac_chip_readl(bus, INTSTS); + if (status == 0 || status == 0xffffffff) { + spin_unlock(&bus->reg_lock); + return IRQ_NONE; + } + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* clear rirb int */ + status = snd_hdac_chip_readb(bus, RIRBSTS); + if (status & RIRB_INT_MASK) { + if (status & RIRB_INT_RESPONSE) + snd_hdac_bus_update_rirb(bus); + snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK); + } +#endif + + spin_unlock(&bus->reg_lock); + + return snd_hdac_chip_readl(bus, INTSTS) ? IRQ_WAKE_THREAD : IRQ_HANDLED; +} + +irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) +{ + struct hdac_bus *bus = context; + struct sof_intel_hda_dev *sof_hda = bus_to_sof_hda(bus); + struct hdac_stream *s; + u32 status = snd_hdac_chip_readl(bus, INTSTS); + u32 sd_status; + + /* check streams */ + list_for_each_entry(s, &bus->stream_list, list) { + if (status & (1 << s->index) && s->opened) { + sd_status = snd_hdac_stream_readb(s, SD_STS); + + dev_vdbg(bus->dev, "stream %d status 0x%x\n", + s->index, sd_status); + + snd_hdac_stream_writeb(s, SD_STS, SD_INT_MASK); + + if (!s->substream || + !s->running || + (sd_status & SOF_HDA_CL_DMA_SD_INT_COMPLETE) == 0) + continue; + + /* Inform ALSA only in case not do that with IPC */ + if (sof_hda->no_ipc_position) + snd_pcm_period_elapsed(s->substream); + + } + } + + return IRQ_HANDLED; +} + +int hda_dsp_stream_init(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct hdac_ext_stream *stream; + struct hdac_stream *hstream; + struct pci_dev *pci = to_pci_dev(sdev->dev); + int sd_offset; + int i, num_playback, num_capture, num_total, ret; + u32 gcap; + + gcap = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_GCAP); + dev_dbg(sdev->dev, "hda global caps = 0x%x\n", gcap); + + /* get stream count from GCAP */ + num_capture = (gcap >> 8) & 0x0f; + num_playback = (gcap >> 12) & 0x0f; + num_total = num_playback + num_capture; + + dev_dbg(sdev->dev, "detected %d playback and %d capture streams\n", + num_playback, num_capture); + + if (num_playback >= SOF_HDA_PLAYBACK_STREAMS) { + dev_err(sdev->dev, "error: too many playback streams %d\n", + num_playback); + return -EINVAL; + } + + if (num_capture >= SOF_HDA_CAPTURE_STREAMS) { + dev_err(sdev->dev, "error: too many capture streams %d\n", + num_playback); + return -EINVAL; + } + + /* + * mem alloc for the position buffer + * TODO: check position buffer update + */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, + SOF_HDA_DPIB_ENTRY_SIZE * num_total, + &bus->posbuf); + if (ret < 0) { + dev_err(sdev->dev, "error: posbuffer dma alloc failed\n"); + return -ENOMEM; + } + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* mem alloc for the CORB/RIRB ringbuffers */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, + PAGE_SIZE, &bus->rb); + if (ret < 0) { + dev_err(sdev->dev, "error: RB alloc failed\n"); + return -ENOMEM; + } +#endif + + /* create capture streams */ + for (i = 0; i < num_capture; i++) { + struct sof_intel_hda_stream *hda_stream; + + hda_stream = devm_kzalloc(sdev->dev, sizeof(*hda_stream), + GFP_KERNEL); + if (!hda_stream) + return -ENOMEM; + + stream = &hda_stream->hda_stream; + + stream->pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + + SOF_HDA_PPHC_BASE + SOF_HDA_PPHC_INTERVAL * i; + + stream->pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + + SOF_HDA_PPLC_BASE + SOF_HDA_PPLC_MULTI * num_total + + SOF_HDA_PPLC_INTERVAL * i; + + /* do we support SPIB */ + if (sdev->bar[HDA_DSP_SPIB_BAR]) { + stream->spib_addr = sdev->bar[HDA_DSP_SPIB_BAR] + + SOF_HDA_SPIB_BASE + SOF_HDA_SPIB_INTERVAL * i + + SOF_HDA_SPIB_SPIB; + + stream->fifo_addr = sdev->bar[HDA_DSP_SPIB_BAR] + + SOF_HDA_SPIB_BASE + SOF_HDA_SPIB_INTERVAL * i + + SOF_HDA_SPIB_MAXFIFO; + } + + hstream = &stream->hstream; + hstream->bus = bus; + hstream->sd_int_sta_mask = 1 << i; + hstream->index = i; + sd_offset = SOF_STREAM_SD_OFFSET(hstream); + hstream->sd_addr = sdev->bar[HDA_DSP_HDA_BAR] + sd_offset; + hstream->stream_tag = i + 1; + hstream->opened = false; + hstream->running = false; + hstream->direction = SNDRV_PCM_STREAM_CAPTURE; + + /* memory alloc for stream BDL */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, + HDA_DSP_BDL_SIZE, &hstream->bdl); + if (ret < 0) { + dev_err(sdev->dev, "error: stream bdl dma alloc failed\n"); + return -ENOMEM; + } + hstream->posbuf = (__le32 *)(bus->posbuf.area + + (hstream->index) * 8); + + list_add_tail(&hstream->list, &bus->stream_list); + } + + /* create playback streams */ + for (i = num_capture; i < num_total; i++) { + struct sof_intel_hda_stream *hda_stream; + + hda_stream = devm_kzalloc(sdev->dev, sizeof(*hda_stream), + GFP_KERNEL); + if (!hda_stream) + return -ENOMEM; + + stream = &hda_stream->hda_stream; + + /* we always have DSP support */ + stream->pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + + SOF_HDA_PPHC_BASE + SOF_HDA_PPHC_INTERVAL * i; + + stream->pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + + SOF_HDA_PPLC_BASE + SOF_HDA_PPLC_MULTI * num_total + + SOF_HDA_PPLC_INTERVAL * i; + + /* do we support SPIB */ + if (sdev->bar[HDA_DSP_SPIB_BAR]) { + stream->spib_addr = sdev->bar[HDA_DSP_SPIB_BAR] + + SOF_HDA_SPIB_BASE + SOF_HDA_SPIB_INTERVAL * i + + SOF_HDA_SPIB_SPIB; + + stream->fifo_addr = sdev->bar[HDA_DSP_SPIB_BAR] + + SOF_HDA_SPIB_BASE + SOF_HDA_SPIB_INTERVAL * i + + SOF_HDA_SPIB_MAXFIFO; + } + + hstream = &stream->hstream; + hstream->bus = bus; + hstream->sd_int_sta_mask = 1 << i; + hstream->index = i; + sd_offset = SOF_STREAM_SD_OFFSET(hstream); + hstream->sd_addr = sdev->bar[HDA_DSP_HDA_BAR] + sd_offset; + hstream->stream_tag = i - num_capture + 1; + hstream->opened = false; + hstream->running = false; + hstream->direction = SNDRV_PCM_STREAM_PLAYBACK; + + /* mem alloc for stream BDL */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, + HDA_DSP_BDL_SIZE, &hstream->bdl); + if (ret < 0) { + dev_err(sdev->dev, "error: stream bdl dma alloc failed\n"); + return -ENOMEM; + } + + hstream->posbuf = (__le32 *)(bus->posbuf.area + + (hstream->index) * 8); + + list_add_tail(&hstream->list, &bus->stream_list); + } + + return 0; +} + +void hda_dsp_stream_free(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct hdac_stream *s, *_s; + struct hdac_ext_stream *stream; + struct sof_intel_hda_stream *hda_stream; + + /* free position buffer */ + if (bus->posbuf.area) + snd_dma_free_pages(&bus->posbuf); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + /* free position buffer */ + if (bus->rb.area) + snd_dma_free_pages(&bus->rb); +#endif + + list_for_each_entry_safe(s, _s, &bus->stream_list, list) { + /* TODO: decouple */ + + /* free bdl buffer */ + if (s->bdl.area) + snd_dma_free_pages(&s->bdl); + list_del(&s->list); + stream = stream_to_hdac_ext_stream(s); + hda_stream = container_of(stream, struct sof_intel_hda_stream, + hda_stream); + devm_kfree(sdev->dev, hda_stream); + } +} -- cgit v1.2.3-59-g8ed1b From ba00ed7572cd581bfd483d13193d36a38a3942a8 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:55 -0500 Subject: ASoC: SOF: Intel: Add Intel specific HDA trace operations Add trace operations for Intel based HDA DSPs Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-trace.c | 94 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 94 insertions(+) create mode 100644 sound/soc/sof/intel/hda-trace.c (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-trace.c b/sound/soc/sof/intel/hda-trace.c new file mode 100644 index 000000000000..33b23bd6a01e --- /dev/null +++ b/sound/soc/sof/intel/hda-trace.c @@ -0,0 +1,94 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Liam Girdwood +// Ranjani Sridharan +// Rander Wang +// Keyon Jie +// + +/* + * Hardware interface for generic Intel audio DSP HDA IP + */ + +#include +#include "../ops.h" +#include "hda.h" + +static int hda_dsp_trace_prepare(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + struct hdac_ext_stream *stream = hda->dtrace_stream; + struct hdac_stream *hstream = &stream->hstream; + struct snd_dma_buffer *dmab = &sdev->dmatb; + int ret; + + hstream->period_bytes = 0;/* initialize period_bytes */ + hstream->bufsize = sdev->dmatb.bytes; + + ret = hda_dsp_stream_hw_params(sdev, stream, dmab, NULL); + if (ret < 0) + dev_err(sdev->dev, "error: hdac prepare failed: %x\n", ret); + + return ret; +} + +int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + int ret; + + hda->dtrace_stream = hda_dsp_stream_get(sdev, + SNDRV_PCM_STREAM_CAPTURE); + + if (!hda->dtrace_stream) { + dev_err(sdev->dev, + "error: no available capture stream for DMA trace\n"); + return -ENODEV; + } + + *stream_tag = hda->dtrace_stream->hstream.stream_tag; + + /* + * initialize capture stream, set BDL address and return corresponding + * stream tag which will be sent to the firmware by IPC message. + */ + ret = hda_dsp_trace_prepare(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: hdac trace init failed: %x\n", ret); + hda_dsp_stream_put(sdev, SNDRV_PCM_STREAM_CAPTURE, *stream_tag); + hda->dtrace_stream = NULL; + *stream_tag = 0; + } + + return ret; +} + +int hda_dsp_trace_release(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + struct hdac_stream *hstream; + + if (hda->dtrace_stream) { + hstream = &hda->dtrace_stream->hstream; + hda_dsp_stream_put(sdev, + SNDRV_PCM_STREAM_CAPTURE, + hstream->stream_tag); + hda->dtrace_stream = NULL; + return 0; + } + + dev_dbg(sdev->dev, "DMA trace stream is not opened!\n"); + return -ENODEV; +} + +int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + + return hda_dsp_stream_trigger(sdev, hda->dtrace_stream, cmd); +} -- cgit v1.2.3-59-g8ed1b From 5507b8103e2653b4f5fc2e4c339d7eacde00da2d Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 12 Apr 2019 11:08:56 -0500 Subject: ASoC: SOF: Intel: Add support for HDAudio codecs Add probe, init and cleanup routines for HDaudio. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-codec.c | 171 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 171 insertions(+) create mode 100644 sound/soc/sof/intel/hda-codec.c (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c new file mode 100644 index 000000000000..b8b37f082309 --- /dev/null +++ b/sound/soc/sof/intel/hda-codec.c @@ -0,0 +1,171 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Keyon Jie +// + +#include +#include +#include +#include +#include +#include "../ops.h" +#include "hda.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) +#include "../../codecs/hdac_hda.h" +#endif /* CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC */ + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) +#define IDISP_VID_INTEL 0x80860000 + +/* load the legacy HDA codec driver */ +#ifdef MODULE +static void hda_codec_load_module(struct hda_codec *codec) +{ + char alias[MODULE_NAME_LEN]; + const char *module = alias; + + snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias)); + dev_dbg(&codec->core.dev, "loading codec module: %s\n", module); + request_module(module); +} +#else +static void hda_codec_load_module(struct hda_codec *codec) {} +#endif + +#endif /* CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC */ + +/* probe individual codec */ +static int hda_codec_probe(struct snd_sof_dev *sdev, int address) +{ + struct hda_bus *hbus = sof_to_hbus(sdev); + struct hdac_device *hdev; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + struct hdac_hda_priv *hda_priv; +#endif + u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | + (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; + u32 resp = -1; + int ret; + + mutex_lock(&hbus->core.cmd_mutex); + snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); + snd_hdac_bus_get_response(&hbus->core, address, &resp); + mutex_unlock(&hbus->core.cmd_mutex); + if (resp == -1) + return -EIO; + dev_dbg(sdev->dev, "HDA codec #%d probed OK: response: %x\n", + address, resp); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + /* snd_hdac_ext_bus_device_exit will use kfree to free hdev */ + hda_priv = kzalloc(sizeof(*hda_priv), GFP_KERNEL); + if (!hda_priv) + return -ENOMEM; + + hda_priv->codec.bus = hbus; + hdev = &hda_priv->codec.core; + + ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev); + if (ret < 0) + return ret; + + /* use legacy bus only for HDA codecs, idisp uses ext bus */ + if ((resp & 0xFFFF0000) != IDISP_VID_INTEL) { + hdev->type = HDA_DEV_LEGACY; + hda_codec_load_module(&hda_priv->codec); + } + + return 0; +#else + /* snd_hdac_ext_bus_device_exit will use kfree to free hdev */ + hdev = kzalloc(sizeof(*hdev), GFP_KERNEL); + if (!hdev) + return -ENOMEM; + + ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev); + + return ret; +#endif +} + +/* Codec initialization */ +int hda_codec_probe_bus(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + int i, ret; + + /* probe codecs in avail slots */ + for (i = 0; i < HDA_MAX_CODECS; i++) { + + if (!(bus->codec_mask & (1 << i))) + continue; + + ret = hda_codec_probe(sdev, i); + if (ret < 0) { + dev_err(bus->dev, "error: codec #%d probe error, ret: %d\n", + i, ret); + return ret; + } + } + + return 0; +} +EXPORT_SYMBOL(hda_codec_probe_bus); + +#if IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI) + +void hda_codec_i915_get(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + + dev_dbg(bus->dev, "Turning i915 HDAC power on\n"); + snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, true); +} +EXPORT_SYMBOL(hda_codec_i915_get); + +void hda_codec_i915_put(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + + dev_dbg(bus->dev, "Turning i915 HDAC power off\n"); + snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); +} +EXPORT_SYMBOL(hda_codec_i915_put); + +int hda_codec_i915_init(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + int ret; + + /* i915 exposes a HDA codec for HDMI audio */ + ret = snd_hdac_i915_init(bus); + if (ret < 0) + return ret; + + hda_codec_i915_get(sdev); + + return 0; +} +EXPORT_SYMBOL(hda_codec_i915_init); + +int hda_codec_i915_exit(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + int ret; + + hda_codec_i915_put(sdev); + + ret = snd_hdac_i915_exit(bus); + + return ret; +} +EXPORT_SYMBOL(hda_codec_i915_exit); + +#endif /* CONFIG_SND_SOC_HDAC_HDMI */ + +MODULE_LICENSE("Dual BSD/GPL"); -- cgit v1.2.3-59-g8ed1b From fdd961e37e47f2d450638f3a653cdb96911969ea Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Fri, 12 Apr 2019 11:08:57 -0500 Subject: ASoC: SOF: Intel: add SKL+ platform DAIs Add declarations for DAIs and utilities for link DMA management Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 351 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 351 insertions(+) create mode 100644 sound/soc/sof/intel/hda-dai.c (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c new file mode 100644 index 000000000000..830328af19c5 --- /dev/null +++ b/sound/soc/sof/intel/hda-dai.c @@ -0,0 +1,351 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Keyon Jie +// + +#include +#include +#include "../sof-priv.h" +#include "hda.h" + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + +struct hda_pipe_params { + u8 host_dma_id; + u8 link_dma_id; + u32 ch; + u32 s_freq; + u32 s_fmt; + u8 linktype; + snd_pcm_format_t format; + int link_index; + int stream; + unsigned int host_bps; + unsigned int link_bps; +}; + +/* + * Unlike GP dma, there is a set of stream registers in hda controller + * to control the link dma channels. Each register controls one link + * dma channel and the relation is fixed. To make sure FW uses correct + * link dma channels, host allocates stream registers and sends the + * corresponding link dma channels to FW to allocate link dma channel + * + * FIXME: this API is abused in the sense that tx_num and rx_num are + * passed as arguments, not returned. We need to find a better way to + * retrieve the stream tag allocated for the link DMA + */ +static int hda_link_dma_get_channels(struct snd_soc_dai *dai, + unsigned int *tx_num, + unsigned int *tx_slot, + unsigned int *rx_num, + unsigned int *rx_slot) +{ + struct hdac_bus *bus; + struct hdac_ext_stream *stream; + struct snd_pcm_substream substream; + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + + bus = sof_to_bus(sdev); + + memset(&substream, 0, sizeof(substream)); + if (*tx_num == 1) { + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + stream = snd_hdac_ext_stream_assign(bus, &substream, + HDAC_EXT_STREAM_TYPE_LINK); + if (!stream) { + dev_err(bus->dev, "error: failed to find a free hda ext stream for playback"); + return -EBUSY; + } + + snd_soc_dai_set_dma_data(dai, &substream, stream); + *tx_slot = hdac_stream(stream)->stream_tag - 1; + + dev_dbg(bus->dev, "link dma channel %d for playback", *tx_slot); + } + + if (*rx_num == 1) { + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + stream = snd_hdac_ext_stream_assign(bus, &substream, + HDAC_EXT_STREAM_TYPE_LINK); + if (!stream) { + dev_err(bus->dev, "error: failed to find a free hda ext stream for capture"); + return -EBUSY; + } + + snd_soc_dai_set_dma_data(dai, &substream, stream); + *rx_slot = hdac_stream(stream)->stream_tag - 1; + + dev_dbg(bus->dev, "link dma channel %d for capture", *rx_slot); + } + + return 0; +} + +static int hda_link_dma_params(struct hdac_ext_stream *stream, + struct hda_pipe_params *params) +{ + struct hdac_stream *hstream = &stream->hstream; + unsigned char stream_tag = hstream->stream_tag; + struct hdac_bus *bus = hstream->bus; + struct hdac_ext_link *link; + unsigned int format_val; + + snd_hdac_ext_stream_decouple(bus, stream, true); + snd_hdac_ext_link_stream_reset(stream); + + format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch, + params->format, + params->link_bps, 0); + + dev_dbg(bus->dev, "format_val=%d, rate=%d, ch=%d, format=%d\n", + format_val, params->s_freq, params->ch, params->format); + + snd_hdac_ext_link_stream_setup(stream, format_val); + + if (stream->hstream.direction == SNDRV_PCM_STREAM_PLAYBACK) { + list_for_each_entry(link, &bus->hlink_list, list) { + if (link->index == params->link_index) + snd_hdac_ext_link_set_stream_id(link, + stream_tag); + } + } + + stream->link_prepared = 1; + + return 0; +} + +static int hda_link_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_bus *bus = hstream->bus; + struct hdac_ext_stream *link_dev; + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct hda_pipe_params p_params = {0}; + struct hdac_ext_link *link; + int stream_tag; + + link_dev = snd_soc_dai_get_dma_data(dai, substream); + + link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); + if (!link) + return -EINVAL; + + stream_tag = hdac_stream(link_dev)->stream_tag; + + /* set the stream tag in the codec dai dma params */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); + else + snd_soc_dai_set_tdm_slot(codec_dai, 0, stream_tag, 0, 0); + + p_params.s_fmt = snd_pcm_format_width(params_format(params)); + p_params.ch = params_channels(params); + p_params.s_freq = params_rate(params); + p_params.stream = substream->stream; + p_params.link_dma_id = stream_tag - 1; + p_params.link_index = link->index; + p_params.format = params_format(params); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + p_params.link_bps = codec_dai->driver->playback.sig_bits; + else + p_params.link_bps = codec_dai->driver->capture.sig_bits; + + return hda_link_dma_params(link_dev, &p_params); +} + +static int hda_link_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + struct snd_sof_pcm *spcm; + int stream = substream->stream; + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) + return -EINVAL; + + /* setup hw_params again only if resuming from system suspend */ + if (!spcm->hw_params_upon_resume[stream]) + return 0; + + dev_dbg(sdev->dev, "hda: prepare stream %d dir %d\n", + spcm->pcm.pcm_id, substream->stream); + + return hda_link_hw_params(substream, &rtd->dpcm[stream].hw_params, + dai); +} + +static int hda_link_pcm_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *link_dev = + snd_soc_dai_get_dma_data(dai, substream); + int ret; + + dev_dbg(dai->dev, "In %s cmd=%d\n", __func__, cmd); + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + /* set up hw_params */ + ret = hda_link_pcm_prepare(substream, dai); + if (ret < 0) { + dev_err(dai->dev, + "error: setting up hw_params during resume\n"); + return ret; + } + + /* fallthrough */ + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + snd_hdac_ext_link_stream_start(link_dev); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + snd_hdac_ext_link_stream_clear(link_dev); + break; + default: + return -EINVAL; + } + return 0; +} + +/* + * FIXME: This API is also abused since it's used for two purposes. + * when the substream argument is NULL this function is used for cleanups + * that aren't necessarily required, and called explicitly by handling + * ASoC core structures, which is not recommended. + * This part will be reworked in follow-up patches. + */ +static int hda_link_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + const char *name; + unsigned int stream_tag; + struct hdac_bus *bus; + struct hdac_ext_link *link; + struct hdac_stream *hstream; + struct hdac_ext_stream *stream; + struct snd_soc_pcm_runtime *rtd; + struct hdac_ext_stream *link_dev; + struct snd_pcm_substream pcm_substream; + + memset(&pcm_substream, 0, sizeof(pcm_substream)); + if (substream) { + hstream = substream->runtime->private_data; + bus = hstream->bus; + rtd = snd_pcm_substream_chip(substream); + link_dev = snd_soc_dai_get_dma_data(dai, substream); + snd_hdac_ext_stream_decouple(bus, link_dev, false); + name = rtd->codec_dai->component->name; + link = snd_hdac_ext_bus_get_link(bus, name); + if (!link) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + stream_tag = hdac_stream(link_dev)->stream_tag; + snd_hdac_ext_link_clear_stream_id(link, stream_tag); + } + + link_dev->link_prepared = 0; + } else { + /* release all hda streams when dai link is unloaded */ + pcm_substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + stream = snd_soc_dai_get_dma_data(dai, &pcm_substream); + if (stream) { + snd_soc_dai_set_dma_data(dai, &pcm_substream, NULL); + snd_hdac_ext_stream_release(stream, + HDAC_EXT_STREAM_TYPE_LINK); + } + + pcm_substream.stream = SNDRV_PCM_STREAM_CAPTURE; + stream = snd_soc_dai_get_dma_data(dai, &pcm_substream); + if (stream) { + snd_soc_dai_set_dma_data(dai, &pcm_substream, NULL); + snd_hdac_ext_stream_release(stream, + HDAC_EXT_STREAM_TYPE_LINK); + } + } + + return 0; +} + +static const struct snd_soc_dai_ops hda_link_dai_ops = { + .hw_params = hda_link_hw_params, + .hw_free = hda_link_hw_free, + .trigger = hda_link_pcm_trigger, + .prepare = hda_link_pcm_prepare, + .get_channel_map = hda_link_dma_get_channels, +}; +#endif + +/* + * common dai driver for skl+ platforms. + * some products who use this DAI array only physically have a subset of + * the DAIs, but no harm is done here by adding the whole set. + */ +struct snd_soc_dai_driver skl_dai[] = { +{ + .name = "SSP0 Pin", +}, +{ + .name = "SSP1 Pin", +}, +{ + .name = "SSP2 Pin", +}, +{ + .name = "SSP3 Pin", +}, +{ + .name = "SSP4 Pin", +}, +{ + .name = "SSP5 Pin", +}, +{ + .name = "DMIC01 Pin", +}, +{ + .name = "DMIC16k Pin", +}, +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) +{ + .name = "iDisp1 Pin", + .ops = &hda_link_dai_ops, +}, +{ + .name = "iDisp2 Pin", + .ops = &hda_link_dai_ops, +}, +{ + .name = "iDisp3 Pin", + .ops = &hda_link_dai_ops, +}, +{ + .name = "Analog CPU DAI", + .ops = &hda_link_dai_ops, +}, +{ + .name = "Digital CPU DAI", + .ops = &hda_link_dai_ops, +}, +{ + .name = "Alt Analog CPU DAI", + .ops = &hda_link_dai_ops, +}, +#endif +}; -- cgit v1.2.3-59-g8ed1b From 273020522ef62361c5d86eebe45a72418ed8dea4 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:08:58 -0500 Subject: ASoC: SOF: Intel: Add platform differentiation for APL and CNL Add platform differentiation operations for different Intel HDA DSP platforms. Signed-off-by: Keyon Jie Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/intel/apl.c | 109 ++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 249 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 358 insertions(+) create mode 100644 sound/soc/sof/intel/apl.c create mode 100644 sound/soc/sof/intel/cnl.c (limited to 'sound') diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c new file mode 100644 index 000000000000..8c6282606944 --- /dev/null +++ b/sound/soc/sof/intel/apl.c @@ -0,0 +1,109 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Liam Girdwood +// Ranjani Sridharan +// Rander Wang +// Keyon Jie +// + +/* + * Hardware interface for audio DSP on Apollolake and GeminiLake + */ + +#include "../sof-priv.h" +#include "hda.h" + +static const struct snd_sof_debugfs_map apl_dsp_debugfs[] = { + {"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"pp", HDA_DSP_PP_BAR, 0, 0x1000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS}, +}; + +/* apollolake ops */ +const struct snd_sof_dsp_ops sof_apl_ops = { + /* probe and remove */ + .probe = hda_dsp_probe, + .remove = hda_dsp_remove, + + /* Register IO */ + .write = sof_io_write, + .read = sof_io_read, + .write64 = sof_io_write64, + .read64 = sof_io_read64, + + /* Block IO */ + .block_read = sof_block_read, + .block_write = sof_block_write, + + /* doorbell */ + .irq_handler = hda_dsp_ipc_irq_handler, + .irq_thread = hda_dsp_ipc_irq_thread, + + /* ipc */ + .send_msg = hda_dsp_ipc_send_msg, + .fw_ready = hda_dsp_ipc_fw_ready, + + .ipc_msg_data = hda_ipc_msg_data, + .ipc_pcm_params = hda_ipc_pcm_params, + + /* debug */ + .debug_map = apl_dsp_debugfs, + .debug_map_count = ARRAY_SIZE(apl_dsp_debugfs), + .dbg_dump = hda_dsp_dump, + + /* stream callbacks */ + .pcm_open = hda_dsp_pcm_open, + .pcm_close = hda_dsp_pcm_close, + .pcm_hw_params = hda_dsp_pcm_hw_params, + .pcm_trigger = hda_dsp_pcm_trigger, + .pcm_pointer = hda_dsp_pcm_pointer, + + /* firmware loading */ + .load_firmware = snd_sof_load_firmware_raw, + + /* firmware run */ + .run = hda_dsp_cl_boot_firmware, + + /* pre/post fw run */ + .pre_fw_run = hda_dsp_pre_fw_run, + .post_fw_run = hda_dsp_post_fw_run, + + /* dsp core power up/down */ + .core_power_up = hda_dsp_enable_core, + .core_power_down = hda_dsp_core_reset_power_down, + + /* trace callback */ + .trace_init = hda_dsp_trace_init, + .trace_release = hda_dsp_trace_release, + .trace_trigger = hda_dsp_trace_trigger, + + /* DAI drivers */ + .drv = skl_dai, + .num_drv = SOF_SKL_NUM_DAIS, + + /* PM */ + .suspend = hda_dsp_suspend, + .resume = hda_dsp_resume, + .runtime_suspend = hda_dsp_runtime_suspend, + .runtime_resume = hda_dsp_runtime_resume, +}; +EXPORT_SYMBOL(sof_apl_ops); + +const struct sof_intel_dsp_desc apl_chip_info = { + /* Apollolake */ + .cores_num = 2, + .init_core_mask = 1, + .cores_mask = HDA_DSP_CORE_MASK(0) | HDA_DSP_CORE_MASK(1), + .ipc_req = HDA_DSP_REG_HIPCI, + .ipc_req_mask = HDA_DSP_REG_HIPCI_BUSY, + .ipc_ack = HDA_DSP_REG_HIPCIE, + .ipc_ack_mask = HDA_DSP_REG_HIPCIE_DONE, + .ipc_ctl = HDA_DSP_REG_HIPCCTL, + .rom_init_timeout = 150, +}; +EXPORT_SYMBOL(apl_chip_info); diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c new file mode 100644 index 000000000000..3e95c1e5e491 --- /dev/null +++ b/sound/soc/sof/intel/cnl.c @@ -0,0 +1,249 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2018 Intel Corporation. All rights reserved. +// +// Authors: Liam Girdwood +// Ranjani Sridharan +// Rander Wang +// Keyon Jie +// + +/* + * Hardware interface for audio DSP on Cannonlake. + */ + +#include "../ops.h" +#include "hda.h" + +static const struct snd_sof_debugfs_map cnl_dsp_debugfs[] = { + {"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"pp", HDA_DSP_PP_BAR, 0, 0x1000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS}, +}; + +static void cnl_ipc_host_done(struct snd_sof_dev *sdev); +static void cnl_ipc_dsp_done(struct snd_sof_dev *sdev); + +static irqreturn_t cnl_ipc_irq_thread(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + u32 hipci; + u32 hipcctl; + u32 hipcida; + u32 hipctdr; + u32 hipctdd; + u32 msg; + u32 msg_ext; + irqreturn_t ret = IRQ_NONE; + + /* here we handle IPC interrupts only */ + if (!(sdev->irq_status & HDA_DSP_ADSPIS_IPC)) + return ret; + + hipcida = snd_sof_dsp_read(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDA); + hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCCTL); + hipctdr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCTDR); + + /* reenable IPC interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, + HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC); + + /* reply message from DSP */ + if (hipcida & CNL_DSP_REG_HIPCIDA_DONE && + hipcctl & CNL_DSP_REG_HIPCCTL_DONE) { + hipci = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + CNL_DSP_REG_HIPCIDR); + msg_ext = hipci & CNL_DSP_REG_HIPCIDR_MSG_MASK; + msg = hipcida & CNL_DSP_REG_HIPCIDA_MSG_MASK; + + dev_vdbg(sdev->dev, + "ipc: firmware response, msg:0x%x, msg_ext:0x%x\n", + msg, msg_ext); + + /* mask Done interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, + CNL_DSP_REG_HIPCCTL, + CNL_DSP_REG_HIPCCTL_DONE, 0); + + /* handle immediate reply from DSP core */ + hda_dsp_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, msg); + + if (sdev->code_loading) { + sdev->code_loading = 0; + wake_up(&sdev->waitq); + } + + cnl_ipc_dsp_done(sdev); + + ret = IRQ_HANDLED; + } + + /* new message from DSP */ + if (hipctdr & CNL_DSP_REG_HIPCTDR_BUSY) { + hipctdd = snd_sof_dsp_read(sdev, HDA_DSP_BAR, + CNL_DSP_REG_HIPCTDD); + msg = hipctdr & CNL_DSP_REG_HIPCTDR_MSG_MASK; + msg_ext = hipctdd & CNL_DSP_REG_HIPCTDD_MSG_MASK; + + dev_vdbg(sdev->dev, + "ipc: firmware initiated, msg:0x%x, msg_ext:0x%x\n", + msg, msg_ext); + + /* handle messages from DSP */ + if ((hipctdr & SOF_IPC_PANIC_MAGIC_MASK) == + SOF_IPC_PANIC_MAGIC) { + snd_sof_dsp_panic(sdev, HDA_DSP_PANIC_OFFSET(msg_ext)); + } else { + snd_sof_ipc_msgs_rx(sdev); + } + + /* + * clear busy interrupt to tell dsp controller this + * interrupt has been accepted, not trigger it again + */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, + CNL_DSP_REG_HIPCTDR, + CNL_DSP_REG_HIPCTDR_BUSY, + CNL_DSP_REG_HIPCTDR_BUSY); + + cnl_ipc_host_done(sdev); + + ret = IRQ_HANDLED; + } + + return ret; +} + +static void cnl_ipc_host_done(struct snd_sof_dev *sdev) +{ + /* + * set done bit to ack dsp the msg has been + * processed and send reply msg to dsp + */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, + CNL_DSP_REG_HIPCTDA, + CNL_DSP_REG_HIPCTDA_DONE, + CNL_DSP_REG_HIPCTDA_DONE); +} + +static void cnl_ipc_dsp_done(struct snd_sof_dev *sdev) +{ + /* + * set DONE bit - tell DSP we have received the reply msg + * from DSP, and processed it, don't send more reply to host + */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, + CNL_DSP_REG_HIPCIDA, + CNL_DSP_REG_HIPCIDA_DONE, + CNL_DSP_REG_HIPCIDA_DONE); + + /* unmask Done interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, + CNL_DSP_REG_HIPCCTL, + CNL_DSP_REG_HIPCCTL_DONE, + CNL_DSP_REG_HIPCCTL_DONE); +} + +static int cnl_ipc_send_msg(struct snd_sof_dev *sdev, + struct snd_sof_ipc_msg *msg) +{ + u32 cmd = msg->header; + + /* send the message */ + sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data, + msg->msg_size); + snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR, + cmd | CNL_DSP_REG_HIPCIDR_BUSY); + + return 0; +} + +/* cannonlake ops */ +const struct snd_sof_dsp_ops sof_cnl_ops = { + /* probe and remove */ + .probe = hda_dsp_probe, + .remove = hda_dsp_remove, + + /* Register IO */ + .write = sof_io_write, + .read = sof_io_read, + .write64 = sof_io_write64, + .read64 = sof_io_read64, + + /* Block IO */ + .block_read = sof_block_read, + .block_write = sof_block_write, + + /* doorbell */ + .irq_handler = hda_dsp_ipc_irq_handler, + .irq_thread = cnl_ipc_irq_thread, + + /* ipc */ + .send_msg = cnl_ipc_send_msg, + .fw_ready = hda_dsp_ipc_fw_ready, + + .ipc_msg_data = hda_ipc_msg_data, + .ipc_pcm_params = hda_ipc_pcm_params, + + /* debug */ + .debug_map = cnl_dsp_debugfs, + .debug_map_count = ARRAY_SIZE(cnl_dsp_debugfs), + .dbg_dump = hda_dsp_dump, + + /* stream callbacks */ + .pcm_open = hda_dsp_pcm_open, + .pcm_close = hda_dsp_pcm_close, + .pcm_hw_params = hda_dsp_pcm_hw_params, + .pcm_trigger = hda_dsp_pcm_trigger, + + /* firmware loading */ + .load_firmware = snd_sof_load_firmware_raw, + + /* pre/post fw run */ + .pre_fw_run = hda_dsp_pre_fw_run, + .post_fw_run = hda_dsp_post_fw_run, + + /* dsp core power up/down */ + .core_power_up = hda_dsp_enable_core, + .core_power_down = hda_dsp_core_reset_power_down, + + /* firmware run */ + .run = hda_dsp_cl_boot_firmware, + + /* trace callback */ + .trace_init = hda_dsp_trace_init, + .trace_release = hda_dsp_trace_release, + .trace_trigger = hda_dsp_trace_trigger, + + /* DAI drivers */ + .drv = skl_dai, + .num_drv = SOF_SKL_NUM_DAIS, + + /* PM */ + .suspend = hda_dsp_suspend, + .resume = hda_dsp_resume, + .runtime_suspend = hda_dsp_runtime_suspend, + .runtime_resume = hda_dsp_runtime_resume, +}; +EXPORT_SYMBOL(sof_cnl_ops); + +const struct sof_intel_dsp_desc cnl_chip_info = { + /* Cannonlake */ + .cores_num = 4, + .init_core_mask = 1, + .cores_mask = HDA_DSP_CORE_MASK(0) | + HDA_DSP_CORE_MASK(1) | + HDA_DSP_CORE_MASK(2) | + HDA_DSP_CORE_MASK(3), + .ipc_req = CNL_DSP_REG_HIPCIDR, + .ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY, + .ipc_ack = CNL_DSP_REG_HIPCIDA, + .ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE, + .ipc_ctl = CNL_DSP_REG_HIPCCTL, + .rom_init_timeout = 300, +}; +EXPORT_SYMBOL(cnl_chip_info); -- cgit v1.2.3-59-g8ed1b From e6b140e918e93ec76265007b86092a6a3068ca1d Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 12 Apr 2019 11:09:01 -0500 Subject: ASoC: Intel: Kconfig: expose common option between SST and SOF drivers Both drivers rely on the same module, expose it for both configurations Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index bd9fd2035c55..fc1396adde71 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -196,13 +196,18 @@ config SND_SOC_INTEL_SKYLAKE_COMMON endif ## SND_SOC_INTEL_SKYLAKE_FAMILY +endif ## SND_SOC_INTEL_SST_TOPLEVEL + +if SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL + config SND_SOC_ACPI_INTEL_MATCH tristate select SND_SOC_ACPI if ACPI # this option controls the compilation of ACPI matching tables and # helpers and is not meant to be selected by the user. -endif ## SND_SOC_INTEL_SST_TOPLEVEL +endif ## SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL + # ASoC codec drivers source "sound/soc/intel/boards/Kconfig" -- cgit v1.2.3-59-g8ed1b From 7e978fa37df7f771a5d02934ff828a0ee36bcf11 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:09:02 -0500 Subject: ASoC: SOF: Add Build support for SOF core and Intel drivers Build SOF core and Intel-specific drivers. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sof/Kconfig | 141 ++++++++++++++++++++++++++ sound/soc/sof/Makefile | 18 ++++ sound/soc/sof/intel/Kconfig | 230 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/Makefile | 19 ++++ 6 files changed, 410 insertions(+) create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/intel/Kconfig create mode 100644 sound/soc/sof/intel/Makefile (limited to 'sound') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index aa35940f5c50..297be0ca3dbc 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -63,6 +63,7 @@ source "sound/soc/rockchip/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/sirf/Kconfig" +source "sound/soc/sof/Kconfig" source "sound/soc/spear/Kconfig" source "sound/soc/sprd/Kconfig" source "sound/soc/sti/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 974fb9821e17..d90ce8a32887 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -47,6 +47,7 @@ obj-$(CONFIG_SND_SOC) += rockchip/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += sirf/ +obj-$(CONFIG_SND_SOC) += sof/ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += sprd/ obj-$(CONFIG_SND_SOC) += sti/ diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig new file mode 100644 index 000000000000..eaa380092c3b --- /dev/null +++ b/sound/soc/sof/Kconfig @@ -0,0 +1,141 @@ +config SND_SOC_SOF_TOPLEVEL + bool "Sound Open Firmware Support" + help + This adds support for Sound Open Firmware (SOF). SOF is a free and + generic open source audio DSP firmware for multiple devices. + Say Y if you have such a device that is supported by SOF. + If unsure select "N". + +if SND_SOC_SOF_TOPLEVEL + +config SND_SOC_SOF_PCI + tristate "SOF PCI enumeration support" + depends on PCI + select SND_SOC_SOF + select SND_SOC_ACPI if ACPI + select SND_SOC_SOF_OPTIONS + select SND_SOC_SOF_INTEL_PCI if SND_SOC_SOF_INTEL_TOPLEVEL + help + This adds support for PCI enumeration. This option is + required to enable Intel Skylake+ devices + Say Y if you need this option + If unsure select "N". + +config SND_SOC_SOF_ACPI + tristate "SOF ACPI enumeration support" + depends on ACPI || COMPILE_TEST + select SND_SOC_SOF + select SND_SOC_ACPI if ACPI + select SND_SOC_SOF_OPTIONS + select SND_SOC_SOF_INTEL_ACPI if SND_SOC_SOF_INTEL_TOPLEVEL + select IOSF_MBI if X86 + help + This adds support for ACPI enumeration. This option is required + to enable Intel Haswell/Broadwell/Baytrail/Cherrytrail devices + Say Y if you need this option + If unsure select "N". + +config SND_SOC_SOF_OPTIONS + tristate + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +if SND_SOC_SOF_OPTIONS + +config SND_SOC_SOF_NOCODEC + tristate "SOF nocodec mode Support" + help + This adds support for a dummy/nocodec machine driver fallback + option if no known codec is detected. This is typically only + enabled for developers or devices where the sound card is + controlled externally + Say Y if you need this nocodec fallback option + If unsure select "N". + +config SND_SOC_SOF_DEBUG + bool "SOF debugging features" + help + This option can be used to enable or disable individual SOF firmware + and driver debugging options. + Say Y if you are debugging SOF FW or drivers. + If unsure select "N". + +if SND_SOC_SOF_DEBUG + +config SND_SOC_SOF_FORCE_NOCODEC_MODE + bool "SOF force nocodec Mode" + depends on SND_SOC_SOF_NOCODEC + help + This forces SOF to use dummy/nocodec as machine driver, even + though there is a codec detected on the real platform. This is + typically only enabled for developers for debug purposes, before + codec/machine driver is ready, or to exclude the impact of those + drivers + Say Y if you need this force nocodec mode option + If unsure select "N". + +config SND_SOC_SOF_DEBUG_XRUN_STOP + bool "SOF stop on XRUN" + help + This option forces PCMs to stop on any XRUN event. This is useful to + preserve any trace data ond pipeline status prior to the XRUN. + Say Y if you are debugging SOF FW pipeline XRUNs. + If unsure select "N". + +config SND_SOC_SOF_DEBUG_VERBOSE_IPC + bool "SOF verbose IPC logs" + help + This option enables more verbose IPC logs, with command types in + human-readable form instead of just 32-bit hex dumps. This is useful + if you are trying to debug IPC with the DSP firmware. + If unsure select "N". + +config SND_SOC_SOF_DEBUG_FORCE_IPC_POSITION + bool "SOF force to use IPC for position update on SKL+" + help + This option force to handle stream position update IPCs and run pcm + elapse to inform ALSA about that, on platforms (e.g. Intel SKL+) that + with other approach (e.g. HDAC DPIB/posbuf) to elapse PCM. + On platforms (e.g. Intel SKL-) where position update IPC is the only + one choice, this setting won't impact anything. + if you are trying to debug pointer update with position IPCs or where + DPIB/posbuf is not ready, select "Y". + If unsure select "N". + +config SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE + bool "SOF enable debugfs caching" + help + This option enables caching of debugfs + memory -> DSP resource (memory, register, etc) + before the audio DSP is suspended. This will increase the suspend + latency and therefore should be used for debug purposes only. + Say Y if you want to enable caching the memory windows. + If unsure, select "N". + +endif ## SND_SOC_SOF_DEBUG + +endif ## SND_SOC_SOF_OPTIONS + +config SND_SOC_SOF + tristate + select SND_SOC_TOPOLOGY + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + The selection is made at the top level and does not exactly follow + module dependencies but since the module or built-in type is decided + at the top level it doesn't matter. + +config SND_SOC_SOF_PROBE_WORK_QUEUE + bool + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + When selected, the probe is handled in two steps, for example to + avoid lockdeps if request_module is used in the probe. + +source "sound/soc/sof/intel/Kconfig" +source "sound/soc/sof/xtensa/Kconfig" + +endif diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile new file mode 100644 index 000000000000..8f14c9d2950b --- /dev/null +++ b/sound/soc/sof/Makefile @@ -0,0 +1,18 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) + +snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ + control.o trace.o utils.o + +snd-sof-pci-objs := sof-pci-dev.o +snd-sof-acpi-objs := sof-acpi-dev.o +snd-sof-nocodec-objs := nocodec.o + +obj-$(CONFIG_SND_SOC_SOF) += snd-sof.o +obj-$(CONFIG_SND_SOC_SOF_NOCODEC) += snd-sof-nocodec.o + + +obj-$(CONFIG_SND_SOC_SOF_ACPI) += sof-acpi-dev.o +obj-$(CONFIG_SND_SOC_SOF_PCI) += sof-pci-dev.o + +obj-$(CONFIG_SND_SOC_SOF_INTEL_TOPLEVEL) += intel/ +obj-$(CONFIG_SND_SOC_SOF_XTENSA) += xtensa/ diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig new file mode 100644 index 000000000000..32ee0fabab92 --- /dev/null +++ b/sound/soc/sof/intel/Kconfig @@ -0,0 +1,230 @@ +config SND_SOC_SOF_INTEL_TOPLEVEL + bool "SOF support for Intel audio DSPs" + depends on X86 || COMPILE_TEST + help + This adds support for Sound Open Firmware for Intel(R) platforms. + Say Y if you have such a device. + If unsure select "N". + +if SND_SOC_SOF_INTEL_TOPLEVEL + +config SND_SOC_SOF_INTEL_ACPI + tristate + select SND_SOC_SOF_BAYTRAIL if SND_SOC_SOF_BAYTRAIL_SUPPORT + select SND_SOC_SOF_BROADWELL if SND_SOC_SOF_BROADWELL_SUPPORT + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_INTEL_PCI + tristate + select SND_SOC_SOF_MERRIFIELD if SND_SOC_SOF_MERRIFIELD_SUPPORT + select SND_SOC_SOF_APOLLOLAKE if SND_SOC_SOF_APOLLOLAKE_SUPPORT + select SND_SOC_SOF_GEMINILAKE if SND_SOC_SOF_GEMINILAKE_SUPPORT + select SND_SOC_SOF_CANNONLAKE if SND_SOC_SOF_CANNONLAKE_SUPPORT + select SND_SOC_SOF_COFFEELAKE if SND_SOC_SOF_COFFEELAKE_SUPPORT + select SND_SOC_SOF_ICELAKE if SND_SOC_SOF_ICELAKE_SUPPORT + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_INTEL_HIFI_EP_IPC + tristate + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_INTEL_ATOM_HIFI_EP + tristate + select SND_SOC_INTEL_COMMON + select SND_SOC_SOF_INTEL_HIFI_EP_IPC + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_INTEL_COMMON + tristate + select SND_SOC_ACPI_INTEL_MATCH + select SND_SOC_SOF_XTENSA + select SND_SOC_INTEL_MACH + select SND_SOC_ACPI if ACPI + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +if SND_SOC_SOF_INTEL_ACPI + +config SND_SOC_SOF_BAYTRAIL_SUPPORT + bool "SOF support for Baytrail, Braswell and Cherrytrail" + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Baytrail, Braswell or Cherrytrail processors. + Say Y if you have such a device. + If unsure select "N". + +config SND_SOC_SOF_BAYTRAIL + tristate + select SND_SOC_SOF_INTEL_ATOM_HIFI_EP + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_BROADWELL_SUPPORT + bool "SOF support for Broadwell" + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Broadwell processors. + Say Y if you have such a device. + If unsure select "N". + +config SND_SOC_SOF_BROADWELL + tristate + select SND_SOC_SOF_INTEL_COMMON + select SND_SOC_SOF_INTEL_HIFI_EP_IPC + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +endif ## SND_SOC_SOF_INTEL_ACPI + +if SND_SOC_SOF_INTEL_PCI + +config SND_SOC_SOF_MERRIFIELD_SUPPORT + bool "SOF support for Tangier/Merrifield" + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Tangier/Merrifield processors. + Say Y if you have such a device. + If unsure select "N". + +config SND_SOC_SOF_MERRIFIELD + tristate + select SND_SOC_SOF_INTEL_ATOM_HIFI_EP + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_APOLLOLAKE_SUPPORT + bool "SOF support for Apollolake" + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Apollolake processors. + Say Y if you have such a device. + If unsure select "N". + +config SND_SOC_SOF_APOLLOLAKE + tristate + select SND_SOC_SOF_HDA_COMMON + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_GEMINILAKE_SUPPORT + bool "SOF support for GeminiLake" + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Geminilake processors. + Say Y if you have such a device. + If unsure select "N". + +config SND_SOC_SOF_GEMINILAKE + tristate + select SND_SOC_SOF_HDA_COMMON + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_CANNONLAKE_SUPPORT + bool "SOF support for Cannonlake" + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Cannonlake processors. + Say Y if you have such a device. + If unsure select "N". + +config SND_SOC_SOF_CANNONLAKE + tristate + select SND_SOC_SOF_HDA_COMMON + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_COFFEELAKE_SUPPORT + bool "SOF support for CoffeeLake" + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Coffeelake processors. + Say Y if you have such a device. + If unsure select "N". + +config SND_SOC_SOF_COFFEELAKE + tristate + select SND_SOC_SOF_HDA_COMMON + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_ICELAKE_SUPPORT + bool "SOF support for Icelake" + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Icelake processors. + Say Y if you have such a device. + If unsure select "N". + +config SND_SOC_SOF_ICELAKE + tristate + select SND_SOC_SOF_HDA_COMMON + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_HDA_COMMON + tristate + select SND_SOC_SOF_INTEL_COMMON + select SND_SOC_SOF_HDA_LINK_BASELINE + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +if SND_SOC_SOF_HDA_COMMON + +config SND_SOC_SOF_HDA_LINK + bool "SOF support for HDA Links(HDA/HDMI)" + depends on SND_SOC_SOF_NOCODEC=n + select SND_SOC_SOF_PROBE_WORK_QUEUE + help + This adds support for HDA links(HDA/HDMI) with Sound Open Firmware + for Intel(R) platforms. + Say Y if you want to enable HDA links with SOF. + If unsure select "N". + +config SND_SOC_SOF_HDA_AUDIO_CODEC + bool "SOF support for HDAudio codecs" + depends on SND_SOC_SOF_HDA_LINK + help + This adds support for HDAudio codecs with Sound Open Firmware + for Intel(R) platforms. + Say Y if you want to enable HDAudio codecs with SOF. + If unsure select "N". + +endif ## SND_SOC_SOF_HDA_COMMON + +config SND_SOC_SOF_HDA_LINK_BASELINE + tristate + select SND_SOC_SOF_HDA if SND_SOC_SOF_HDA_LINK + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_HDA + tristate + select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK + select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +endif ## SND_SOC_SOF_INTEL_PCI + +endif ## SND_SOC_SOF_INTEL_TOPLEVEL diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile new file mode 100644 index 000000000000..b8f58e006e29 --- /dev/null +++ b/sound/soc/sof/intel/Makefile @@ -0,0 +1,19 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) + +snd-sof-intel-byt-objs := byt.o +snd-sof-intel-bdw-objs := bdw.o + +snd-sof-intel-ipc-objs := intel-ipc.o + +snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \ + hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \ + hda-dai.o hda-bus.o \ + apl.o cnl.o + +snd-sof-intel-hda-objs := hda-codec.o + +obj-$(CONFIG_SND_SOC_SOF_INTEL_ATOM_HIFI_EP) += snd-sof-intel-byt.o +obj-$(CONFIG_SND_SOC_SOF_BROADWELL) += snd-sof-intel-bdw.o +obj-$(CONFIG_SND_SOC_SOF_INTEL_HIFI_EP_IPC) += snd-sof-intel-ipc.o +obj-$(CONFIG_SND_SOC_SOF_HDA_COMMON) += snd-sof-intel-hda-common.o +obj-$(CONFIG_SND_SOC_SOF_HDA) += snd-sof-intel-hda.o -- cgit v1.2.3-59-g8ed1b From f35bf70f61d389754fafd7fce75efbb3bd2eea87 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 12 Apr 2019 11:09:03 -0500 Subject: ASoC: Intel: Make sure BDW based machine drivers build for SOF BDW uses hard coded IPC calls to set SSP, not needed in SOF as SSP is configured via topology. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw-rt5677.c | 4 ++++ sound/soc/intel/boards/broadwell.c | 4 ++++ 2 files changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 1844c88ea4e2..6520a8ea5537 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -180,6 +180,7 @@ static const struct snd_soc_ops bdw_rt5677_ops = { .hw_params = bdw_rt5677_hw_params, }; +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) static int bdw_rt5677_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); @@ -198,6 +199,7 @@ static int bdw_rt5677_rtd_init(struct snd_soc_pcm_runtime *rtd) return 0; } +#endif static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd) { @@ -265,7 +267,9 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) .init = bdw_rt5677_rtd_init, +#endif .trigger = { SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index b86c746d9b7a..0f18f8964f51 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -131,6 +131,7 @@ static const struct snd_soc_ops broadwell_rt286_ops = { .hw_params = broadwell_rt286_hw_params, }; +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); @@ -149,6 +150,7 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) return 0; } +#endif /* broadwell digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link broadwell_rt286_dais[] = { @@ -161,7 +163,9 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) .init = broadwell_rtd_init, +#endif .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, .dpcm_capture = 1, -- cgit v1.2.3-59-g8ed1b From 10b02b53a99865f2005497c3933342c1f7946cd6 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 12 Apr 2019 11:09:04 -0500 Subject: ASoC: Intel: select relevant machine drivers for SOF SOF can only support specific machine drivers, handle dependencies Signed-off-by: Pierre-Louis Bossart Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 22 +++++++++++++++------- 1 file changed, 15 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 12d6b73e9531..bf6d7a11f94a 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -1,6 +1,6 @@ menuconfig SND_SOC_INTEL_MACH bool "Intel Machine drivers" - depends on SND_SOC_INTEL_SST_TOPLEVEL + depends on SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL help Intel ASoC Machine Drivers. If you have a Intel machine that has an audio controller with a DSP and I2S or DMIC port, then @@ -24,6 +24,10 @@ config SND_SOC_INTEL_HASWELL_MACH Say Y or m if you have such a device. If unsure select "N". +endif ## SND_SOC_INTEL_HASWELL + +if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL + config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM && GPIOLIB @@ -43,7 +47,7 @@ config SND_SOC_INTEL_BROADWELL_MACH Ultrabook platforms. Say Y or m if you have such a device. This is a recommended option. If unsure select "N". -endif ## SND_SOC_INTEL_HASWELL +endif ## SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL if SND_SOC_INTEL_BAYTRAIL @@ -68,7 +72,7 @@ config SND_SOC_INTEL_BYT_RT5640_MACH endif ## SND_SOC_INTEL_BAYTRAIL -if SND_SST_ATOM_HIFI2_PLATFORM +if SND_SST_ATOM_HIFI2_PLATFORM || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "Baytrail and Baytrail-CR with RT5640 codec" @@ -158,6 +162,10 @@ config SND_SOC_INTEL_BYT_CHT_ES8316_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +endif ## SND_SST_ATOM_HIFI2_PLATFORM || SND_SOC_SOF_BAYTRAIL + +if SND_SST_ATOM_HIFI2_PLATFORM + config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH tristate "Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)" depends on X86_INTEL_LPSS && I2C && ACPI @@ -314,7 +322,7 @@ config SND_SOC_INTEL_KBL_RT5660_MACH endif ## SND_SOC_INTEL_KBL -if SND_SOC_INTEL_GLK +if SND_SOC_INTEL_GLK || (SND_SOC_SOF_GEMINILAKE && SND_SOC_SOF_HDA_LINK) config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" @@ -330,9 +338,9 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". -endif ## SND_SOC_INTEL_GLK +endif ## SND_SOC_INTEL_GLK || (SND_SOC_SOF_GEMINILAKE && SND_SOC_SOF_HDA_LINK) -if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC +if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "SKL/KBL/BXT/APL with HDA Codecs" @@ -344,6 +352,6 @@ config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". -endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC +endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC endif ## SND_SOC_INTEL_MACH -- cgit v1.2.3-59-g8ed1b From 607ca3bd220f4022e6f5356026b19dafc363863a Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 26 Apr 2019 16:35:41 +0800 Subject: ALSA: hda/realtek - EAPD turn on later Let EAPD turn on after set pin output. [ NOTE: This change is supposed to reduce the possible click noises at (runtime) PM resume. The functionality should be same (i.e. the verbs are executed correctly) no matter which order is, so this should be safe to apply for all codecs -- tiwai ] Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 810479766090..674ddff65bef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -803,11 +803,10 @@ static int alc_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); + snd_hda_gen_init(codec); alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); - snd_hda_gen_init(codec); - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); return 0; -- cgit v1.2.3-59-g8ed1b From 4ab1ae3453c92940dc752e2c98aa494d3eaedc09 Mon Sep 17 00:00:00 2001 From: Fuqian Huang Date: Fri, 26 Apr 2019 11:16:24 +0800 Subject: ALSA: gus: fix misuse of %x Pointers should be printed with %p or %px rather than cast to long type and printed with %lx. Drop the address printing. Signed-off-by: Fuqian Huang Signed-off-by: Takashi Iwai --- sound/isa/gus/gus_mem.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 4ac76f46dd76..d708ae1525e4 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -306,7 +306,7 @@ static void snd_gf1_mem_info_read(struct snd_info_entry *entry, used = 0; for (block = alloc->first, i = 0; block; block = block->next, i++) { used += block->size; - snd_iprintf(buffer, "Block %i at 0x%lx onboard 0x%x size %i (0x%x):\n", i, (long) block, block->ptr, block->size, block->size); + snd_iprintf(buffer, "Block %i onboard 0x%x size %i (0x%x):\n", i, block->ptr, block->size, block->size); if (block->share || block->share_id[0] || block->share_id[1] || block->share_id[2] || block->share_id[3]) -- cgit v1.2.3-59-g8ed1b From cb5173594d50c72b7bfa14113dfc5084b4d2f726 Mon Sep 17 00:00:00 2001 From: Wenwen Wang Date: Sat, 27 Apr 2019 01:06:46 -0500 Subject: ALSA: usb-audio: Fix a memory leak bug In parse_audio_selector_unit(), the string array 'namelist' is allocated through kmalloc_array(), and each string pointer in this array, i.e., 'namelist[]', is allocated through kmalloc() in the following for loop. Then, a control instance 'kctl' is created by invoking snd_ctl_new1(). If an error occurs during the creation process, the string array 'namelist', including all string pointers in the array 'namelist[]', should be freed, before the error code ENOMEM is returned. However, the current code does not free 'namelist[]', resulting in memory leaks. To fix the above issue, free all string pointers 'namelist[]' in a loop. Signed-off-by: Wenwen Wang Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c095d9751924..e003b5e7b01a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2675,6 +2675,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, kctl = snd_ctl_new1(&mixer_selectunit_ctl, cval); if (! kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); + for (i = 0; i < desc->bNrInPins; i++) + kfree(namelist[i]); kfree(namelist); kfree(cval); return -ENOMEM; -- cgit v1.2.3-59-g8ed1b From 4d95c51776b2edb4d4ebcea00b6e5a1fe538ce66 Mon Sep 17 00:00:00 2001 From: Bard liao Date: Sun, 28 Apr 2019 04:53:39 +0800 Subject: ALSA: hda: fix unregister device twice on ASoC driver snd_hda_codec_device_new() is used by both legacy HDA and ASoC driver. However, we will call snd_hdac_device_unregister() in snd_hdac_ext_bus_device_remove() for ASoC device. This patch uses the type flag in hdac_device struct to determine is it a ASoC device or legacy HDA device and call snd_hdac_device_unregister() in snd_hda_codec_dev_free() only if it is a legacy HDA device. Signed-off-by: Bard liao Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 701a69d856f5..b20eb7fc83eb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -832,7 +832,13 @@ static int snd_hda_codec_dev_free(struct snd_device *device) struct hda_codec *codec = device->device_data; codec->in_freeing = 1; - snd_hdac_device_unregister(&codec->core); + /* + * snd_hda_codec_device_new() is used by legacy HDA and ASoC driver. + * We can't unregister ASoC device since it will be unregistered in + * snd_hdac_ext_bus_device_remove(). + */ + if (codec->core.type == HDA_DEV_LEGACY) + snd_hdac_device_unregister(&codec->core); codec_display_power(codec, false); put_device(hda_codec_dev(codec)); return 0; -- cgit v1.2.3-59-g8ed1b From b60ee2e281b6c67cbe0bb8412f3cae53903173c6 Mon Sep 17 00:00:00 2001 From: Bard liao Date: Sun, 28 Apr 2019 04:53:40 +0800 Subject: ASoC: hdac_hda: overwrite hdev type to HDA_DEV_ASOC In ASoC driver, snd_hdac_device_register() will be called by snd_hdac_ext_bus_device_init() and snd_hdac_device_unregister() will called by snd_hdac_ext_bus_device_remove(). However when ASoC codec driver call snd_hda_codec_device_new() to create a new hda codec, it will assign snd_hda_codec_dev_free() to the dev_free ops and snd_hda_codec_dev_free() will call snd_hdac_device_unregister(). As a result, snd_hdac_device_unregister() will be called twice in ASoC driver. To prevent it, we use hdev type to determine if the hda codec is registered by legacy HDA driver or ASoC driver and unregister device in snd_hda_codec_dev_free() only if it is a legacy HDA device. This patch will overwrite the hdev type so that we can know it is a ASoC device. Signed-off-by: Bard liao Signed-off-by: Takashi Iwai --- sound/soc/codecs/hdac_hda.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index f889d94c8e3c..7d4940256914 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -328,6 +328,12 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) dev_err(&hdev->dev, "failed to create hda codec %d\n", ret); goto error_no_pm; } + /* + * Overwrite type to HDA_DEV_ASOC since it is a ASoC driver + * hda_codec.c will check this flag to determine if unregister + * device is needed. + */ + hdev->type = HDA_DEV_ASOC; /* * snd_hda_codec_device_new decrements the usage count so call get pm -- cgit v1.2.3-59-g8ed1b From cbb88db76a1536e02e93e5bd37ebbfbb6c4043a9 Mon Sep 17 00:00:00 2001 From: Wenwen Wang Date: Mon, 29 Apr 2019 12:45:40 -0500 Subject: ALSA: usx2y: fix a double free bug In usX2Y_In04_init(), a new urb is firstly created through usb_alloc_urb() and saved to 'usX2Y->In04urb'. Then, a buffer is allocated through kmalloc() and saved to 'usX2Y->In04Buf'. If the allocation of the buffer fails, the error code ENOMEM is returned after usb_free_urb(), which frees the created urb. However, the urb is actually freed at card->private_free callback, i.e., snd_usX2Y_card_private_free(). So the free operation here leads to a double free bug. To fix the above issue, simply remove usb_free_urb(). Signed-off-by: Wenwen Wang Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usbusx2y.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 9f7bbed2c0f0..e8687b3bd3c8 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -293,10 +293,8 @@ int usX2Y_In04_init(struct usX2Ydev *usX2Y) if (! (usX2Y->In04urb = usb_alloc_urb(0, GFP_KERNEL))) return -ENOMEM; - if (! (usX2Y->In04Buf = kmalloc(21, GFP_KERNEL))) { - usb_free_urb(usX2Y->In04urb); + if (! (usX2Y->In04Buf = kmalloc(21, GFP_KERNEL))) return -ENOMEM; - } init_waitqueue_head(&usX2Y->In04WaitQueue); usb_fill_int_urb(usX2Y->In04urb, usX2Y->dev, usb_rcvintpipe(usX2Y->dev, 0x4), -- cgit v1.2.3-59-g8ed1b From f495222e28275222ab6fd93813bd3d462e16d340 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Apr 2019 12:18:28 +0200 Subject: ALSA: hda - Register irq handler after the chip initialization Currently the IRQ handler in HD-audio controller driver is registered before the chip initialization. That is, we have some window opened between the azx_acquire_irq() call and the CORB/RIRB setup. If an interrupt is triggered in this small window, the IRQ handler may access to the uninitialized RIRB buffer, which leads to a NULL dereference Oops. This is usually no big problem since most of Intel chips do register the IRQ via MSI, and we've already fixed the order of the IRQ enablement and the CORB/RIRB setup in the former commit b61749a89f82 ("sound: enable interrupt after dma buffer initialization"), hence the IRQ won't be triggered in that room. However, some platforms use a shared IRQ, and this may allow the IRQ trigger by another source. Another possibility is the kdump environment: a stale interrupt might be present in there, the IRQ handler can be falsely triggered as well. For covering this small race, let's move the azx_acquire_irq() call after hda_intel_init_chip() call. Although this is a bit radical change, it can cover more widely than checking the CORB/RIRB setup locally in the callee side. Reported-by: Liwei Song Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2ec91085fa3e..789308f54785 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1788,9 +1788,6 @@ static int azx_first_init(struct azx *chip) chip->msi = 0; } - if (azx_acquire_irq(chip, 0) < 0) - return -EBUSY; - pci_set_master(pci); synchronize_irq(bus->irq); @@ -1904,6 +1901,9 @@ static int azx_first_init(struct azx *chip) return -ENODEV; } + if (azx_acquire_irq(chip, 0) < 0) + return -EBUSY; + strcpy(card->driver, "HDA-Intel"); strlcpy(card->shortname, driver_short_names[chip->driver_type], sizeof(card->shortname)); -- cgit v1.2.3-59-g8ed1b From 7b6531c5054e7804ccce25f389a2d4810357f5c9 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 17 Apr 2019 17:26:28 +0100 Subject: ASoC: hdmi-codec: fix spelling mistake "plalform" -> "platform" There is a spelling mistake in a dev_err message. Fix it. Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 35df73e42cbc..b9d9dde9fbaf 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -757,7 +757,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) dev_dbg(dev, "%s()\n", __func__); if (!hcd) { - dev_err(dev, "%s: No plalform data\n", __func__); + dev_err(dev, "%s: No platform data\n", __func__); return -EINVAL; } -- cgit v1.2.3-59-g8ed1b From f2a1fdb50d3187266a91bcf99b504e99695772a6 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Wed, 17 Apr 2019 23:29:32 +0800 Subject: ASoC: Mediatek: MT8183: Fix build err while CONFIG_I2C set to module During randconfig builds, I occasionally run into an invalid configuration WARNING: unmet direct dependencies detected for SND_SOC_TS3A227E Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=m] Selected by [y]: - SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_MT8183 [=y] sound/soc/codecs/ts3a227e.o: In function `ts3a227e_i2c_probe': ts3a227e.c:(.text+0x684): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/ts3a227e.o: In function `ts3a227e_driver_init': ts3a227e.c:(.init.text+0x18): undefined reference to `i2c_register_driver' sound/soc/codecs/ts3a227e.o: In function `ts3a227e_driver_exit': ts3a227e.c:(.exit.text+0x14): undefined reference to `i2c_del_driver' This patch add I2C dependency to fix this. Reported-by: Hulk Robot Fixes: ebbddc75bbe8 ("ASoC: Mediatek: MT8183: Add machine driver with DA7219") Signed-off-by: YueHaibing Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 874404bcccfd..f70b7109f2b6 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -118,6 +118,7 @@ config SND_SOC_MT8183 config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A tristate "ASoC Audio driver for MT8183 with MT6358 TS3A227E MAX98357A codec" + depends on I2C depends on SND_SOC_MT8183 select SND_SOC_MT6358 select SND_SOC_MAX98357A -- cgit v1.2.3-59-g8ed1b From bb24a31ed584b172507dc2fd991e19d110b5aae8 Mon Sep 17 00:00:00 2001 From: "Agrawal, Akshu" Date: Wed, 17 Apr 2019 10:05:13 +0000 Subject: ASoC: AMD: Configure wclk and bclk of master codec With CCF support in da7219, we can now set the correct rate of wclk and bclk. Signed-off-by: Akshu Agrawal Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 66 +++++++++++++++++++++++------------- 1 file changed, 42 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 8a619a75b3a9..16b0ea3a3d72 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -46,7 +46,8 @@ #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -static struct clk *da7219_dai_clk; +static struct clk *da7219_dai_wclk; +static struct clk *da7219_dai_bclk; extern bool bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) @@ -72,7 +73,8 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - da7219_dai_clk = clk_get(component->dev, "da7219-dai-bclk"); + da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); + da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | @@ -94,12 +96,15 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int da7219_clk_enable(struct snd_pcm_substream *substream) +static int da7219_clk_enable(struct snd_pcm_substream *substream, + int wclk_rate, int bclk_rate) { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - ret = clk_prepare_enable(da7219_dai_clk); + clk_set_rate(da7219_dai_wclk, wclk_rate); + clk_set_rate(da7219_dai_bclk, bclk_rate); + ret = clk_prepare_enable(da7219_dai_bclk); if (ret < 0) { dev_err(rtd->dev, "can't enable master clock %d\n", ret); return ret; @@ -110,7 +115,7 @@ static int da7219_clk_enable(struct snd_pcm_substream *substream) static void da7219_clk_disable(void) { - clk_disable_unprepare(da7219_dai_clk); + clk_disable_unprepare(da7219_dai_bclk); } static const unsigned int channels[] = { @@ -151,7 +156,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_SP_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) @@ -173,12 +178,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; - return da7219_clk_enable(substream); -} - -static void cz_da7219_shutdown(struct snd_pcm_substream *substream) -{ - da7219_clk_disable(); + return 0; } static int cz_max_startup(struct snd_pcm_substream *substream) @@ -199,12 +199,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); -} - -static void cz_max_shutdown(struct snd_pcm_substream *substream) -{ - da7219_clk_disable(); + return 0; } static int cz_dmic0_startup(struct snd_pcm_substream *substream) @@ -225,7 +220,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->cap_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_dmic1_startup(struct snd_pcm_substream *substream) @@ -247,10 +242,28 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; - return da7219_clk_enable(substream); + return 0; } -static void cz_dmic_shutdown(struct snd_pcm_substream *substream) +static int cz_da7219_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int wclk, bclk; + + wclk = params_rate(params); + bclk = wclk * params_channels(params) * + snd_pcm_format_width(params_format(params)); + /* ADAU7002 spec: "The ADAU7002 requires a BCLK rate + * that is minimum of 64x the LRCLK sample rate." + * DA7219 is the only clk source so for all codecs + * we have to limit bclk to 64X lrclk. + */ + if (bclk < (wclk * 64)) + bclk = wclk * 64; + return da7219_clk_enable(substream, wclk, bclk); +} + +static void cz_da7219_shutdown(struct snd_pcm_substream *substream) { da7219_clk_disable(); } @@ -258,26 +271,31 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, - .shutdown = cz_max_shutdown, + .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, - .shutdown = cz_dmic_shutdown, + .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, - .shutdown = cz_dmic_shutdown, + .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { -- cgit v1.2.3-59-g8ed1b From 5011454ee34a84700e9a8c7eb1138efceea9d662 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 17 Apr 2019 14:31:40 +0300 Subject: ASoC: pcm3168a: Enable TDM support for DSP_A/B modes The 24-bit TDM mode also applies to DSP_A and DSP_B modes. Most dais on the SoC side can not interpret I2S/Left_j with other than 2 channels of audio. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3168a.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 08d3fe192e65..e0d5839fe1a7 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -457,13 +457,16 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, if (chan > 2) { switch (fmt) { case PCM3168A_FMT_I2S: + case PCM3168A_FMT_DSP_A: fmt = PCM3168A_FMT_I2S_TDM; break; case PCM3168A_FMT_LEFT_J: + case PCM3168A_FMT_DSP_B: fmt = PCM3168A_FMT_LEFT_J_TDM; break; default: - dev_err(component->dev, "TDM is supported under I2S/Left_J only\n"); + dev_err(component->dev, + "TDM is supported under DSP/I2S/Left_J only\n"); return -EINVAL; } } @@ -526,6 +529,8 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, break; case PCM3168A_FMT_LEFT_J: case PCM3168A_FMT_I2S: + case PCM3168A_FMT_DSP_A: + case PCM3168A_FMT_DSP_B: sample_min = 24; channel_max = channel_maxs[tx]; break; -- cgit v1.2.3-59-g8ed1b From a0c34c7629bee46ffd8121987d27df25a6433cc7 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Wed, 17 Apr 2019 23:09:15 +0800 Subject: ASoC: fsl_micfil: Remove set but not used variable 'osr' Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/fsl/fsl_micfil.c: In function 'get_clk_div': sound/soc/fsl/fsl_micfil.c:154:6: warning: variable 'osr' set but not used [-Wunused-but-set-variable] It is never used since introduction in commit 47a70e6fc9a8 ("ASoC: Add MICFIL SoC Digital Audio Interface driver.") Signed-off-by: YueHaibing Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_micfil.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 40c07e756481..f7f2d29f1bfe 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -151,12 +151,9 @@ static inline int get_clk_div(struct fsl_micfil *micfil, { u32 ctrl2_reg; long mclk_rate; - int osr; int clk_div; regmap_read(micfil->regmap, REG_MICFIL_CTRL2, &ctrl2_reg); - osr = 16 - ((ctrl2_reg & MICFIL_CTRL2_CICOSR_MASK) - >> MICFIL_CTRL2_CICOSR_SHIFT); mclk_rate = clk_get_rate(micfil->mclk); -- cgit v1.2.3-59-g8ed1b From 83b4f50ca2b2e93346195b51f58e8089f9f35c0b Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Wed, 17 Apr 2019 23:01:57 +0800 Subject: ASoC: tlv320aic32x4: Remove set but not used variable 'mclk_rate' Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_setup_clocks': sound/soc/codecs/tlv320aic32x4.c:669:16: warning: variable 'mclk_rate' set but not used [-Wunused-but-set-variable] It is not used since introduction in commit 96c3bb00239d ("ASoC: tlv320aic32x4: Dynamically Determine Clocking") Signed-off-by: YueHaibing Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 6edee05ff9f0..83608f386aef 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -684,9 +684,8 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, u8 madc, nadc, mdac, ndac, max_nadc, min_mdac, max_ndac; u8 dosr_increment; u16 max_dosr, min_dosr; - unsigned long mclk_rate, adc_clock_rate, dac_clock_rate; + unsigned long adc_clock_rate, dac_clock_rate; int ret; - struct clk *mclk; struct clk_bulk_data clocks[] = { { .id = "pll" }, @@ -700,9 +699,6 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, if (ret) return ret; - mclk = clk_get_parent(clocks[1].clk); - mclk_rate = clk_get_rate(mclk); - if (sample_rate <= 48000) { aosr = 128; adc_resource_class = 6; -- cgit v1.2.3-59-g8ed1b From a1a86e1bd4a87fd09171fd8555fe7490917e4e94 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Tue, 16 Apr 2019 22:52:51 +0800 Subject: ASoC: amd: acp3x: Make acp3x_dai_i2s_ops static Fix sparse warning: sound/soc/amd/raven/acp3x-pcm-dma.c:561:24: warning: symbol 'acp3x_dai_i2s_ops' was not declared. Should it be static? Reported-by: Hulk Robot Signed-off-by: YueHaibing Signed-off-by: Mark Brown --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 1a2e15ff1456..9775bda2a4ca 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -558,7 +558,7 @@ static int acp3x_dai_i2s_trigger(struct snd_pcm_substream *substream, return ret; } -struct snd_soc_dai_ops acp3x_dai_i2s_ops = { +static struct snd_soc_dai_ops acp3x_dai_i2s_ops = { .hw_params = acp3x_dai_i2s_hwparams, .trigger = acp3x_dai_i2s_trigger, .set_fmt = acp3x_dai_i2s_set_fmt, -- cgit v1.2.3-59-g8ed1b From ea751227c813ab833609afecfeedaf0aa26f327e Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 16 Apr 2019 15:12:23 +0200 Subject: ASoC: imx: fix fiq dependencies During randconfig builds, I occasionally run into an invalid configuration of the freescale FIQ sound support: WARNING: unmet direct dependencies detected for SND_SOC_IMX_PCM_FIQ Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m] Selected by [y]: - SND_SOC_FSL_SPDIF [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m]!=n && (MXC_TZIC [=n] || MXC_AVIC [=y]) sound/soc/fsl/imx-ssi.o: In function `imx_ssi_remove': imx-ssi.c:(.text+0x28): undefined reference to `imx_pcm_fiq_exit' sound/soc/fsl/imx-ssi.o: In function `imx_ssi_probe': imx-ssi.c:(.text+0xa64): undefined reference to `imx_pcm_fiq_init' The Kconfig warning is a result of the symbol being defined inside of the "if SND_IMX_SOC" block, and is otherwise harmless. The link error is more tricky and happens with SND_SOC_IMX_SSI=y, which may or may not imply FIQ support. However, if SND_SOC_FSL_SSI is set to =m at the same time, that selects SND_SOC_IMX_PCM_FIQ as a loadable module dependency, which then causes a link failure from imx-ssi. The solution here is to make SND_SOC_IMX_PCM_FIQ built-in whenever one of its potential users is built-in. Fixes: ff40260f79dc ("ASoC: fsl: refine DMA/FIQ dependencies") Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d87c842806bd..55ed47c599e2 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -189,16 +189,17 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC +config SND_SOC_IMX_PCM_FIQ + tristate + default y if SND_SOC_IMX_SSI=y && (SND_SOC_FSL_SSI=m || SND_SOC_FSL_SPDIF=m) && (MXC_TZIC || MXC_AVIC) + select FIQ + if SND_IMX_SOC config SND_SOC_IMX_SSI tristate select SND_SOC_FSL_UTILS -config SND_SOC_IMX_PCM_FIQ - tristate - select FIQ - comment "SoC Audio support for Freescale i.MX boards:" config SND_MXC_SOC_WM1133_EV1 -- cgit v1.2.3-59-g8ed1b From 6f547c96b45de0d42de91ef56c7d291aa6d3c88f Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Tue, 16 Apr 2019 22:47:18 +0800 Subject: ASoC: atmel: tse850: Make some functions static Fix sparse warnings: sound/soc/atmel/tse850-pcm5142.c:120:5: warning: symbol 'tse850_get_mix' was not declared. Should it be static? sound/soc/atmel/tse850-pcm5142.c:132:5: warning: symbol 'tse850_put_mix' was not declared. Should it be static? sound/soc/atmel/tse850-pcm5142.c:154:5: warning: symbol 'tse850_get_ana' was not declared. Should it be static? sound/soc/atmel/tse850-pcm5142.c:187:5: warning: symbol 'tse850_put_ana' was not declared. Should it be static? Reported-by: Hulk Robot Signed-off-by: YueHaibing Signed-off-by: Mark Brown --- sound/soc/atmel/tse850-pcm5142.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 214adcad5419..ae445184614a 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -117,8 +117,8 @@ static int tse850_put_mux2(struct snd_kcontrol *kctrl, return snd_soc_dapm_put_enum_double(kctrl, ucontrol); } -int tse850_get_mix(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_value *ucontrol) +static int tse850_get_mix(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kctrl); struct snd_soc_card *card = dapm->card; @@ -129,8 +129,8 @@ int tse850_get_mix(struct snd_kcontrol *kctrl, return 0; } -int tse850_put_mix(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_value *ucontrol) +static int tse850_put_mix(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kctrl); struct snd_soc_card *card = dapm->card; @@ -151,8 +151,8 @@ int tse850_put_mix(struct snd_kcontrol *kctrl, return 1; } -int tse850_get_ana(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_value *ucontrol) +static int tse850_get_ana(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kctrl); struct snd_soc_card *card = dapm->card; @@ -184,8 +184,8 @@ int tse850_get_ana(struct snd_kcontrol *kctrl, return 0; } -int tse850_put_ana(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_value *ucontrol) +static int tse850_put_ana(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kctrl); struct snd_soc_card *card = dapm->card; -- cgit v1.2.3-59-g8ed1b From 680ae69d52279474ecb204f0f7bae1f4d9361cbd Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Sun, 14 Apr 2019 22:14:50 +0300 Subject: ASoC: mpc5200_psc_i2s: Fix invalid license ID As the file had no other license notice/reference, it falls under the project license and therefore the proper SPDX id is: GPL-2.0-only Cc: Daniel Baluta Fixes: 864a8472c4412 ("ASoC: mpc5200_psc_i2s: Switch to SPDX identifier") Reported-by: Thomas Gleixner Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 6de97461ba25..9bc01f374b39 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL +// SPDX-License-Identifier: GPL-2.0-only // // Freescale MPC5200 PSC in I2S mode // ALSA SoC Digital Audio Interface (DAI) driver -- cgit v1.2.3-59-g8ed1b From ac097cac496f69e97083c6b128c5a209a85c6fcb Mon Sep 17 00:00:00 2001 From: Andra Danciu Date: Sun, 14 Apr 2019 22:14:49 +0300 Subject: ASoC: mpc5200_dma: Fix invalid license ID As the file had no other license notice/reference, it falls under the project license and therefore the proper SPDX id is: GPL-2.0-only Cc: Daniel Baluta Fixes: 1edfc2485d8dc ("ASoC: mpc5200_dma: Switch to SPDX identifier") Reported-by: Thomas Gleixner Signed-off-by: Andra Danciu Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 4396442c2fdd..ccf9301889fe 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -1,4 +1,4 @@ -// SPDX-License-Identifier: GPL +// SPDX-License-Identifier: GPL-2.0-only // // Freescale MPC5200 PSC DMA // ALSA SoC Platform driver -- cgit v1.2.3-59-g8ed1b From 164a263bf8d003e4cbb197d52b74d26df72604d7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Apr 2019 17:38:00 +0100 Subject: ASoC: Intel: Make boards more available for compile test The Intel boards have very strict dependencies which make them less available for compile test than is desirable, with requirements for specific drivers that are only needed at runtime but not at build time. Relax this a bit if COMPILE_TEST is enabled to improve build coverage for these drivers. Signed-off-by: Mark Brown Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 79 +++++++++++++++++++++++++++++------------- 1 file changed, 54 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index bf6d7a11f94a..fa0c88311559 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -16,7 +16,9 @@ if SND_SOC_INTEL_HASWELL config SND_SOC_INTEL_HASWELL_MACH tristate "Haswell Lynxpoint" - depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM + depends on I2C + depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5640 help This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell @@ -30,7 +32,10 @@ if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" - depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM && GPIOLIB + depends on I2C + depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST + depends on GPIOLIB || COMPILE_TEST + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5677 help This adds support for Intel Broadwell platform based boards with @@ -40,7 +45,9 @@ config SND_SOC_INTEL_BDW_RT5677_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "Broadwell Wildcatpoint" - depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM + depends on I2C + depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT286 help This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell @@ -53,7 +60,8 @@ if SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "Baytrail with MAX98090 codec" - depends on X86_INTEL_LPSS && I2C + depends on I2C + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_MAX98090 help This adds audio driver for Intel Baytrail platform based boards @@ -63,7 +71,8 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "Baytrail with RT5640 codec" - depends on X86_INTEL_LPSS && I2C + depends on I2C + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5640 help This adds audio driver for Intel Baytrail platform based boards @@ -76,7 +85,8 @@ if SND_SST_ATOM_HIFI2_PLATFORM || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "Baytrail and Baytrail-CR with RT5640 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5640 help @@ -87,7 +97,8 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "Baytrail and Baytrail-CR with RT5651 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5651 help @@ -98,7 +109,8 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5670 help @@ -109,7 +121,8 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5645 help @@ -120,7 +133,8 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH tristate "Cherrytrail & Braswell with MAX98090 & TI codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_MAX98090 select SND_SOC_TS3A227E help @@ -131,7 +145,8 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH config SND_SOC_INTEL_CHT_BSW_NAU8824_MACH tristate "Cherrytrail & Braswell with NAU88L24 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_NAU8824 help @@ -142,7 +157,8 @@ config SND_SOC_INTEL_CHT_BSW_NAU8824_MACH config SND_SOC_INTEL_BYT_CHT_DA7213_MACH tristate "Baytrail & Cherrytrail with DA7212/7213 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_DA7213 help @@ -153,7 +169,8 @@ config SND_SOC_INTEL_BYT_CHT_DA7213_MACH config SND_SOC_INTEL_BYT_CHT_ES8316_MACH tristate "Baytrail & Cherrytrail with ES8316 codec" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_ES8316 help @@ -168,7 +185,8 @@ if SND_SST_ATOM_HIFI2_PLATFORM config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH tristate "Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)" - depends on X86_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on X86_INTEL_LPSS || COMPILE_TEST help This adds support for ASoC machine driver for the MinnowBoard Max or Up boards and provides access to I2S signals on the Low-Speed @@ -184,7 +202,8 @@ if SND_SOC_INTEL_SKL config SND_SOC_INTEL_SKL_RT286_MACH tristate "SKL with RT286 I2S mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT286 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI @@ -196,7 +215,8 @@ config SND_SOC_INTEL_SKL_RT286_MACH config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH tristate "SKL with NAU88L25 and SSM4567 in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_NAU8825 select SND_SOC_SSM4567 select SND_SOC_DMIC @@ -209,7 +229,8 @@ config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH tristate "SKL with NAU88L25 and MAX98357A in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_NAU8825 select SND_SOC_MAX98357A select SND_SOC_DMIC @@ -226,7 +247,8 @@ if SND_SOC_INTEL_APL config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC @@ -240,7 +262,8 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH config SND_SOC_INTEL_BXT_RT298_MACH tristate "Broxton with RT298 I2S mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT298 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI @@ -257,7 +280,8 @@ if SND_SOC_INTEL_KBL config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH tristate "KBL with RT5663 and MAX98927 in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5663 select SND_SOC_MAX98927 select SND_SOC_DMIC @@ -271,7 +295,8 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SPI select SND_SOC_RT5663 select SND_SOC_RT5514 @@ -286,7 +311,8 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC @@ -298,7 +324,8 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH tristate "KBL with DA7219 and MAX98927 in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_DA7219 select SND_SOC_MAX98927 select SND_SOC_MAX98373 @@ -312,7 +339,8 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH config SND_SOC_INTEL_KBL_RT5660_MACH tristate "KBL with RT5660 in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5660 select SND_SOC_HDAC_HDMI help @@ -326,7 +354,8 @@ if SND_SOC_INTEL_GLK || (SND_SOC_SOF_GEMINILAKE && SND_SOC_SOF_HDA_LINK) config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" - depends on MFD_INTEL_LPSS && I2C && ACPI + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_RT5682 select SND_SOC_MAX98357A select SND_SOC_DMIC -- cgit v1.2.3-59-g8ed1b From d3692cb5c56daa2fb8046a8f4ebe1b2b8b22c895 Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Wed, 10 Apr 2019 12:27:31 -0500 Subject: ASoC: Intel: common: add Geminilake Realtek+Maxim machine driver entry This patch adds glk_rt5682_max98357a_i2s machine driver entry into machine table. Both Skylake and SOF platform drivers can use this machine drivers. Signed-off-by: Yong Zhi Signed-off-by: Naveen Manohar Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 3f2061475ae4..616eb09e78a0 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -31,6 +31,15 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .sof_fw_filename = "sof-glk.ri", .sof_tplg_filename = "sof-glk-da7219.tplg", }, + { + .id = "10EC5682", + .drv_name = "glk_rt5682_max98357a", + .fw_filename = "intel/dsp_fw_glk.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &glk_codecs, + .sof_fw_filename = "sof-glk.ri", + .sof_tplg_filename = "sof-glk-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_glk_machines); -- cgit v1.2.3-59-g8ed1b From 51dd97d1df5fb9ac58b9b358e63e67b530f6ae21 Mon Sep 17 00:00:00 2001 From: Kangjie Lu Date: Thu, 14 Mar 2019 22:48:32 -0500 Subject: ASoC: rt5645: fix a NULL pointer dereference devm_kcalloc() may fail and return NULL. The fix returns ENOMEM in case it fails to avoid NULL pointer dereference. Signed-off-by: Kangjie Lu Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9a0751978090..f842775dbf2c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3419,6 +3419,9 @@ static int rt5645_probe(struct snd_soc_component *component) RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s), GFP_KERNEL); + if (!rt5645->eq_param) + return -ENOMEM; + return 0; } -- cgit v1.2.3-59-g8ed1b From a2be42f18d409213bb7e7a736e3ef6ba005115bb Mon Sep 17 00:00:00 2001 From: Kangjie Lu Date: Thu, 14 Mar 2019 22:51:20 -0500 Subject: ASoC: cs43130: fix a NULL pointer dereference In case create_singlethread_workqueue fails, the fix returns -ENOMEM to avoid potential NULL pointer dereference. Signed-off-by: Kangjie Lu Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 3f7b255587e6..80d672710eae 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2322,6 +2322,8 @@ static int cs43130_probe(struct snd_soc_component *component) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); + if (!cs43130->wq) + return -ENOMEM; INIT_WORK(&cs43130->work, cs43130_imp_meas); } -- cgit v1.2.3-59-g8ed1b From 8bb678d75954e9e5b9ecabfe95b20f650fd91edb Mon Sep 17 00:00:00 2001 From: Viorel Suman Date: Wed, 10 Apr 2019 11:06:38 +0000 Subject: ASoC: imx-audmix: fix object reference leaks in probe Release the reference to the underlying device taken by of_find_device_by_node() call. Signed-off-by: Viorel Suman Reported-by: Julia Lawall Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmix.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 7983bd339c01..9aaf3e5b45b9 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -171,6 +171,7 @@ static int imx_audmix_probe(struct platform_device *pdev) np->full_name); return -EINVAL; } + put_device(&audmix_pdev->dev); num_dai = of_count_phandle_with_args(audmix_np, "dais", NULL); if (num_dai != FSL_AUDMIX_MAX_DAIS) { @@ -216,6 +217,7 @@ static int imx_audmix_probe(struct platform_device *pdev) dev_err(&pdev->dev, "failed to find SAI platform device\n"); return -EINVAL; } + put_device(&cpu_pdev->dev); dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s", fe_name_pref, args.np->full_name + 1); @@ -280,6 +282,8 @@ static int imx_audmix_probe(struct platform_device *pdev) dev_err(&pdev->dev, "failed to find SAI platform device\n"); return -EINVAL; } + put_device(&cpu_pdev->dev); + priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1"); if (IS_ERR(priv->cpu_mclk)) { ret = PTR_ERR(priv->cpu_mclk); -- cgit v1.2.3-59-g8ed1b From f2a36a78423ee83b1aac79956891fcad0e833885 Mon Sep 17 00:00:00 2001 From: Viorel Suman Date: Wed, 10 Apr 2019 11:06:36 +0000 Subject: ASoC: fsl_audmix: remove "model" attribute Use "of_device_id.data" to specify the machine driver instead of "model" DTS attribute. Signed-off-by: Viorel Suman Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_audmix.c | 43 +++++++++++++++++++++++-------------------- 1 file changed, 23 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index dabde0342c95..dc802d5c4ccd 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -445,13 +445,29 @@ static const struct regmap_config fsl_audmix_regmap_config = { .cache_type = REGCACHE_FLAT, }; +static const struct of_device_id fsl_audmix_ids[] = { + { + .compatible = "fsl,imx8qm-audmix", + .data = "imx-audmix", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, fsl_audmix_ids); + static int fsl_audmix_probe(struct platform_device *pdev) { struct fsl_audmix *priv; struct resource *res; + const char *mdrv; + const struct of_device_id *of_id; void __iomem *regs; int ret; - const char *sprop; + + of_id = of_match_device(fsl_audmix_ids, &pdev->dev); + if (!of_id || !of_id->data) + return -EINVAL; + + mdrv = of_id->data; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -487,19 +503,12 @@ static int fsl_audmix_probe(struct platform_device *pdev) return ret; } - sprop = of_get_property(pdev->dev.of_node, "model", NULL); - if (sprop) { - priv->pdev = platform_device_register_data(&pdev->dev, sprop, 0, - NULL, 0); - if (IS_ERR(priv->pdev)) { - ret = PTR_ERR(priv->pdev); - dev_err(&pdev->dev, - "failed to register platform %s: %d\n", sprop, - ret); - } - } else { - dev_err(&pdev->dev, "[model] attribute missing.\n"); - ret = -EINVAL; + priv->pdev = platform_device_register_data(&pdev->dev, mdrv, 0, NULL, + 0); + if (IS_ERR(priv->pdev)) { + ret = PTR_ERR(priv->pdev); + dev_err(&pdev->dev, "failed to register platform %s: %d\n", + mdrv, ret); } return ret; @@ -553,12 +562,6 @@ static const struct dev_pm_ops fsl_audmix_pm = { pm_runtime_force_resume) }; -static const struct of_device_id fsl_audmix_ids[] = { - { .compatible = "fsl,imx8qm-audmix", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, fsl_audmix_ids); - static struct platform_driver fsl_audmix_driver = { .probe = fsl_audmix_probe, .remove = fsl_audmix_remove, -- cgit v1.2.3-59-g8ed1b From 1df1e5457cd1e115bd279f7d12015a4834057f68 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Mon, 8 Apr 2019 18:47:28 +0800 Subject: ASoC: Mediatek: MT8183: change supported formats of DL2 and UL1 DL2 and UL1 are for BTSCO. They should only provide 16-bit, mono, 8kHz and 16kHz to userspace. Change the formats accordingly. Signed-off-by: Tzung-Bi Shih Signed-off-by: Mark Brown --- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 42 ++++++++++++++++++++++ 1 file changed, 42 insertions(+) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 681bedb524f9..4e44e5689d6f 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -69,6 +69,46 @@ mt8183_mt6358_ts3a227_max98357_dapm_routes[] = { {"IT6505_8CH", NULL, "TDM"}, }; +static int +mt8183_mt6358_ts3a227_max98357_bt_sco_startup( + struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 8000, 16000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const unsigned int channels[] = { + 1, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + return 0; +} + +static const struct snd_soc_ops mt8183_mt6358_ts3a227_max98357_bt_sco_ops = { + .startup = mt8183_mt6358_ts3a227_max98357_bt_sco_startup, +}; + static struct snd_soc_dai_link mt8183_mt6358_ts3a227_max98357_dai_links[] = { /* FE */ @@ -93,6 +133,7 @@ mt8183_mt6358_ts3a227_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8183_mt6358_ts3a227_max98357_bt_sco_ops, }, { .name = "Playback_3", @@ -115,6 +156,7 @@ mt8183_mt6358_ts3a227_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8183_mt6358_ts3a227_max98357_bt_sco_ops, }, { .name = "Capture_2", -- cgit v1.2.3-59-g8ed1b From d232591c9482ddb121b60782d2e7dac1c4880bdb Mon Sep 17 00:00:00 2001 From: Shunli Wang Date: Tue, 9 Apr 2019 17:13:37 +0800 Subject: ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3 This patch add the control path from UL2 or UL3 to I2S2. The patch is based on broonie tree "for-next" branch. Signed-off-by: Shunli Wang Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 38 ++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) (limited to 'sound') diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 43be51bf0329..1bc0fafe5e29 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -291,11 +291,15 @@ static struct snd_soc_dai_driver mt8183_memif_dai_driver[] = { static const struct snd_kcontrol_new memif_ul1_ch1_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH1", AFE_CONN21, I_ADDA_UL_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I2S0_CH1", AFE_CONN21, + I_I2S0_CH1, 1, 0), }; static const struct snd_kcontrol_new memif_ul1_ch2_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH2", AFE_CONN22, I_ADDA_UL_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I2S0_CH2", AFE_CONN21, + I_I2S0_CH2, 1, 0), }; static const struct snd_kcontrol_new memif_ul2_ch1_mix[] = { @@ -307,6 +311,8 @@ static const struct snd_kcontrol_new memif_ul2_ch1_mix[] = { I_DL2_CH1, 1, 0), SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH1", AFE_CONN5, I_DL3_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I2S2_CH1", AFE_CONN5, + I_I2S2_CH1, 1, 0), }; static const struct snd_kcontrol_new memif_ul2_ch2_mix[] = { @@ -318,16 +324,22 @@ static const struct snd_kcontrol_new memif_ul2_ch2_mix[] = { I_DL2_CH2, 1, 0), SOC_DAPM_SINGLE_AUTODISABLE("DL3_CH2", AFE_CONN6, I_DL3_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I2S2_CH2", AFE_CONN6, + I_I2S2_CH2, 1, 0), }; static const struct snd_kcontrol_new memif_ul3_ch1_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH1", AFE_CONN32, I_ADDA_UL_CH1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I2S2_CH1", AFE_CONN32, + I_I2S2_CH1, 1, 0), }; static const struct snd_kcontrol_new memif_ul3_ch2_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("ADDA_UL_CH2", AFE_CONN33, I_ADDA_UL_CH2, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I2S2_CH2", AFE_CONN33, + I_I2S2_CH2, 1, 0), }; static const struct snd_kcontrol_new memif_ul4_ch1_mix[] = { @@ -380,16 +392,22 @@ static const struct snd_soc_dapm_route mt8183_memif_routes[] = { {"UL1", NULL, "UL1_CH2"}, {"UL1_CH1", "ADDA_UL_CH1", "ADDA Capture"}, {"UL1_CH2", "ADDA_UL_CH2", "ADDA Capture"}, + {"UL1_CH1", "I2S0_CH1", "I2S0"}, + {"UL1_CH2", "I2S0_CH2", "I2S0"}, {"UL2", NULL, "UL2_CH1"}, {"UL2", NULL, "UL2_CH2"}, {"UL2_CH1", "ADDA_UL_CH1", "ADDA Capture"}, {"UL2_CH2", "ADDA_UL_CH2", "ADDA Capture"}, + {"UL2_CH1", "I2S2_CH1", "I2S2"}, + {"UL2_CH2", "I2S2_CH2", "I2S2"}, {"UL3", NULL, "UL3_CH1"}, {"UL3", NULL, "UL3_CH2"}, {"UL3_CH1", "ADDA_UL_CH1", "ADDA Capture"}, {"UL3_CH2", "ADDA_UL_CH2", "ADDA Capture"}, + {"UL3_CH1", "I2S2_CH1", "I2S2"}, + {"UL3_CH2", "I2S2_CH2", "I2S2"}, {"UL4", NULL, "UL4_CH1"}, {"UL4", NULL, "UL4_CH2"}, @@ -421,7 +439,9 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = DL1_HD_SFT, .agent_disable_reg = -1, + .agent_disable_shift = -1, .msb_reg = -1, + .msb_shift = -1, }, [MT8183_MEMIF_DL2] = { .name = "DL2", @@ -438,7 +458,9 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = DL2_HD_SFT, .agent_disable_reg = -1, + .agent_disable_shift = -1, .msb_reg = -1, + .msb_shift = -1, }, [MT8183_MEMIF_DL3] = { .name = "DL3", @@ -455,7 +477,9 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = DL3_HD_SFT, .agent_disable_reg = -1, + .agent_disable_shift = -1, .msb_reg = -1, + .msb_shift = -1, }, [MT8183_MEMIF_VUL2] = { .name = "VUL2", @@ -472,7 +496,9 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = VUL2_HD_SFT, .agent_disable_reg = -1, + .agent_disable_shift = -1, .msb_reg = -1, + .msb_shift = -1, }, [MT8183_MEMIF_AWB] = { .name = "AWB", @@ -489,7 +515,9 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = AWB_HD_SFT, .agent_disable_reg = -1, + .agent_disable_shift = -1, .msb_reg = -1, + .msb_shift = -1, }, [MT8183_MEMIF_AWB2] = { .name = "AWB2", @@ -506,7 +534,9 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = AWB2_HD_SFT, .agent_disable_reg = -1, + .agent_disable_shift = -1, .msb_reg = -1, + .msb_shift = -1, }, [MT8183_MEMIF_VUL12] = { .name = "VUL12", @@ -523,7 +553,9 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = VUL12_HD_SFT, .agent_disable_reg = -1, + .agent_disable_shift = -1, .msb_reg = -1, + .msb_shift = -1, }, [MT8183_MEMIF_MOD_DAI] = { .name = "MOD_DAI", @@ -540,7 +572,9 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = MOD_DAI_HD_SFT, .agent_disable_reg = -1, + .agent_disable_shift = -1, .msb_reg = -1, + .msb_shift = -1, }, [MT8183_MEMIF_HDMI] = { .name = "HDMI", @@ -553,10 +587,13 @@ static const struct mtk_base_memif_data memif_data[MT8183_MEMIF_NUM] = { .mono_reg = -1, .mono_shift = -1, .enable_reg = -1, /* control in tdm for sync start */ + .enable_shift = -1, .hd_reg = AFE_MEMIF_HD_MODE, .hd_shift = HDMI_HD_SFT, .agent_disable_reg = -1, + .agent_disable_shift = -1, .msb_reg = -1, + .msb_shift = -1, }, }; @@ -671,6 +708,7 @@ static const struct mtk_base_irq_data irq_data[MT8183_IRQ_NUM] = { .irq_cnt_shift = 0, .irq_cnt_maskbit = 0x3ffff, .irq_fs_reg = -1, + .irq_fs_shift = -1, .irq_fs_maskbit = -1, .irq_en_reg = AFE_IRQ_MCU_CON0, .irq_en_shift = IRQ8_MCU_ON_SFT, -- cgit v1.2.3-59-g8ed1b From a23e34c064b933cf5f24e443f86bbfdd624d2e8b Mon Sep 17 00:00:00 2001 From: Annaliese McDermond Date: Mon, 8 Apr 2019 21:41:59 -0700 Subject: ASoC: tlv320aic32x4: Fix potential uninitialized variable Fix compiler warning about uninitialized variable reported by Stephen Rothwell . Signed-off-by: Annaliese McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4-clk.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 9e4899eb1d8e..156c153c12ab 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -82,7 +82,7 @@ static int clk_aic32x4_pll_get_muldiv(struct clk_aic32x4 *pll, int ret; ret = regmap_read(pll->regmap, AIC32X4_PLLPR, &val); - if (ret) + if (ret < 0) return ret; settings->r = val & AIC32X4_PLL_R_MASK; settings->p = (val & AIC32X4_PLL_P_MASK) >> AIC32X4_PLL_P_SHIFT; -- cgit v1.2.3-59-g8ed1b From 62be484f7ad8443c393293a415392fbf3190c864 Mon Sep 17 00:00:00 2001 From: Viorel Suman Date: Wed, 10 Apr 2019 11:06:39 +0000 Subject: ASoC: fsl_audmix: cache pdev->dev pointer There should be no trouble to understand dev = pdev->dev. This can save some space to have more print info or save some wrapped lines. Signed-off-by: Viorel Suman Suggested-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_audmix.c | 27 +++++++++++++-------------- 1 file changed, 13 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index dc802d5c4ccd..3897a54a11fe 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -456,6 +456,7 @@ MODULE_DEVICE_TABLE(of, fsl_audmix_ids); static int fsl_audmix_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; struct fsl_audmix *priv; struct resource *res; const char *mdrv; @@ -463,52 +464,50 @@ static int fsl_audmix_probe(struct platform_device *pdev) void __iomem *regs; int ret; - of_id = of_match_device(fsl_audmix_ids, &pdev->dev); + of_id = of_match_device(fsl_audmix_ids, dev); if (!of_id || !of_id->data) return -EINVAL; mdrv = of_id->data; - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; /* Get the addresses */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - regs = devm_ioremap_resource(&pdev->dev, res); + regs = devm_ioremap_resource(dev, res); if (IS_ERR(regs)) return PTR_ERR(regs); - priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "ipg", regs, + priv->regmap = devm_regmap_init_mmio_clk(dev, "ipg", regs, &fsl_audmix_regmap_config); if (IS_ERR(priv->regmap)) { - dev_err(&pdev->dev, "failed to init regmap\n"); + dev_err(dev, "failed to init regmap\n"); return PTR_ERR(priv->regmap); } - priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg"); + priv->ipg_clk = devm_clk_get(dev, "ipg"); if (IS_ERR(priv->ipg_clk)) { - dev_err(&pdev->dev, "failed to get ipg clock\n"); + dev_err(dev, "failed to get ipg clock\n"); return PTR_ERR(priv->ipg_clk); } platform_set_drvdata(pdev, priv); - pm_runtime_enable(&pdev->dev); + pm_runtime_enable(dev); - ret = devm_snd_soc_register_component(&pdev->dev, &fsl_audmix_component, + ret = devm_snd_soc_register_component(dev, &fsl_audmix_component, fsl_audmix_dai, ARRAY_SIZE(fsl_audmix_dai)); if (ret) { - dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + dev_err(dev, "failed to register ASoC DAI\n"); return ret; } - priv->pdev = platform_device_register_data(&pdev->dev, mdrv, 0, NULL, - 0); + priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0); if (IS_ERR(priv->pdev)) { ret = PTR_ERR(priv->pdev); - dev_err(&pdev->dev, "failed to register platform %s: %d\n", - mdrv, ret); + dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret); } return ret; -- cgit v1.2.3-59-g8ed1b From 882c8b4af315bf921431142c1e9f5f69df99da34 Mon Sep 17 00:00:00 2001 From: Seppo Ingalsuo Date: Tue, 30 Apr 2019 18:11:34 -0500 Subject: ASoC: dapm: fix kcontrols for effect widgets This patch adds the handling of snd_soc_dapm_effect that was missing. Signed-off-by: Ranjani Sridharan Signed-off-by: Seppo Ingalsuo Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0382a47b30bd..29cdfbf4c888 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -883,6 +883,7 @@ static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w, case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: wname_in_long_name = true; kcname_in_long_name = true; @@ -2370,6 +2371,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, case snd_soc_dapm_dac: case snd_soc_dapm_adc: case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: @@ -3197,6 +3199,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) dapm_new_mux(w); break; case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: dapm_new_pga(w); break; -- cgit v1.2.3-59-g8ed1b From 414a7321d60bc0abced4a760e22b8187e2b4aecf Mon Sep 17 00:00:00 2001 From: Logesh Date: Wed, 1 May 2019 14:34:24 +0530 Subject: ASoC: da7213: fix DAI_CLK_EN register bit overwrite If the da7213 codec is configured as Master with the DAPM power down delay time set, 'snd_soc_component_write' function overwrites the DAI_CLK_EN bit of DAI_CLK_MODE register which leads to audio play only once until it re-initialize after codec power up. Signed-off-by: Logesh Reviewed-by: Adam Thomson Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 5 ++++- sound/soc/codecs/da7213.h | 2 ++ 2 files changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 92d006a5283e..425c11d63e49 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1305,7 +1305,10 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* By default only 64 BCLK per WCLK is supported */ dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_64; - snd_soc_component_write(component, DA7213_DAI_CLK_MODE, dai_clk_mode); + snd_soc_component_update_bits(component, DA7213_DAI_CLK_MODE, + DA7213_DAI_BCLKS_PER_WCLK_MASK | + DA7213_DAI_CLK_POL_MASK | DA7213_DAI_WCLK_POL_MASK, + dai_clk_mode); snd_soc_component_update_bits(component, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK, dai_ctrl); snd_soc_component_write(component, DA7213_DAI_OFFSET, dai_offset); diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 5a78dba1dcb5..9d31efc3cfe5 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -181,7 +181,9 @@ #define DA7213_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) #define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) #define DA7213_DAI_CLK_POL_INV (0x1 << 2) +#define DA7213_DAI_CLK_POL_MASK (0x1 << 2) #define DA7213_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7213_DAI_WCLK_POL_MASK (0x1 << 3) #define DA7213_DAI_CLK_EN_MASK (0x1 << 7) /* DA7213_DAI_CTRL = 0x29 */ -- cgit v1.2.3-59-g8ed1b From 7c88b92816dfe5eab224b96577b50ac00b4be68a Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Mon, 29 Apr 2019 12:25:12 +0000 Subject: ASoC: sprd: Fix return value check in sprd_mcdt_probe() In case of error, the function devm_ioremap_resource() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Fixes: d7bff893e04f ("ASoC: sprd: Add Spreadtrum multi-channel data transfer support") Signed-off-by: Wei Yongjun Reviewed-by: Baolin Wang Signed-off-by: Mark Brown --- sound/soc/sprd/sprd-mcdt.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 28f5e649733d..e9318d7a4810 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -951,8 +951,8 @@ static int sprd_mcdt_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); mcdt->base = devm_ioremap_resource(&pdev->dev, res); - if (!mcdt->base) - return -ENOMEM; + if (IS_ERR(mcdt->base)) + return PTR_ERR(mcdt->base); mcdt->dev = &pdev->dev; spin_lock_init(&mcdt->lock); -- cgit v1.2.3-59-g8ed1b From 81a812c98b6eda7e3101305d57354433e3edc541 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Mon, 29 Apr 2019 12:37:13 +0000 Subject: ASoC: sprd: Fix to use list_for_each_entry_safe() when delete items Since we will remove items off the list using list_del() we need to use a safe version of the list_for_each_entry() macro aptly named list_for_each_entry_safe(). Fixes: d7bff893e04f ("ASoC: sprd: Add Spreadtrum multi-channel data transfer support") Signed-off-by: Wei Yongjun Reviewed-by: Baolin Wang Signed-off-by: Mark Brown --- sound/soc/sprd/sprd-mcdt.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index e9318d7a4810..7448015a4935 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -978,12 +978,12 @@ static int sprd_mcdt_probe(struct platform_device *pdev) static int sprd_mcdt_remove(struct platform_device *pdev) { - struct sprd_mcdt_chan *temp; + struct sprd_mcdt_chan *chan, *temp; mutex_lock(&sprd_mcdt_list_mutex); - list_for_each_entry(temp, &sprd_mcdt_chan_list, list) - list_del(&temp->list); + list_for_each_entry_safe(chan, temp, &sprd_mcdt_chan_list, list) + list_del(&chan->list); mutex_unlock(&sprd_mcdt_list_mutex); -- cgit v1.2.3-59-g8ed1b From 81ed884501a59a49c71cd810578f793f3890c131 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 1 May 2019 11:13:32 +0100 Subject: ASoC: wm_adsp: Use DSP ops pointers to stop watchdog in error handlers Whilst this isn't strictly necessary as the code is already DSP specific better to use the pointers to avoid potential issues in the future if one core ends up having multiple methods of stopping the watchdog. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a9298bfddd9c..bb10c4aeceb4 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -4285,7 +4285,7 @@ irqreturn_t wm_adsp2_bus_error(struct wm_adsp *dsp) if (val & ADSP2_WDT_TIMEOUT_STS_MASK) { adsp_err(dsp, "watchdog timeout error\n"); - wm_adsp_stop_watchdog(dsp); + dsp->ops->stop_watchdog(dsp); wm_adsp_fatal_error(dsp); } @@ -4396,7 +4396,7 @@ irqreturn_t wm_halo_wdt_expire(int irq, void *data) mutex_lock(&dsp->pwr_lock); adsp_warn(dsp, "WDT Expiry Fault\n"); - wm_halo_stop_watchdog(dsp); + dsp->ops->stop_watchdog(dsp); wm_adsp_fatal_error(dsp); mutex_unlock(&dsp->pwr_lock); -- cgit v1.2.3-59-g8ed1b From 07f80454369e5a8141dbbf4ae0a535230f223f2b Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 1 May 2019 11:23:08 +0100 Subject: ASoC: SOF: Intel: fix spelling mistake "incompatble" -> "incompatible" There is a spelling mistake in a hda_dsp_rom_msg message, fix it. Signed-off-by: Colin Ian King Reviewed-by: Mukesh Ojha Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index b8fc19790f3b..84baf275b467 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -54,7 +54,7 @@ static const struct hda_dsp_msg_code hda_dsp_rom_msg[] = { {HDA_DSP_ROM_L2_CACHE_ERROR, "error: L2 cache error"}, {HDA_DSP_ROM_LOAD_OFFSET_TO_SMALL, "error: load offset too small"}, {HDA_DSP_ROM_API_PTR_INVALID, "error: API ptr invalid"}, - {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatble"}, + {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatible"}, {HDA_DSP_ROM_UNHANDLED_INTERRUPT, "error: unhandled interrupt"}, {HDA_DSP_ROM_MEMORY_HOLE_ECC, "error: ECC memory hole"}, {HDA_DSP_ROM_KERNEL_EXCEPTION, "error: kernel exception"}, -- cgit v1.2.3-59-g8ed1b From b75a9799dc371ce5f51a73536f248f60a1be74ec Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 1 May 2019 11:13:31 +0100 Subject: ASoC: wm_adsp: Correct region base typo in wm_halo_setup_algs Due to a typo the wrong base is being supplied for the primary algorithm on Halo firmwares, which will cause the controls to not function. Fixes: 170b1e123f38 ("ASoC: wm_adsp: Add support for new Halo core DSPs") Reported-by: Stuart Henderson Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bb10c4aeceb4..b26e6b825a90 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2445,7 +2445,7 @@ static int wm_halo_setup_algs(struct wm_adsp *dsp) wmfw_v3_parse_id_header(dsp, &halo_id.fw, n_algs); ret = wm_halo_create_regions(dsp, halo_id.fw.id, - halo_id.ym_base, halo_id.ym_base); + halo_id.xm_base, halo_id.ym_base); if (ret) return ret; -- cgit v1.2.3-59-g8ed1b From 903c220b1ece12f17c868e43f2243b8f81ff2d4c Mon Sep 17 00:00:00 2001 From: "S.j. Wang" Date: Sun, 28 Apr 2019 02:24:27 +0000 Subject: ASoC: fsl_esai: Fix missing break in switch statement case ESAI_HCKT_EXTAL and case ESAI_HCKR_EXTAL should be independent of each other, so replace fall-through with break. Fixes: 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Signed-off-by: Shengjiu Wang Acked-by: Nicolin Chen Cc: Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7410bbfd2af..bad0dfed6b68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -251,7 +251,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; - /* fall through */ + break; case ESAI_HCKR_EXTAL: ecr |= esai_priv->synchronous ? ESAI_ECR_ETI : ESAI_ECR_ERI; break; -- cgit v1.2.3-59-g8ed1b From 406dcbc55a0a20fd155be889a4a0c4b812f7c18e Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Mon, 29 Apr 2019 11:44:07 +0800 Subject: ASoC: rt5645: Headphone Jack sense inverts on the LattePanda board The LattePanda board has a sound card chtrt5645, when there is nothing plugged in the headphone jack, the system thinks the headphone is plugged in, while we plug a headphone in the jack, the system thinks the headphone is unplugged. If adding quirk=0x21 in the module parameter, the headphone jack can work well. So let us fix it via platform_data. https://bugs.launchpad.net/bugs/182459 Signed-off-by: Hui Wang Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index f842775dbf2c..cd45d41df4ec 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3634,6 +3634,11 @@ static const struct rt5645_platform_data jd_mode3_platform_data = { .jd_mode = 3, }; +static const struct rt5645_platform_data lattepanda_board_platform_data = { + .jd_mode = 2, + .inv_jd1_1 = true +}; + static const struct dmi_system_id dmi_platform_data[] = { { .ident = "Chrome Buddy", @@ -3731,6 +3736,15 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&intel_braswell_platform_data, }, + { + .ident = "LattePanda board", + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + DMI_EXACT_MATCH(DMI_BOARD_VERSION, "Default string"), + }, + .driver_data = (void *)&lattepanda_board_platform_data, + }, { } }; -- cgit v1.2.3-59-g8ed1b From 1cd472d2ac1654f939ae01164b29e81fc76e0a93 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Mon, 29 Apr 2019 11:57:33 +0100 Subject: ASoC: da7219: Use clk_round_rate to handle enabled bclk/wclk case For some platforms where DA7219 is the DAI clock master, BCLK/WCLK will be set and enabled prior to the codec's hw_params() function being called. It is possible the platform requires a different BCLK configuration than would be chosen by hw_params(), for example S16_LE format needed with a 64-bit frame to satisfy certain devices using the clocks. To handle those kinds of scenarios, the use of clk_round_rate() is now employed as part of hw_params(). If BCLK is already enabled then this function will just return the currently set rate, if it is valid for the desired frame size, so the subsequent call to clk_set_rate() will succeed and nothing changes with regards to clocking. In addition the specific BCLK & WCLK recalc_rate() implementations needed updating to always give back a real value, as those functions are called as part of the clk init code and a real value is needed for the clk_round_rate() call to work as expected. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- sound/soc/codecs/da7219.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 5f5fa3416af3..206d01c6eb7e 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1621,6 +1621,21 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, if (bclk) { bclk_rate = frame_size * sr; + /* + * Rounding the rate here avoids failure trying to set a + * new rate on an already enabled bclk. In that + * instance this will just set the same rate as is + * currently in use, and so should continue without + * problem, as long as the BCLK rate is suitable for the + * desired frame size. + */ + bclk_rate = clk_round_rate(bclk, bclk_rate); + if ((bclk_rate / sr) < frame_size) { + dev_err(component->dev, + "BCLK rate mismatch against frame size"); + return -EINVAL; + } + ret = clk_set_rate(bclk, bclk_rate); if (ret) { dev_err(component->dev, @@ -1927,9 +1942,6 @@ static unsigned long da7219_wclk_recalc_rate(struct clk_hw *hw, struct snd_soc_component *component = da7219->component; u8 fs = snd_soc_component_read32(component, DA7219_SR); - if (!da7219->master) - return 0; - switch (fs & DA7219_SR_MASK) { case DA7219_SR_8000: return 8000; @@ -2016,9 +2028,6 @@ static unsigned long da7219_bclk_recalc_rate(struct clk_hw *hw, u8 bclks_per_wclk = snd_soc_component_read32(component, DA7219_DAI_CLK_MODE); - if (!da7219->master) - return 0; - switch (bclks_per_wclk & DA7219_DAI_BCLKS_PER_WCLK_MASK) { case DA7219_DAI_BCLKS_PER_WCLK_32: return parent_rate * 32; -- cgit v1.2.3-59-g8ed1b From cd8926e38e8cc53413a2a4ed2f705db7437a55fb Mon Sep 17 00:00:00 2001 From: Yu-Hsuan Hsu Date: Thu, 2 May 2019 12:07:43 +0800 Subject: ASoC: da7219: Update the support rate list If we want to set rate to 64000 on da7219, it fails and returns "snd_pcm_hw_params: Invalid argument". We should remove 64000 from support rate list because it is not available. Signed-off-by: Yu-Hsuan Hsu Reviewed-by: Adam Thomson Signed-off-by: Mark Brown --- sound/soc/codecs/da7219.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 121a8190f93e..9f6970eed6f6 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1583,20 +1583,26 @@ static const struct snd_soc_dai_ops da7219_dai_ops = { #define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +#define DA7219_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + static struct snd_soc_dai_driver da7219_dai = { .name = "da7219-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = DA7219_DAI_CH_NUM_MAX, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = DA7219_RATES, .formats = DA7219_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = DA7219_DAI_CH_NUM_MAX, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = DA7219_RATES, .formats = DA7219_FORMATS, }, .ops = &da7219_dai_ops, -- cgit v1.2.3-59-g8ed1b From 30180e8436046344b12813dc954b2e01dfdcd22d Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 29 Apr 2019 15:29:39 +0200 Subject: ASoC: hdmi-codec: unlock the device on startup errors If the hdmi codec startup fails, it should clear the current_substream pointer to free the device. This is properly done for the audio_startup() callback but for snd_pcm_hw_constraint_eld(). Make sure the pointer cleared if an error is reported. Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 35df73e42cbc..fb2f0ac1f16f 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -439,8 +439,12 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, if (!ret) { ret = snd_pcm_hw_constraint_eld(substream->runtime, hcp->eld); - if (ret) + if (ret) { + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); return ret; + } } /* Select chmap supported */ hdmi_codec_eld_chmap(hcp); -- cgit v1.2.3-59-g8ed1b From 6a7c59c6d9f3b280e81d7a04bbe4e55e90152dce Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 29 Apr 2019 11:47:49 +0200 Subject: ASoC: fix valid stream condition A stream may specify a rate range using 'rate_min' and 'rate_max', so a stream may be valid and not specify any rates. However, as stream cannot be valid and not have any channel. Let's use this condition instead to determine if a stream is valid or not. Fixes: cde79035c6cf ("ASoC: Handle multiple codecs with split playback / capture") Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d21247546f7f..57088bd69e5d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -43,8 +43,8 @@ static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) else codec_stream = &dai->driver->capture; - /* If the codec specifies any rate at all, it supports the stream. */ - return codec_stream->rates; + /* If the codec specifies any channels at all, it supports the stream */ + return codec_stream->channels_min; } /** -- cgit v1.2.3-59-g8ed1b From f47b9ad927c6370b80922af434dda98764a43804 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 29 Apr 2019 11:47:50 +0200 Subject: ASoC: skip hw_free on codec dai for which the stream is invalid Like for hw_params, hw_free should not be called on codec dai for which the current stream is invalid. Fixes: cde79035c6cf ("ASoC: Handle multiple codecs with split playback / capture") Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 57088bd69e5d..a810f6eeffee 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1031,6 +1031,9 @@ interface_err: codec_err: for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) { + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); codec_dai->rate = 0; @@ -1088,6 +1091,9 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* now free hw params for the DAIs */ for_each_rtd_codec_dai(rtd, i, codec_dai) { + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); } -- cgit v1.2.3-59-g8ed1b From c437ba0300cec1ca8758fdd71a3b2f8cd02c6e2a Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 2 May 2019 12:33:40 +0100 Subject: ASoC: SOF: remove redundant null checks of dai Currently there are two null checks of pointer dai in function sof_connect_dai_widget and yet there is no null check of dai at the end of the function when checking !dai->name. The latter would be a null pointer deference if dai is null (as picked up by static analysis), however the function is only ever called when dai is successfully allocated, so the null checks are redundant. Clean up the code by removing the null checks. Addresses-Coverity: ("Dereference after null check") Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 2b9de1b97447..1f71857298a9 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1127,15 +1127,13 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, switch (w->id) { case snd_soc_dapm_dai_out: rtd->cpu_dai->capture_widget = w; - if (dai) - dai->name = rtd->dai_link->name; + dai->name = rtd->dai_link->name; dev_dbg(sdev->dev, "tplg: connected widget %s -> DAI link %s\n", w->name, rtd->dai_link->name); break; case snd_soc_dapm_dai_in: rtd->cpu_dai->playback_widget = w; - if (dai) - dai->name = rtd->dai_link->name; + dai->name = rtd->dai_link->name; dev_dbg(sdev->dev, "tplg: connected widget %s -> DAI link %s\n", w->name, rtd->dai_link->name); break; -- cgit v1.2.3-59-g8ed1b From 726fc60babe4a46e946e69a9dbd3e21aaec4d58e Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Mon, 29 Apr 2019 15:29:40 +0200 Subject: ASoC: hdmi-codec: stream is already locked in hw_params startup() should have run before hw_params() is called, so the current_substream pointer should already be properly set. There is no reason to call hdmi_codec_new_stream() again in the hw_params() callback Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b9d9dde9fbaf..ef6d6959ecc5 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -492,10 +492,6 @@ static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = hdmi_codec_new_stream(substream, dai); - if (ret) - return ret; - hdmi_audio_infoframe_init(&hp.cea); hp.cea.channels = params_channels(params); hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; -- cgit v1.2.3-59-g8ed1b From 0efa3334d65b7f421ba12382dfa58f6ff5bf83c4 Mon Sep 17 00:00:00 2001 From: Ross Zwisler Date: Mon, 29 Apr 2019 12:25:17 -0600 Subject: ASoC: Intel: avoid Oops if DMA setup fails Currently in sst_dsp_new() if we get an error return from sst_dma_new() we just print an error message and then still complete the function successfully. This means that we are trying to run without sst->dma properly set up, which will result in NULL pointer dereference when sst->dma is later used. This was happening for me in sst_dsp_dma_get_channel(): struct sst_dma *dma = dsp->dma; ... dma->ch = dma_request_channel(mask, dma_chan_filter, dsp); This resulted in: BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 IP: sst_dsp_dma_get_channel+0x4f/0x125 [snd_soc_sst_firmware] Fix this by adding proper error handling for the case where we fail to set up DMA. This change only affects Haswell and Broadwell systems. Baytrail systems explicilty opt-out of DMA via sst->pdata->resindex_dma_base being set to -1. Signed-off-by: Ross Zwisler Cc: stable@vger.kernel.org Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/sst-firmware.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 1e067504b604..f830e59f93ea 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -1251,11 +1251,15 @@ struct sst_dsp *sst_dsp_new(struct device *dev, goto irq_err; err = sst_dma_new(sst); - if (err) - dev_warn(dev, "sst_dma_new failed %d\n", err); + if (err) { + dev_err(dev, "sst_dma_new failed %d\n", err); + goto dma_err; + } return sst; +dma_err: + free_irq(sst->irq, sst); irq_err: if (sst->ops->free) sst->ops->free(sst); -- cgit v1.2.3-59-g8ed1b From 73affae222ec0ffe44d870e0085e55cdbe85d30d Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 30 Apr 2019 18:09:16 -0500 Subject: ASoC: SOF: add Kconfig option for strict ABI checks When the kernel is more recent than firmware files, it will always behave in backwards-compatible ways. Add optional behavior to check if the kernel is older than the firmware files, so that the kernel fails early instead of attempting to use new functionality it does not support. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/Kconfig | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index eaa380092c3b..a1a9ffe605dc 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -53,6 +53,21 @@ config SND_SOC_SOF_NOCODEC Say Y if you need this nocodec fallback option If unsure select "N". +config SND_SOC_SOF_STRICT_ABI_CHECKS + bool "SOF strict ABI checks" + help + This option enables strict ABI checks for firmware and topology + files. + When these files are more recent than the kernel, the kernel + will handle the functionality it supports and may report errors + during topology creation or run-time usage if new functionality + is invoked. + This option will stop topology creation and firmware load upfront. + It is intended for SOF CI/releases and not for users or distros. + Say Y if you want strict ABI checks for an SOF release + If you are not involved in SOF releases and CI development + select "N". + config SND_SOC_SOF_DEBUG bool "SOF debugging features" help -- cgit v1.2.3-59-g8ed1b From 4acb1c2ea42e19101eae4ac9f002853553b8b721 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 30 Apr 2019 18:09:17 -0500 Subject: ASOC: SOF: ipc: add support for stricter ABI checks Fail early if firmware is more recent than kernel and Kconfig is selected. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/ipc.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index d00373ceca12..bf58d58e7379 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -762,6 +762,15 @@ int snd_sof_ipc_valid(struct snd_sof_dev *sdev) return -EINVAL; } + if (v->abi_version > SOF_ABI_VERSION) { + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS)) { + dev_warn(sdev->dev, "warn: FW ABI is more recent than kernel\n"); + } else { + dev_err(sdev->dev, "error: FW ABI is more recent than kernel\n"); + return -EINVAL; + } + } + if (ready->debug.bits.build) { dev_info(sdev->dev, "Firmware debug build %d on %s-%s - options:\n" -- cgit v1.2.3-59-g8ed1b From 8e3a6e45a77bd4167554b4bd0633a2adabf1bd77 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 30 Apr 2019 18:09:18 -0500 Subject: ASoC: SOF: topology: add support for stricter ABI checks Fail early if topology is more recent than kernel and Kconfig is selected. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 43 ++++++++++++++++++++++++++++++------------- 1 file changed, 30 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 1f71857298a9..c88afa872a58 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -3049,6 +3049,7 @@ static int sof_manifest(struct snd_soc_component *scomp, int index, { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); u32 size; + u32 abi_version; size = le32_to_cpu(man->priv.size); @@ -3058,20 +3059,36 @@ static int sof_manifest(struct snd_soc_component *scomp, int index, return 0; } - if (size == SOF_TPLG_ABI_SIZE) { - dev_info(sdev->dev, - "Topology: ABI %d:%d:%d Kernel ABI %d:%d:%d\n", - man->priv.data[0], man->priv.data[1], - man->priv.data[2], SOF_ABI_MAJOR, SOF_ABI_MINOR, - SOF_ABI_PATCH); - if (SOF_ABI_VER(man->priv.data[0], man->priv.data[1], - man->priv.data[2]) <= SOF_ABI_VERSION) - return 0; + if (size != SOF_TPLG_ABI_SIZE) { + dev_err(sdev->dev, "error: invalid topology ABI size\n"); + return -EINVAL; } - dev_err(sdev->dev, - "error: Incompatible ABI version %d:%d:%d\n", - man->priv.data[0], man->priv.data[1], man->priv.data[2]); - return -EINVAL; + + dev_info(sdev->dev, + "Topology: ABI %d:%d:%d Kernel ABI %d:%d:%d\n", + man->priv.data[0], man->priv.data[1], + man->priv.data[2], SOF_ABI_MAJOR, SOF_ABI_MINOR, + SOF_ABI_PATCH); + + abi_version = SOF_ABI_VER(man->priv.data[0], + man->priv.data[1], + man->priv.data[2]); + + if (SOF_ABI_VERSION_INCOMPATIBLE(SOF_ABI_VERSION, abi_version)) { + dev_err(sdev->dev, "error: incompatible topology ABI version\n"); + return -EINVAL; + } + + if (abi_version > SOF_ABI_VERSION) { + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_STRICT_ABI_CHECKS)) { + dev_warn(sdev->dev, "warn: topology ABI is more recent than kernel\n"); + } else { + dev_err(sdev->dev, "error: topology ABI is more recent than kernel\n"); + return -EINVAL; + } + } + + return 0; } /* vendor specific kcontrol handlers available for binding */ -- cgit v1.2.3-59-g8ed1b From acd1c1883d40d1b03f3b70a82afcab9aa3b94c98 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Tue, 30 Apr 2019 18:09:19 -0500 Subject: ASoC: SOF: Intel: cnl: add pointer ops to use DPIB position Add .pcm_pointer ops for cannonlake to read DPIB/posbuf and get pointer for ALSA, to align with apollolake. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/cnl.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 3e95c1e5e491..36ae9b88d0dc 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -199,6 +199,7 @@ const struct snd_sof_dsp_ops sof_cnl_ops = { .pcm_close = hda_dsp_pcm_close, .pcm_hw_params = hda_dsp_pcm_hw_params, .pcm_trigger = hda_dsp_pcm_trigger, + .pcm_pointer = hda_dsp_pcm_pointer, /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, -- cgit v1.2.3-59-g8ed1b From df7e0de588252ccb11c2caf320df1dce23219917 Mon Sep 17 00:00:00 2001 From: Zhu Yingjiang Date: Tue, 30 Apr 2019 18:09:20 -0500 Subject: ASoC: SOF: Intel: hda: add the SSP Host Device memory space The DSP SSP device memory can be conditionally accessed by the host(depending on access policy). Add the SSP base memory offset of APL and CNL. Signed-off-by: Zhu Yingjiang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 4efcf85af6b5..c4550f2e493d 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -338,6 +338,13 @@ #define HDA_ADSP_FW_STATUS_SKL HDA_ADSP_SRAM0_BASE_SKL #define HDA_ADSP_ERROR_CODE_SKL (HDA_ADSP_FW_STATUS_SKL + 0x4) +/* Host Device Memory Space */ +#define APL_SSP_BASE_OFFSET 0x2000 +#define CNL_SSP_BASE_OFFSET 0x10000 + +/* Host Device Memory Size of a Single SSP */ +#define SSP_DEV_MEM_SIZE 0x1000 + #define HDA_IDISP_CODEC(x) ((x) & BIT(2)) struct sof_intel_dsp_bdl { -- cgit v1.2.3-59-g8ed1b From b095fe47bc1e1c3bf114fde7b76a387ee7b16934 Mon Sep 17 00:00:00 2001 From: Zhu Yingjiang Date: Tue, 30 Apr 2019 18:09:21 -0500 Subject: ASoC: SOF: Intel: hda: add SSP info to the chip info struct add SSP info of APL and CNL, to the sof_intel_dsp_desc structure. The max SSP count the platform support and the SSP base address. Signed-off-by: Zhu Yingjiang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/apl.c | 2 ++ sound/soc/sof/intel/cnl.c | 2 ++ sound/soc/sof/intel/hda.h | 4 ++++ sound/soc/sof/intel/shim.h | 2 ++ 4 files changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index 8c6282606944..026dde839621 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -105,5 +105,7 @@ const struct sof_intel_dsp_desc apl_chip_info = { .ipc_ack_mask = HDA_DSP_REG_HIPCIE_DONE, .ipc_ctl = HDA_DSP_REG_HIPCCTL, .rom_init_timeout = 150, + .ssp_count = APL_SSP_COUNT, + .ssp_base_offset = APL_SSP_BASE_OFFSET, }; EXPORT_SYMBOL(apl_chip_info); diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 36ae9b88d0dc..3afd96d9c925 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -246,5 +246,7 @@ const struct sof_intel_dsp_desc cnl_chip_info = { .ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE, .ipc_ctl = CNL_DSP_REG_HIPCCTL, .rom_init_timeout = 300, + .ssp_count = CNL_SSP_COUNT, + .ssp_base_offset = CNL_SSP_BASE_OFFSET, }; EXPORT_SYMBOL(cnl_chip_info); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index c4550f2e493d..b2cf6fa15575 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -345,6 +345,10 @@ /* Host Device Memory Size of a Single SSP */ #define SSP_DEV_MEM_SIZE 0x1000 +/* SSP Count of the Platform */ +#define APL_SSP_COUNT 6 +#define CNL_SSP_COUNT 3 + #define HDA_IDISP_CODEC(x) ((x) & BIT(2)) struct sof_intel_dsp_bdl { diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index 11fd77cb4e6d..f7a3f62e45d4 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -162,6 +162,8 @@ struct sof_intel_dsp_desc { int ipc_ack_mask; int ipc_ctl; int rom_init_timeout; + int ssp_count; /* ssp count of the platform */ + int ssp_base_offset; /* base address of the SSPs */ }; extern const struct snd_sof_dsp_ops sof_tng_ops; -- cgit v1.2.3-59-g8ed1b From 74ed4097f5894a07b515361dd673ad3fadb665b2 Mon Sep 17 00:00:00 2001 From: Zhu Yingjiang Date: Tue, 30 Apr 2019 18:09:22 -0500 Subject: ASoC: SOF: Intel: hda: set I2S slave before enabling DSP By default, the I2S ports are configured in master mode during DSP powerup sequences, the FS and BCLK lines will be driven on startup, even when the topology file explicitly requires the SSP to be slave. This may be problematic for external components configured in master mode who don't expect the Intel SOC/PCH to drive. Fix by configuring the SSP as slave before the SSP outputs are enabled to avoid this transient behavior. When the topology file configures the SSP as clock master, the initial slave configuration will be overridden. Signed-off-by: Zhu Yingjiang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 11 +++++++++++ sound/soc/sof/intel/hda.h | 6 ++++++ 2 files changed, 17 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 6a44bc349e44..6427f0b3a2f1 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -84,6 +84,7 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, const void *fwdata, const struct sof_intel_dsp_desc *chip = hda->desc; unsigned int status; int ret; + int i; /* step 1: power up corex */ ret = hda_dsp_core_power_up(sdev, chip->cores_mask); @@ -92,6 +93,16 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, const void *fwdata, goto err; } + /* DSP is powered up, set all SSPs to slave mode */ + for (i = 0; i < chip->ssp_count; i++) { + snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, + chip->ssp_base_offset + + i * SSP_DEV_MEM_SIZE + + SSP_SSC1_OFFSET, + SSP_SET_SLAVE, + SSP_SET_SLAVE); + } + /* step 2: purge FW request */ snd_sof_dsp_write(sdev, HDA_DSP_BAR, chip->ipc_req, chip->ipc_req_mask | (HDA_DSP_IPC_PURGE_FW | diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index b2cf6fa15575..3799a3518106 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -349,6 +349,12 @@ #define APL_SSP_COUNT 6 #define CNL_SSP_COUNT 3 +/* SSP Registers */ +#define SSP_SSC1_OFFSET 0x4 +#define SSP_SET_SCLK_SLAVE BIT(25) +#define SSP_SET_SFRM_SLAVE BIT(24) +#define SSP_SET_SLAVE (SSP_SET_SCLK_SLAVE | SSP_SET_SFRM_SLAVE) + #define HDA_IDISP_CODEC(x) ((x) & BIT(2)) struct sof_intel_dsp_bdl { -- cgit v1.2.3-59-g8ed1b From d1a6459fab4357875ccffa44636c13d8a1291146 Mon Sep 17 00:00:00 2001 From: Zhu Yingjiang Date: Tue, 30 Apr 2019 18:09:23 -0500 Subject: ASoC: SOF: Intel: hda: set bus->idx as 0 Setting the bus->idx as 0, for we only have one HDA bus atm. This need to be fixed when there are more than one HDA bus. Signed-off-by: Zhu Yingjiang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-bus.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index 62cc9921bb55..a7e6d8227df6 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -80,8 +80,6 @@ static const struct hdac_io_ops io_ops = { void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev, const struct hdac_ext_bus_ops *ext_ops) { - static int idx; - memset(bus, 0, sizeof(*bus)); bus->dev = dev; @@ -90,7 +88,12 @@ void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev, bus->irq = -1; bus->ext_ops = ext_ops; - bus->idx = idx++; + + /* + * There is only one HDA bus atm. keep the index as 0. + * Need to fix when there are more than one HDA bus. + */ + bus->idx = 0; spin_lock_init(&bus->reg_lock); -- cgit v1.2.3-59-g8ed1b From b0056fda7c8a474bef9bf01368f66caadcdd464c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 30 Apr 2019 18:09:24 -0500 Subject: ASoC: SOF: Intel: hda-pcm: remove useless dependency on hdac_ext Nothing depends on definitions in hdaudio_ext.h, don't include it Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-pcm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 5714a79fbe1a..9b730f183529 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -15,7 +15,6 @@ * Hardware interface for generic Intel audio DSP HDA IP */ -#include #include #include #include "../ops.h" -- cgit v1.2.3-59-g8ed1b From e2803e610aecb36ea4fec5a04861547664580d0c Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Tue, 30 Apr 2019 18:09:25 -0500 Subject: ASoC: SOF: PCM: add period_elapsed work to fix race condition in interrupt context The IPC implementation in SOF requires sending IPCs serially: we should not send a new IPC command to the firmware before we get an ACK (or time out) from firmware, and the IRQ processing is complete. snd_pcm_period_elapsed() can be called in interrupt context before IRQ_HANDLED is returned. When the PCM is done draining, a STOP IPC will then be sent, which breaks the expectation that IPCs are handled serially and leads to IPC timeouts. This patch adds a workqueue to defer the call to snd_pcm_elapsed() after the IRQ is handled. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 2 ++ 2 files changed, 50 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index be4984c4da4e..649968841dad 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -52,6 +52,48 @@ static int sof_pcm_dsp_params(struct snd_sof_pcm *spcm, struct snd_pcm_substream return ret; } +/* + * sof pcm period elapse work + */ +static void sof_pcm_period_elapsed_work(struct work_struct *work) +{ + struct snd_sof_pcm_stream *sps = + container_of(work, struct snd_sof_pcm_stream, + period_elapsed_work); + + snd_pcm_period_elapsed(sps->substream); +} + +/* + * sof pcm period elapse, this could be called at irq thread context. + */ +void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(sdev, rtd); + if (!spcm) { + dev_err(sdev->dev, + "error: period elapsed for unknown stream!\n"); + return; + } + + /* + * snd_pcm_period_elapsed() can be called in interrupt context + * before IRQ_HANDLED is returned. Inside snd_pcm_period_elapsed(), + * when the PCM is done draining or xrun happened, a STOP IPC will + * then be sent and this IPC will hit IPC timeout. + * To avoid sending IPC before the previous IPC is handled, we + * schedule delayed work here to call the snd_pcm_period_elapsed(). + */ + schedule_work(&spcm->stream[substream->stream].period_elapsed_work); +} +EXPORT_SYMBOL(snd_sof_pcm_period_elapsed); + /* this may get called several times by oss emulation */ static int sof_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -169,6 +211,9 @@ static int sof_pcm_hw_params(struct snd_pcm_substream *substream, /* save pcm hw_params */ memcpy(&spcm->params[substream->stream], params, sizeof(*params)); + INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work, + sof_pcm_period_elapsed_work); + return ret; } @@ -203,6 +248,9 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream) sizeof(stream), &reply, sizeof(reply)); snd_pcm_lib_free_pages(substream); + + cancel_work_sync(&spcm->stream[substream->stream].period_elapsed_work); + return ret; } diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 35e78ffecce2..675bb10c82f5 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -274,6 +274,7 @@ struct snd_sof_pcm_stream { struct snd_dma_buffer page_table; struct sof_ipc_stream_posn posn; struct snd_pcm_substream *substream; + struct work_struct period_elapsed_work; }; /* ALSA SOF PCM device */ @@ -495,6 +496,7 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_sof_dev *sdev, int *direction); struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_sof_dev *sdev, unsigned int pcm_id); +void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream); /* * Stream IPC -- cgit v1.2.3-59-g8ed1b From 8242d53975a95c8c35c5fd0d2c1689b53be07db8 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Tue, 30 Apr 2019 18:09:26 -0500 Subject: ASoC: SOF: Intel: use snd_sof_pcm_period_elapsed Switch to a wrapper function which schedules the actual call of snd_pcm_period_elapsed after the current IPC is completed. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 6290b2df5e62..47eff161c60e 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -465,8 +465,8 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) { struct hdac_bus *bus = context; struct sof_intel_hda_dev *sof_hda = bus_to_sof_hda(bus); - struct hdac_stream *s; u32 status = snd_hdac_chip_readl(bus, INTSTS); + struct hdac_stream *s; u32 sd_status; /* check streams */ @@ -486,7 +486,7 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) /* Inform ALSA only in case not do that with IPC */ if (sof_hda->no_ipc_position) - snd_pcm_period_elapsed(s->substream); + snd_sof_pcm_period_elapsed(s->substream); } } -- cgit v1.2.3-59-g8ed1b From ff758e9f73a0fcb93a53349e8fbace4ac019d031 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Tue, 30 Apr 2019 18:09:27 -0500 Subject: ASoC: SOF: ipc: use snd_sof_pcm_period_elapsed Switch to a wrapper function which schedules the actual call of snd_pcm_period_elapsed after the current IPC is completed. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index bf58d58e7379..437e80a07bc2 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -445,7 +445,7 @@ static void ipc_period_elapsed(struct snd_sof_dev *sdev, u32 msg_id) /* only inform ALSA for period_wakeup mode */ if (!stream->substream->runtime->no_period_wakeup) - snd_pcm_period_elapsed(stream->substream); + snd_sof_pcm_period_elapsed(stream->substream); } /* DSP notifies host of an XRUN within FW */ -- cgit v1.2.3-59-g8ed1b From 6d60a39e5b787ddea5def220d6288cbdf6d63eb8 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Tue, 30 Apr 2019 18:09:28 -0500 Subject: ASoC: SOF: Intel: hda-ipc: simplify handling of IPC IRQ When using a shared IRQ between IPC interrupt and stream IOC interrupt, the interrupt handlers need to check the interrupt source before scheduling their respective IRQ threads. In the case of IPC handler, it should check if it is an IPC interrupt before waking up the IPC IRQ thread. The IPC IRQ thread, once scheduled, does not need to check the IRQ source again. So, remove the superfluous check in the thread. Remove the irq_status field from snd_sof_dev struct also as it is no longer needed. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/cnl.c | 4 ---- sound/soc/sof/intel/hda-ipc.c | 13 +++++-------- sound/soc/sof/sof-priv.h | 1 - 3 files changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 3afd96d9c925..d128839b2450 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -39,10 +39,6 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context) u32 msg_ext; irqreturn_t ret = IRQ_NONE; - /* here we handle IPC interrupts only */ - if (!(sdev->irq_status & HDA_DSP_ADSPIS_IPC)) - return ret; - hipcida = snd_sof_dsp_read(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDA); hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCCTL); hipctdr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCTDR); diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index a938f568dbb1..73ead7070cde 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -145,10 +145,6 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) u32 msg; u32 msg_ext; - /* here we handle IPC interrupts only */ - if (!(sdev->irq_status & HDA_DSP_ADSPIS_IPC)) - return IRQ_NONE; - /* read IPC status */ hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCIE); @@ -234,19 +230,20 @@ irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context) { struct snd_sof_dev *sdev = context; int ret = IRQ_NONE; + u32 irq_status; spin_lock(&sdev->hw_lock); /* store status */ - sdev->irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, - HDA_DSP_REG_ADSPIS); + irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS); + dev_vdbg(sdev->dev, "irq handler: irq_status:0x%x\n", irq_status); /* invalid message ? */ - if (sdev->irq_status == 0xffffffff) + if (irq_status == 0xffffffff) goto out; /* IPC message ? */ - if (sdev->irq_status & HDA_DSP_ADSPIS_IPC) { + if (irq_status & HDA_DSP_ADSPIS_IPC) { /* disable IPC interrupt */ snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 675bb10c82f5..bbc285018f9a 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -366,7 +366,6 @@ struct snd_sof_dev { struct snd_sof_mailbox host_box; /* Host initiated IPC */ struct snd_sof_mailbox stream_box; /* Stream position update */ struct snd_sof_ipc_msg *msg; - u64 irq_status; int ipc_irq; u32 next_comp_id; /* monotonic - reset during S3 */ -- cgit v1.2.3-59-g8ed1b From e8e55dbe08960b714e5ce0148926ee17331ce4d5 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Tue, 30 Apr 2019 18:09:29 -0500 Subject: ASoC: SOF: Intel: hda-stream: store stream capabilities Add stream_max into struct sof_intel_hda_dev to store the total hda stream number that the platform can support, and initialize it at stream_init. This can be used later e.g. for stream bitmask. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 4 ++++ sound/soc/sof/intel/hda.h | 3 +++ 2 files changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 47eff161c60e..39f4212bde4b 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -500,6 +500,7 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) struct hdac_ext_stream *stream; struct hdac_stream *hstream; struct pci_dev *pci = to_pci_dev(sdev->dev); + struct sof_intel_hda_dev *sof_hda = bus_to_sof_hda(bus); int sd_offset; int i, num_playback, num_capture, num_total, ret; u32 gcap; @@ -657,6 +658,9 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) list_add_tail(&hstream->list, &bus->stream_list); } + /* store total stream count (playback + capture) from GCAP */ + sof_hda->stream_max = num_total; + return 0; } diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 3799a3518106..7cb2fb92c4eb 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -383,6 +383,9 @@ struct sof_intel_hda_dev { /* if position update IPC needed */ u32 no_ipc_position; + /* the maximum number of streams (playback + capture) supported */ + u32 stream_max; + int irq; /* DMIC device */ -- cgit v1.2.3-59-g8ed1b From 20d0aff739bed801bbe67ebed8012f3d9849b29d Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Tue, 30 Apr 2019 18:09:30 -0500 Subject: ASoC: SOF: Intel: hda-stream: handle real stream interrupts only The stream and IPC share the same interrupt. The stream interrupt handler mistakenly uses the ipc interrupt and return IRQ_HANDLED, causing the ipc interrupt to be missed. Make sure the stream interrupt handler only deals with stream-related interrupts. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 39f4212bde4b..c92006f89499 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -433,6 +433,8 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, irqreturn_t hda_dsp_stream_interrupt(int irq, void *context) { struct hdac_bus *bus = context; + struct sof_intel_hda_dev *sof_hda = bus_to_sof_hda(bus); + u32 stream_mask; u32 status; if (!pm_runtime_active(bus->dev)) @@ -441,7 +443,10 @@ irqreturn_t hda_dsp_stream_interrupt(int irq, void *context) spin_lock(&bus->reg_lock); status = snd_hdac_chip_readl(bus, INTSTS); - if (status == 0 || status == 0xffffffff) { + stream_mask = GENMASK(sof_hda->stream_max - 1, 0) | AZX_INT_CTRL_EN; + + /* Not stream interrupt or register inaccessible, ignore it.*/ + if (!(status & stream_mask) || status == 0xffffffff) { spin_unlock(&bus->reg_lock); return IRQ_NONE; } -- cgit v1.2.3-59-g8ed1b From ed3baacd76baa045fe101ce04a9e46d48c188495 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 30 Apr 2019 18:09:31 -0500 Subject: ASoC: SOF: intel: hda: add hw_params_upon_resume flag for hda stream The prepare() ioctl for BE dai link gets called both when the stream is started and when it is resumed from suspend. SOF uses this ioctl to set the hw params again only if the stream has been suspended. When the stream is started, the hw_params ioctl gets called before prepare() and hw_params is set for the BE dai link. So the prepare call does not need to do anything further. When the stream resumes after system suspend, SOF requires that the hw_params be set again for the BE dai. In order to determine which streams should set the hw params during prepare(), an internal flag called "hw_params_upon_resume" is introduced in struct sof_intel_hda_stream. The flag is set for hda streams when the sof device suspends and is cleared after hw_params is set. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 1 + sound/soc/sof/intel/hda-dai.c | 23 ++++++++++++++--------- sound/soc/sof/intel/hda-dsp.c | 16 ++++++++++++++++ sound/soc/sof/intel/hda.h | 2 ++ sound/soc/sof/ops.h | 6 ++++++ sound/soc/sof/pm.c | 3 +++ sound/soc/sof/sof-priv.h | 1 + 8 files changed, 44 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index 026dde839621..d1fb2f2ba1e1 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -91,6 +91,7 @@ const struct snd_sof_dsp_ops sof_apl_ops = { .resume = hda_dsp_resume, .runtime_suspend = hda_dsp_runtime_suspend, .runtime_resume = hda_dsp_runtime_resume, + .set_hw_params_upon_resume = hda_dsp_set_hw_params_upon_resume, }; EXPORT_SYMBOL(sof_apl_ops); diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index d128839b2450..2eac20bbcaea 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -225,6 +225,7 @@ const struct snd_sof_dsp_ops sof_cnl_ops = { .resume = hda_dsp_resume, .runtime_suspend = hda_dsp_runtime_suspend, .runtime_resume = hda_dsp_runtime_resume, + .set_hw_params_upon_resume = hda_dsp_set_hw_params_upon_resume, }; EXPORT_SYMBOL(sof_cnl_ops); diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 830328af19c5..e1decf25aeac 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -131,12 +131,17 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, struct hdac_ext_stream *link_dev; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct sof_intel_hda_stream *hda_stream; struct hda_pipe_params p_params = {0}; struct hdac_ext_link *link; int stream_tag; link_dev = snd_soc_dai_get_dma_data(dai, substream); + hda_stream = container_of(link_dev, struct sof_intel_hda_stream, + hda_stream); + hda_stream->hw_params_upon_resume = 0; + link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); if (!link) return -EINVAL; @@ -168,22 +173,22 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, static int hda_link_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hdac_ext_stream *link_dev = + snd_soc_dai_get_dma_data(dai, substream); + struct sof_intel_hda_stream *hda_stream; struct snd_sof_dev *sdev = - snd_soc_component_get_drvdata(dai->component); - struct snd_sof_pcm *spcm; + snd_soc_component_get_drvdata(dai->component); + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); int stream = substream->stream; - spcm = snd_sof_find_spcm_dai(sdev, rtd); - if (!spcm) - return -EINVAL; + hda_stream = container_of(link_dev, struct sof_intel_hda_stream, + hda_stream); /* setup hw_params again only if resuming from system suspend */ - if (!spcm->hw_params_upon_resume[stream]) + if (!hda_stream->hw_params_upon_resume) return 0; - dev_dbg(sdev->dev, "hda: prepare stream %d dir %d\n", - spcm->pcm.pcm_id, substream->stream); + dev_dbg(sdev->dev, "hda: prepare stream dir %d\n", substream->stream); return hda_link_hw_params(substream, &rtd->dpcm[stream].hw_params, dai); diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 311fed502e09..5b73115a0b78 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -453,3 +453,19 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, int state) return 0; } + +void hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct sof_intel_hda_stream *hda_stream; + struct hdac_ext_stream *stream; + struct hdac_stream *s; + + /* set internal flag for BE */ + list_for_each_entry(s, &bus->stream_list, list) { + stream = stream_to_hdac_ext_stream(s); + hda_stream = container_of(stream, struct sof_intel_hda_stream, + hda_stream); + hda_stream->hw_params_upon_resume = 1; + } +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 7cb2fb92c4eb..4b00e5d88565 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -409,6 +409,7 @@ static inline struct hda_bus *sof_to_hbus(struct snd_sof_dev *s) struct sof_intel_hda_stream { struct hdac_ext_stream hda_stream; struct sof_intel_stream stream; + int hw_params_upon_resume; /* set up hw_params upon resume */ }; #define bus_to_sof_hda(bus) \ @@ -443,6 +444,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, int state); int hda_dsp_resume(struct snd_sof_dev *sdev); int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev, int state); int hda_dsp_runtime_resume(struct snd_sof_dev *sdev); +void hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev); void hda_dsp_dump_skl(struct snd_sof_dev *sdev, u32 flags); void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags); diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 2a5d4c63f160..b8e2bf1fee24 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -134,6 +134,12 @@ static inline int snd_sof_dsp_runtime_suspend(struct snd_sof_dev *sdev, return 0; } +static inline void snd_sof_dsp_hw_params_upon_resume(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->set_hw_params_upon_resume) + sof_ops(sdev)->set_hw_params_upon_resume(sdev); +} + static inline int snd_sof_dsp_set_clk(struct snd_sof_dev *sdev, u32 freq) { if (sof_ops(sdev)->set_clk) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index fc599e1b6f65..8ef1d51025d8 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -227,6 +227,9 @@ static void sof_set_hw_params_upon_resume(struct snd_sof_dev *sdev) spcm->hw_params_upon_resume[dir] = 1; } } + + /* set internal flag for BE */ + snd_sof_dsp_hw_params_upon_resume(sdev); } #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index bbc285018f9a..170adc21ef17 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -166,6 +166,7 @@ struct snd_sof_dsp_ops { int (*runtime_suspend)(struct snd_sof_dev *sof_dev, int state); /* optional */ int (*runtime_resume)(struct snd_sof_dev *sof_dev); /* optional */ + void (*set_hw_params_upon_resume)(struct snd_sof_dev *sdev); /* optional */ /* DSP clocking */ int (*set_clk)(struct snd_sof_dev *sof_dev, u32 freq); /* optional */ -- cgit v1.2.3-59-g8ed1b From 5e4a27fda2c8f1796fb673e72ae5dfc51675a785 Mon Sep 17 00:00:00 2001 From: Pan Xiuli Date: Tue, 30 Apr 2019 18:09:32 -0500 Subject: ASoC: SOF: IPC: add ipc dump function Dump IPC status when IPC timed out. IPC status is platform specific and need bind to plaform. Signed-off-by: Pan Xiuli Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/ipc.c | 1 + sound/soc/sof/ops.h | 6 ++++++ sound/soc/sof/sof-priv.h | 1 + 3 files changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 437e80a07bc2..ba1bb17a8d1e 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -207,6 +207,7 @@ static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg, dev_err(sdev->dev, "error: ipc timed out for 0x%x size %d\n", hdr->cmd, hdr->size); snd_sof_dsp_dbg_dump(ipc->sdev, SOF_DBG_REGS | SOF_DBG_MBOX); + snd_sof_ipc_dump(ipc->sdev); snd_sof_trace_notify_for_error(ipc->sdev); ret = -ETIMEDOUT; } else { diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index b8e2bf1fee24..80fc3b374c2b 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -155,6 +155,12 @@ static inline void snd_sof_dsp_dbg_dump(struct snd_sof_dev *sdev, u32 flags) return sof_ops(sdev)->dbg_dump(sdev, flags); } +static inline void snd_sof_ipc_dump(struct snd_sof_dev *sdev) +{ + if (sof_ops(sdev)->ipc_dump) + return sof_ops(sdev)->ipc_dump(sdev); +} + /* register IO */ static inline void snd_sof_dsp_write(struct snd_sof_dev *sdev, u32 bar, u32 offset, u32 value) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 170adc21ef17..1e85d6f9c5c3 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -176,6 +176,7 @@ struct snd_sof_dsp_ops { int debug_map_count; /* optional */ void (*dbg_dump)(struct snd_sof_dev *sof_dev, u32 flags); /* optional */ + void (*ipc_dump)(struct snd_sof_dev *sof_dev); /* optional */ /* host DMA trace initialization */ int (*trace_init)(struct snd_sof_dev *sdev, -- cgit v1.2.3-59-g8ed1b From f3da49f055dc55825984c9218fe06b913837cb13 Mon Sep 17 00:00:00 2001 From: Pan Xiuli Date: Tue, 30 Apr 2019 18:09:33 -0500 Subject: ASoC: SOF: Intel: APL: add ipc dump function Add IPC dump function for APL plaform Signed-off-by: Pan Xiuli Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/hda.c | 18 ++++++++++++++++++ sound/soc/sof/intel/hda.h | 1 + 3 files changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index d1fb2f2ba1e1..f215d80dce2c 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -55,6 +55,7 @@ const struct snd_sof_dsp_ops sof_apl_ops = { .debug_map = apl_dsp_debugfs, .debug_map_count = ARRAY_SIZE(apl_dsp_debugfs), .dbg_dump = hda_dsp_dump, + .ipc_dump = hda_ipc_dump, /* stream callbacks */ .pcm_open = hda_dsp_pcm_open, diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 84baf275b467..7e3980a2f7ba 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -179,6 +179,24 @@ void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags) } } +void hda_ipc_dump(struct snd_sof_dev *sdev) +{ + u32 hipcie; + u32 hipct; + u32 hipcctl; + + /* read IPC status */ + hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCIE); + hipct = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCT); + hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCCTL); + + /* dump the IPC regs */ + /* TODO: parse the raw msg */ + dev_err(sdev->dev, + "error: host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n", + hipcie, hipct, hipcctl); +} + static int hda_init(struct snd_sof_dev *sdev) { struct hda_bus *hbus; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 4b00e5d88565..92d45c43b4b1 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -447,6 +447,7 @@ int hda_dsp_runtime_resume(struct snd_sof_dev *sdev); void hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev); void hda_dsp_dump_skl(struct snd_sof_dev *sdev, u32 flags); void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags); +void hda_ipc_dump(struct snd_sof_dev *sdev); /* * DSP PCM Operations. -- cgit v1.2.3-59-g8ed1b From dc20e5f315e1967f3d1abb4ede8594cf51bdda47 Mon Sep 17 00:00:00 2001 From: Pan Xiuli Date: Tue, 30 Apr 2019 18:09:34 -0500 Subject: ASoC: SOF: Intel: CNL: add ipc dump function Add ipc dump function to CNL+ platforms. Signed-off-by: Pan Xiuli Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/cnl.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 2eac20bbcaea..08a1a3d3c08d 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -158,6 +158,24 @@ static int cnl_ipc_send_msg(struct snd_sof_dev *sdev, return 0; } +static void cnl_ipc_dump(struct snd_sof_dev *sdev) +{ + u32 hipcctl; + u32 hipcida; + u32 hipctdr; + + /* read IPC status */ + hipcida = snd_sof_dsp_read(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDA); + hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCCTL); + hipctdr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCTDR); + + /* dump the IPC regs */ + /* TODO: parse the raw msg */ + dev_err(sdev->dev, + "error: host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n", + hipcida, hipctdr, hipcctl); +} + /* cannonlake ops */ const struct snd_sof_dsp_ops sof_cnl_ops = { /* probe and remove */ @@ -189,6 +207,7 @@ const struct snd_sof_dsp_ops sof_cnl_ops = { .debug_map = cnl_dsp_debugfs, .debug_map_count = ARRAY_SIZE(cnl_dsp_debugfs), .dbg_dump = hda_dsp_dump, + .ipc_dump = cnl_ipc_dump, /* stream callbacks */ .pcm_open = hda_dsp_pcm_open, -- cgit v1.2.3-59-g8ed1b From 2b131b5a25f5917028714fd708a3ca3916683e48 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Tue, 30 Apr 2019 17:37:42 -0500 Subject: ASoC: Intel: skl_hda_dsp_generic: add DMIC support Add dmic dai links using naming conventions used in previous machine drivers. Tested on whiskylake & icelake with SOF driver. Due to a missing topology file, the DMIC functionality could not be tested with the Skylake driver but was tested for non-regressions on a GeminiLake platform without DMICs. Signed-off-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_hda_dsp_common.c | 22 ++++++++++++++++++++-- sound/soc/intel/boards/skl_hda_dsp_common.h | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 17 ++++++++++++++++- 3 files changed, 37 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index 3fdbf239da74..8b68f41a5b88 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -78,7 +78,6 @@ struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { .platform_name = "0000:00:1f.3", .dpcm_playback = 1, .dpcm_capture = 1, - .init = NULL, .no_pcm = 1, }, { @@ -90,7 +89,26 @@ struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { .platform_name = "0000:00:1f.3", .dpcm_playback = 1, .dpcm_capture = 1, - .init = NULL, + .no_pcm = 1, + }, + { + .name = "dmic01", + .id = 6, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "dmic16k", + .id = 7, + .cpu_dai_name = "DMIC16k Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .dpcm_capture = 1, .no_pcm = 1, }, }; diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h index 87c50aff56cd..daa582e513b2 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -15,7 +15,7 @@ #include #include -#define HDA_DSP_MAX_BE_DAI_LINKS 5 +#define HDA_DSP_MAX_BE_DAI_LINKS 7 struct skl_hda_hdmi_pcm { struct list_head head; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index b9a21e64ead2..e69dc36b6600 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -97,6 +97,9 @@ static struct snd_soc_card hda_soc_card = { }; #define IDISP_DAI_COUNT 3 +#define HDAC_DAI_COUNT 2 +#define DMIC_DAI_COUNT 2 + /* there are two routes per iDisp output */ #define IDISP_ROUTE_COUNT (IDISP_DAI_COUNT * 2) #define IDISP_CODEC_MASK 0x4 @@ -112,8 +115,20 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) codec_count = hweight_long(codec_mask); if (codec_count == 1 && codec_mask & IDISP_CODEC_MASK) { - num_links = IDISP_DAI_COUNT; + num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT; num_route = IDISP_ROUTE_COUNT; + + /* + * rearrange the dai link array and make the + * dmic dai links follow idsp dai links for only + * num_links of dai links need to be registered + * to ASoC. + */ + for (i = 0; i < DMIC_DAI_COUNT; i++) { + skl_hda_be_dai_links[IDISP_DAI_COUNT + i] = + skl_hda_be_dai_links[IDISP_DAI_COUNT + + HDAC_DAI_COUNT + i]; + } } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) { num_links = ARRAY_SIZE(skl_hda_be_dai_links); num_route = ARRAY_SIZE(skl_hda_map), -- cgit v1.2.3-59-g8ed1b From 6dd5055f1d320399f5319326ab9051fa33873933 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Tue, 30 Apr 2019 17:37:43 -0500 Subject: ASoC: Intel: skl_hda_dsp_generic:refine code style Semicolon is better than comma. Signed-off-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index e69dc36b6600..fc52d3a32354 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -131,7 +131,7 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) } } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) { num_links = ARRAY_SIZE(skl_hda_be_dai_links); - num_route = ARRAY_SIZE(skl_hda_map), + num_route = ARRAY_SIZE(skl_hda_map); card->dapm_widgets = skl_hda_widgets; card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); } else { -- cgit v1.2.3-59-g8ed1b From f70abd75b7c6c04d3219d0b3a0f3f15411b042fb Mon Sep 17 00:00:00 2001 From: Bard liao Date: Tue, 30 Apr 2019 17:37:44 -0500 Subject: ASoC: Intel: add sof-rt5682 machine driver The machine driver is a generic machine driver for SOF with rt5682 codec. it currently supports BYT/CHT/ICL/CML. Depending on the SOC hdmi/speaker and DMIC support are added dynamically. Only add information related to SOF since the machine driver was only tested with SOF. Signed-off-by: Sathya Prakash M R Signed-off-by: Bard liao Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 16 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/sof_rt5682.c | 563 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 6 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 6 + sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 19 + sound/soc/intel/common/soc-acpi-intel-icl-match.c | 6 + 7 files changed, 618 insertions(+) create mode 100644 sound/soc/intel/boards/sof_rt5682.c (limited to 'sound') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index fa0c88311559..e39473a6a5d9 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -383,4 +383,20 @@ config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC +if SND_SOC_SOF_HDA_COMMON || SND_SOC_SOF_BAYTRAIL +config SND_SOC_INTEL_SOF_RT5682_MACH + tristate "SOF with rt5682 codec in I2S Mode" + depends on I2C && ACPI + depends on (SND_SOC_SOF_HDA_COMMON && MFD_INTEL_LPSS) ||\ + (SND_SOC_SOF_BAYTRAIL && X86_INTEL_LPSS) + select SND_SOC_RT5682 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI if SND_SOC_SOF_HDA_COMMON + help + This adds support for ASoC machine driver for SOF platforms + with rt5682 codec. + Say Y if you have such a device. + If unsure select "N". +endif ## SND_SOC_SOF_HDA_COMMON || SND_SOC_SOF_BAYTRAIL + endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index bf072ea299b7..451b3bd7d9c5 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -16,6 +16,7 @@ snd-soc-sst-cht-bsw-nau8824-objs := cht_bsw_nau8824.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o +snd-soc-sof_rt5682-objs := sof_rt5682.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o @@ -26,6 +27,7 @@ snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c new file mode 100644 index 000000000000..f28fb98cc306 --- /dev/null +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -0,0 +1,563 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2019 Intel Corporation. + +/* + * Intel SOF Machine Driver with Realtek rt5682 Codec + * and speaker codec MAX98357A + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/rt5682.h" +#include "../../codecs/hdac_hdmi.h" + +#define NAME_SIZE 32 + +#define SOF_RT5682_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) +#define SOF_RT5682_SSP_CODEC_MASK (GENMASK(2, 0)) +#define SOF_RT5682_MCLK_EN BIT(3) +#define SOF_RT5682_MCLK_24MHZ BIT(4) +#define SOF_SPEAKER_AMP_PRESENT BIT(5) +#define SOF_RT5682_SSP_AMP(quirk) ((quirk) & GENMASK(8, 6)) +#define SOF_RT5682_SSP_AMP_MASK (GENMASK(8, 6)) +#define SOF_RT5682_SSP_AMP_SHIFT 6 + +/* Default: MCLK on, MCLK 19.2M, SSP0 */ +static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0); + +static int is_legacy_cpu; + +static struct snd_soc_jack sof_hdmi[3]; + +struct sof_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct sof_card_private { + struct snd_soc_jack sof_headset; + struct list_head hdmi_pcm_list; +}; + +static int sof_rt5682_quirk_cb(const struct dmi_system_id *id) +{ + sof_rt5682_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_rt5682_quirk_table[] = { + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "WhiskeyLake Client"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(1)), + }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Hatch"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0)), + }, + {} +}; + +static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct sof_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + /* dai_link id is 1:1 mapped to the PCM device */ + pcm->device = rtd->dai_link->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + int ret; + + /* need to enable ASRC function for 24MHz mclk rate */ + if ((sof_rt5682_quirk & SOF_RT5682_MCLK_EN) && + (sof_rt5682_quirk & SOF_RT5682_MCLK_24MHZ)) { + rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | + RT5682_AD_STEREO1_FILTER, + RT5682_CLK_SEL_I2S1_ASRC); + } + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->sof_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int clk_id, clk_freq, pll_out, ret; + + if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) { + clk_id = RT5682_PLL1_S_MCLK; + if (sof_rt5682_quirk & SOF_RT5682_MCLK_24MHZ) + clk_freq = 24000000; + else + clk_freq = 19200000; + } else { + clk_id = RT5682_PLL1_S_BCLK1; + clk_freq = params_rate(params) * 50; + } + + pll_out = params_rate(params) * 512; + + ret = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_pll err = %d\n", ret); + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* + * slot_width should equal or large than data length, set them + * be the same + */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, + params_width(params)); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops sof_rt5682_ops = { + .hw_params = sof_rt5682_hw_params, +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + +static int sof_card_late_probe(struct snd_soc_card *card) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + char jack_name[NAME_SIZE]; + struct sof_hdmi_pcm *pcm; + int err = 0; + int i = 0; + + /* HDMI is not supported by SOF on Baytrail/CherryTrail */ + if (is_legacy_cpu) + return 0; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &sof_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &sof_hdmi[i]); + if (err < 0) + return err; + + i++; + } + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +static const struct snd_kcontrol_new sof_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget sof_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_route sof_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* other jacks */ + { "IN1P", NULL, "Headset Mic" }, + +}; + +static const struct snd_soc_dapm_route speaker_map[] = { + /* speaker */ + { "Spk", NULL, "Speaker" }, +}; + +static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map, + ARRAY_SIZE(speaker_map)); + + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +/* sof audio machine driver for rt5682 codec */ +static struct snd_soc_card sof_audio_card_rt5682 = { + .name = "sof_rt5682", + .owner = THIS_MODULE, + .controls = sof_controls, + .num_controls = ARRAY_SIZE(sof_controls), + .dapm_widgets = sof_widgets, + .num_dapm_widgets = ARRAY_SIZE(sof_widgets), + .dapm_routes = sof_map, + .num_dapm_routes = ARRAY_SIZE(sof_map), + .fully_routed = true, + .late_probe = sof_card_late_probe, +}; + +static const struct x86_cpu_id legacy_cpi_ids[] = { + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_SILVERMONT }, /* Baytrail */ + { X86_VENDOR_INTEL, 6, INTEL_FAM6_ATOM_AIRMONT }, /* Cherrytrail */ + {} +}; + +static struct snd_soc_dai_link_component rt5682_component[] = { + { + .name = "i2c-10EC5682:00", + .dai_name = "rt5682-aif1", + } +}; + +static struct snd_soc_dai_link_component dmic_component[] = { + { + .name = "dmic-codec", + .dai_name = "dmic-hifi", + } +}; + +static struct snd_soc_dai_link_component max98357a_component[] = { + { + .name = "MX98357A:00", + .dai_name = "HiFi", + } +}; + +static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, + int ssp_codec, + int ssp_amp, + int dmic_num, + int hdmi_num) +{ + struct snd_soc_dai_link_component *idisp_components; + struct snd_soc_dai_link *links; + int i, id = 0; + + links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * + sof_audio_card_rt5682.num_links, GFP_KERNEL); + if (!links) + goto devm_err; + + /* codec SSP */ + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_codec); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].codecs = rt5682_component; + links[id].num_codecs = ARRAY_SIZE(rt5682_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_rt5682_codec_init; + links[id].ops = &sof_rt5682_ops; + links[id].nonatomic = true; + links[id].dpcm_playback = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + if (is_legacy_cpu) { + links[id].cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, + "ssp%d-port", + ssp_codec); + if (!links[id].cpu_dai_name) + goto devm_err; + } else { + /* + * Currently, On SKL+ platforms MCLK will be turned off in sof + * runtime suspended, and it will go into runtime suspended + * right after playback is stop. However, rt5682 will output + * static noise if sysclk turns off during playback. Set + * ignore_pmdown_time to power down rt5682 immediately and + * avoid the noise. + * It can be removed once we can control MCLK by driver. + */ + links[id].ignore_pmdown_time = 1; + links[id].cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_codec); + if (!links[id].cpu_dai_name) + goto devm_err; + } + id++; + + /* dmic */ + for (i = 1; i <= dmic_num; i++) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "dmic%02d", i); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, + "DMIC%02d Pin", i); + if (!links[id].cpu_dai_name) + goto devm_err; + + links[id].codecs = dmic_component; + links[id].num_codecs = ARRAY_SIZE(dmic_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].ignore_suspend = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + id++; + } + + /* HDMI */ + if (hdmi_num > 0) { + idisp_components = devm_kzalloc(dev, + sizeof(struct snd_soc_dai_link_component) * + hdmi_num, GFP_KERNEL); + if (!idisp_components) + goto devm_err; + } + for (i = 1; i <= hdmi_num; i++) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d Pin", i); + if (!links[id].cpu_dai_name) + goto devm_err; + + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = devm_kasprintf(dev, + GFP_KERNEL, + "intel-hdmi-hifi%d", + i); + if (!idisp_components[i - 1].dai_name) + goto devm_err; + + links[id].codecs = &idisp_components[i - 1]; + links[id].num_codecs = 1; + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_hdmi_init; + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + id++; + } + + /* speaker amp */ + if (sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_amp); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].codecs = max98357a_component; + links[id].num_codecs = ARRAY_SIZE(max98357a_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = speaker_codec_init, + links[id].nonatomic = true; + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + if (is_legacy_cpu) { + links[id].cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, + "ssp%d-port", + ssp_amp); + if (!links[id].cpu_dai_name) + goto devm_err; + + } else { + links[id].cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_amp); + if (!links[id].cpu_dai_name) + goto devm_err; + } + } + + return links; +devm_err: + return NULL; +} + +static int sof_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *dai_links; + struct snd_soc_acpi_mach *mach; + struct sof_card_private *ctx; + int dmic_num, hdmi_num; + int ret, ssp_amp, ssp_codec; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + if (x86_match_cpu(legacy_cpi_ids)) { + is_legacy_cpu = 1; + dmic_num = 0; + hdmi_num = 0; + /* default quirk for legacy cpu */ + sof_rt5682_quirk = SOF_RT5682_SSP_CODEC(2); + } else { + dmic_num = 1; + hdmi_num = 3; + } + + dmi_check_system(sof_rt5682_quirk_table); + + dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk); + + ssp_amp = (sof_rt5682_quirk & SOF_RT5682_SSP_AMP_MASK) >> + SOF_RT5682_SSP_AMP_SHIFT; + + ssp_codec = sof_rt5682_quirk & SOF_RT5682_SSP_CODEC_MASK; + + /* compute number of dai links */ + sof_audio_card_rt5682.num_links = 1 + dmic_num + hdmi_num; + if (sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) + sof_audio_card_rt5682.num_links++; + + dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, + dmic_num, hdmi_num); + if (!dai_links) + return -ENOMEM; + + sof_audio_card_rt5682.dai_link = dai_links; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + sof_audio_card_rt5682.dev = &pdev->dev; + mach = (&pdev->dev)->platform_data; + + /* set platform name for each dailink */ + ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_rt5682, + mach->mach_params.platform); + if (ret) + return ret; + + snd_soc_card_set_drvdata(&sof_audio_card_rt5682, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &sof_audio_card_rt5682); +} + +static struct platform_driver sof_audio = { + .probe = sof_audio_probe, + .driver = { + .name = "sof_rt5682", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(sof_audio) + +/* Module information */ +MODULE_DESCRIPTION("SOF Audio Machine driver"); +MODULE_AUTHOR("Bard Liao "); +MODULE_AUTHOR("Sathya Prakash M R "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sof_rt5682"); diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index fe812a909db4..0cfab247876a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -185,6 +185,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_baytrail_machines[] = { .sof_fw_filename = "sof-byt.ri", .sof_tplg_filename = "sof-byt-es8316.tplg", }, + { + .id = "10EC5682", + .drv_name = "sof_rt5682", + .sof_fw_filename = "sof-byt.ri", + .sof_tplg_filename = "sof-byt-rt5682.tplg", + }, /* some Baytrail platforms rely on RT5645, use CHT machine driver */ { .id = "10EC5645", diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index deafd87cc764..ff9c31a39ad4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -160,6 +160,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .sof_fw_filename = "sof-cht.ri", .sof_tplg_filename = "sof-cht-rt5640.tplg", }, + { + .id = "10EC5682", + .drv_name = "sof_rt5682", + .sof_fw_filename = "sof-cht.ri", + .sof_tplg_filename = "sof-cht-rt5682.tplg", + }, /* some CHT-T platforms rely on RT5651, use Baytrail machine driver */ { .id = "10EC5651", diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index a914dd238d0a..df7c52cad5c3 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -14,6 +14,11 @@ static struct skl_machine_pdata cnl_pdata = { .use_tplg_pcm = true, }; +static struct snd_soc_acpi_codecs cml_codecs = { + .num_codecs = 1, + .codecs = {"10EC5682"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { { .id = "INT34C2", @@ -23,6 +28,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .sof_fw_filename = "sof-cnl.ri", .sof_tplg_filename = "sof-cnl-rt274.tplg", }, + { + .id = "10EC5682", + .drv_name = "sof_rt5682", + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-rt5682.tplg", + }, + { + .id = "MX98357A", + .drv_name = "sof_rt5682", + .quirk_data = &cml_codecs, + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg", + }, + {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines); diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index e5a6be5bc0ee..0b430b9b3673 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -23,6 +23,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = { .sof_fw_filename = "sof-icl.ri", .sof_tplg_filename = "sof-icl-rt274.tplg", }, + { + .id = "10EC5682", + .drv_name = "sof_rt5682", + .sof_fw_filename = "sof-icl.ri", + .sof_tplg_filename = "sof-icl-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_machines); -- cgit v1.2.3-59-g8ed1b From 16ec5dfe0327ddcf279957bffe4c8fe527088c63 Mon Sep 17 00:00:00 2001 From: Mac Chiang Date: Thu, 2 May 2019 14:12:04 +0800 Subject: ASoC: Intel: kbl_da7219_max98357a: Map BTN_0 to KEY_PLAYPAUSE On kbl_rt5663_max98927, commit 38a5882e4292 ("ASoC: Intel: kbl_rt5663_max98927: Map BTN_0 to KEY_PLAYPAUSE") This key pair mapping to play/pause when playing Youtube The Android 3.5mm Headset jack specification mentions that BTN_0 should be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE, which has much broader userspace support. For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle play/pause of videos and audio, but does not handle KEY_MEDIA. Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the new USB headset spec requires KEY_PLAYPAUSE for BTN_0. https://source.android.com/devices/accessories/headset/usb-headset-spec The same fix is required on Chrome kbl_da7219_max98357a. Signed-off-by: Mac Chiang Reviewed-by: Benson Leung Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 38f6ab74709d..07491a0f8fb8 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -188,7 +188,7 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); -- cgit v1.2.3-59-g8ed1b From 2854cd34fbab5f28a356d3667c26b7856a7b73e2 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 6 May 2019 15:51:19 +0300 Subject: ALSA: synth: emux: soundfont.c: divide by zero in calc_gus_envelope_time() This function is called from load_guspatch() and the rate is specified by the user. If they accidentally selected zero then it would crash the kernel. I've just changed the zero to a one. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/synth/emux/soundfont.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 31a4ea94830e..9b5d70104489 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -856,6 +856,8 @@ calc_gus_envelope_time(int rate, int start, int end) int r, p, t; r = (3 - ((rate >> 6) & 3)) * 3; p = rate & 0x3f; + if (!p) + p = 1; t = end - start; if (t < 0) t = -t; if (13 > r) -- cgit v1.2.3-59-g8ed1b From 1587a061ef562de0d97c82a95863e191bcd69d63 Mon Sep 17 00:00:00 2001 From: Baolin Wang Date: Fri, 12 Apr 2019 14:40:17 +0800 Subject: ASoC: sprd: Add reserved DMA memory support For Spreadtrum audio platform driver, it need allocate a larger DMA buffer dynamically to copy audio data between userspace and kernel space, but that will increase the risk of memory allocation failure especially the system is under heavy load situation. To make sure the audio can work in this scenario, we usually reserve one region of memory to be used as a shared pool of DMA buffers for the platform component. So add of_reserved_mem_device_init_by_idx() function to initialize the shared pool of DMA buffers to be used by the platform component. Signed-off-by: Baolin Wang Signed-off-by: Mark Brown --- sound/soc/sprd/sprd-pcm-dma.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/sprd/sprd-pcm-dma.c b/sound/soc/sprd/sprd-pcm-dma.c index 9be6d4b2bf74..d38ebbbbf169 100644 --- a/sound/soc/sprd/sprd-pcm-dma.c +++ b/sound/soc/sprd/sprd-pcm-dma.c @@ -6,6 +6,7 @@ #include #include #include +#include #include #include #include @@ -530,8 +531,14 @@ static const struct snd_soc_component_driver sprd_soc_component = { static int sprd_soc_platform_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; int ret; + ret = of_reserved_mem_device_init_by_idx(&pdev->dev, np, 0); + if (ret) + dev_warn(&pdev->dev, + "no reserved DMA memory for audio platform device\n"); + ret = devm_snd_soc_register_component(&pdev->dev, &sprd_soc_component, NULL, 0); if (ret) -- cgit v1.2.3-59-g8ed1b From 8c2e6728c2bf95765b724e07d0278ae97cd1ee0d Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Mon, 6 May 2019 22:09:31 +0800 Subject: ALSA: hda/hdmi - Read the pin sense from register when repolling The driver will check the monitor presence when resuming from suspend, starting poll or interrupt triggers. In these 3 situations, the jack_dirty will be set to 1 first, then the hda_jack.c reads the pin_sense from register, after reading the register, the jack_dirty will be set to 0. But hdmi_repoll_work() is enabled in these 3 situations, It will read the pin_sense a couple of times subsequently, since the jack_dirty is 0 now, It does not read the register anymore, instead it uses the shadow pin_sense which is read at the first time. It is meaningless to check the shadow pin_sense a couple of times, we need to read the register to check the real plugging state, so we set the jack_dirty to 1 in the hdmi_repoll_work(). Signed-off-by: Hui Wang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8b3ac690efa3..380001d6e797 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1663,6 +1663,11 @@ static void hdmi_repoll_eld(struct work_struct *work) container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work); struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; + struct hda_jack_tbl *jack; + + jack = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); + if (jack) + jack->jack_dirty = 1; if (per_pin->repoll_count++ > 6) per_pin->repoll_count = 0; -- cgit v1.2.3-59-g8ed1b From 7f641e26a6df9269cb25dd7a4b0a91d6586ed441 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Mon, 6 May 2019 22:09:32 +0800 Subject: ALSA: hda/hdmi - Consider eld_valid when reporting jack event On the machines with AMD GPU or Nvidia GPU, we often meet this issue: after s3, there are 4 HDMI/DP audio devices in the gnome-sound-setting even there is no any monitors plugged. When this problem happens, we check the /proc/asound/cardX/eld#N.M, we will find the monitor_present=1, eld_valid=0. The root cause is BIOS or GPU driver makes the PRESENCE valid even no monitor plugged, and of course the driver will not get the valid eld_data subsequently. In this situation, we should not report the jack_plugged event, to do so, let us change the function hdmi_present_sense_via_verbs(). In this function, it reads the pin_sense via snd_hda_pin_sense(), after calling this function, the jack_dirty is 0, and before exiting via_verbs(), we change the shadow pin_sense according to both monitor_present and eld_valid, then in the snd_hda_jack_report_sync(), since the jack_dirty is still 0, it will report jack event according to this modified shadow pin_sense. After this change, the driver will not report Jack_is_plugged event through hdmi_present_sense_via_verbs() if monitor_present is 1 and eld_valid is 0. Signed-off-by: Hui Wang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 380001d6e797..0c61c05503f5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1551,9 +1551,11 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, ret = !repoll || !eld->monitor_present || eld->eld_valid; jack = snd_hda_jack_tbl_get(codec, pin_nid); - if (jack) + if (jack) { jack->block_report = !ret; - + jack->pin_sense = (eld->monitor_present && eld->eld_valid) ? + AC_PINSENSE_PRESENCE : 0; + } mutex_unlock(&per_pin->lock); return ret; } -- cgit v1.2.3-59-g8ed1b From a46eb523220e242affb9a6bc9bb8efc05f4f7459 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Fri, 3 May 2019 12:32:14 -0700 Subject: ASoC: RT5677-SPI: Disable 16Bit SPI Transfers The current algorithm allows 3 types of transfers, 16bit, 32bit and burst. According to Realtek, 16bit transfers have a special restriction in that it is restricted to the memory region of 0x18020000 ~ 0x18021000. This region is the memory location of the I2C registers. The current algorithm does not uphold this restriction and therefore fails to complete writes. Since this has been broken for some time it likely no one is using it. Better to simply disable the 16 bit writes. This will allow users to properly load firmware over SPI without data corruption. Signed-off-by: Curtis Malainey Reviewed-by: Ben Zhang Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5677-spi.c | 35 ++++++++++++++++------------------- 1 file changed, 16 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 167a02773a0b..84b6bd8b50e1 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -58,13 +58,15 @@ static DEFINE_MUTEX(spi_mutex); * RT5677_SPI_READ/WRITE_32: Transfer 4 bytes * RT5677_SPI_READ/WRITE_BURST: Transfer any multiples of 8 bytes * - * For example, reading 260 bytes at 0x60030002 uses the following commands: - * 0x60030002 RT5677_SPI_READ_16 2 bytes + * Note: + * 16 Bit writes and reads are restricted to the address range + * 0x18020000 ~ 0x18021000 + * + * For example, reading 256 bytes at 0x60030004 uses the following commands: * 0x60030004 RT5677_SPI_READ_32 4 bytes * 0x60030008 RT5677_SPI_READ_BURST 240 bytes * 0x600300F8 RT5677_SPI_READ_BURST 8 bytes * 0x60030100 RT5677_SPI_READ_32 4 bytes - * 0x60030104 RT5677_SPI_READ_16 2 bytes * * Input: * @read: true for read commands; false for write commands @@ -79,15 +81,13 @@ static u8 rt5677_spi_select_cmd(bool read, u32 align, u32 remain, u32 *len) { u8 cmd; - if (align == 2 || align == 6 || remain == 2) { - cmd = RT5677_SPI_READ_16; - *len = 2; - } else if (align == 4 || remain <= 6) { + if (align == 4 || remain <= 4) { cmd = RT5677_SPI_READ_32; *len = 4; } else { cmd = RT5677_SPI_READ_BURST; - *len = min_t(u32, remain & ~7, RT5677_SPI_BURST_LEN); + *len = (((remain - 1) >> 3) + 1) << 3; + *len = min_t(u32, *len, RT5677_SPI_BURST_LEN); } return read ? cmd : cmd + 1; } @@ -108,7 +108,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) } } -/* Read DSP address space using SPI. addr and len have to be 2-byte aligned. */ +/* Read DSP address space using SPI. addr and len have to be 4-byte aligned. */ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) { u32 offset; @@ -124,7 +124,7 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) if (!g_spi) return -ENODEV; - if ((addr & 1) || (len & 1)) { + if ((addr & 3) || (len & 3)) { dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); return -EACCES; } @@ -159,13 +159,13 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) } EXPORT_SYMBOL_GPL(rt5677_spi_read); -/* Write DSP address space using SPI. addr has to be 2-byte aligned. - * If len is not 2-byte aligned, an extra byte of zero is written at the end +/* Write DSP address space using SPI. addr has to be 4-byte aligned. + * If len is not 4-byte aligned, then extra zeros are written at the end * as padding. */ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) { - u32 offset, len_with_pad = len; + u32 offset; int status = 0; struct spi_transfer t; struct spi_message m; @@ -178,22 +178,19 @@ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) if (!g_spi) return -ENODEV; - if (addr & 1) { + if (addr & 3) { dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len); return -EACCES; } - if (len & 1) - len_with_pad = len + 1; - memset(&t, 0, sizeof(t)); t.tx_buf = buf; t.speed_hz = RT5677_SPI_FREQ; spi_message_init_with_transfers(&m, &t, 1); - for (offset = 0; offset < len_with_pad;) { + for (offset = 0; offset < len;) { spi_cmd = rt5677_spi_select_cmd(false, (addr + offset) & 7, - len_with_pad - offset, &t.len); + len - offset, &t.len); /* Construct SPI message header */ buf[0] = spi_cmd; -- cgit v1.2.3-59-g8ed1b From ecb2795c08bc825ebd604997e5be440b060c5b18 Mon Sep 17 00:00:00 2001 From: Jon Hunter Date: Wed, 1 May 2019 15:29:38 +0100 Subject: ASoC: max98090: Fix restore of DAPM Muxes The max98090 driver defines 3 DAPM muxes; one for the right line output (LINMOD Mux), one for the left headphone mixer source (MIXHPLSEL Mux) and one for the right headphone mixer source (MIXHPRSEL Mux). The same bit is used for the mux as well as the DAPM enable, and although the mux can be correctly configured, after playback has completed, the mux will be reset during the disable phase. This is preventing the state of these muxes from being saved and restored correctly on system reboot. Fix this by marking these muxes as SND_SOC_NOPM. Note this has been verified this on the Tegra124 Nyan Big which features the MAX98090 codec. Signed-off-by: Jon Hunter Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/max98090.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 30c242c38d99..7619ea31ab50 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1194,14 +1194,14 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { &max98090_right_rcv_mixer_controls[0], ARRAY_SIZE(max98090_right_rcv_mixer_controls)), - SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER, - M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + SND_SOC_DAPM_MUX("LINMOD Mux", SND_SOC_NOPM, 0, 0, + &max98090_linmod_mux), - SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL, - M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", SND_SOC_NOPM, 0, 0, + &max98090_mixhplsel_mux), - SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, - M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", SND_SOC_NOPM, 0, 0, + &max98090_mixhprsel_mux), SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, M98090_HPLEN_SHIFT, 0, NULL, 0), -- cgit v1.2.3-59-g8ed1b From 863137f0bc5eb2a3a65d1d29778ac65642171b17 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 6 May 2019 14:44:04 +0200 Subject: ASoC: stm32: spdifrx: update pcm hardware constraints - Set period minimum size. Ensure at least 5ms period up to 48kHz/16 bits to prevent underrun/overrun. - Remove MDMA constraints on period maximum size and set period maximum to half the buffer maximum size. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_spdifrx.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index b4c3d983e195..aa83b50efabb 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -845,7 +845,8 @@ static struct snd_soc_dai_driver stm32_spdifrx_dai[] = { static const struct snd_pcm_hardware stm32_spdifrx_pcm_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP, .buffer_bytes_max = 8 * PAGE_SIZE, - .period_bytes_max = 2048, /* MDMA constraint */ + .period_bytes_min = 1024, + .period_bytes_max = 4 * PAGE_SIZE, .periods_min = 2, .periods_max = 8, }; -- cgit v1.2.3-59-g8ed1b From 19e42536b27121bcf6ee841b25054f8bacafd8c7 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 6 May 2019 14:44:05 +0200 Subject: ASoC: stm32: spdifrx: change trace level on iec control Change trace level to debug to avoid spurious messages. Return quietly when accessing iec958 control, while no S/PDIF signal is available. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_spdifrx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index aa83b50efabb..3d64200edbb5 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -496,7 +496,7 @@ static int stm32_spdifrx_get_ctrl_data(struct stm32_spdifrx_data *spdifrx) if (wait_for_completion_interruptible_timeout(&spdifrx->cs_completion, msecs_to_jiffies(100)) <= 0) { - dev_err(&spdifrx->pdev->dev, "Failed to get control data\n"); + dev_dbg(&spdifrx->pdev->dev, "Failed to get control data\n"); ret = -EAGAIN; } -- cgit v1.2.3-59-g8ed1b From 4c88519133bdd802fb0df4707b5a8c066af7154d Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Mon, 6 May 2019 12:01:40 -0700 Subject: ASoC: sound/soc/sof/: fix kconfig dependency warning Fix kconfig warning for unmet dependency for IOSF_MBI when PCI is not set/enabled. Fixes this warning: WARNING: unmet direct dependencies detected for IOSF_MBI Depends on [n]: PCI [=n] Selected by [y]: - SND_SOC_SOF_ACPI [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_SOF_TOPLEVEL [=y] && (ACPI [=y] || COMPILE_TEST [=n]) && X86 [=y] Signed-off-by: Randy Dunlap Cc: Liam Girdwood Cc: Mark Brown Cc: alsa-devel@alsa-project.org Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index a1a9ffe605dc..b204c65698f9 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -28,7 +28,7 @@ config SND_SOC_SOF_ACPI select SND_SOC_ACPI if ACPI select SND_SOC_SOF_OPTIONS select SND_SOC_SOF_INTEL_ACPI if SND_SOC_SOF_INTEL_TOPLEVEL - select IOSF_MBI if X86 + select IOSF_MBI if X86 && PCI help This adds support for ACPI enumeration. This option is required to enable Intel Haswell/Broadwell/Baytrail/Cherrytrail devices -- cgit v1.2.3-59-g8ed1b From ab0c433f32f86370152bc9cb4900e5fe9a11055c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2019 17:02:22 +0200 Subject: ASoC: da7219: Fix a compile warning at CONFIG_COMMON_CLK=n MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A trivial fix for the randconfig build error: sound/soc/codecs/da7219.c:2366:6: warning: unused variable ‘i’ [-Wunused-variable] Fixes: d90ba6c8b53e ("ASoC: da7219: Expose BCLK and WCLK control through CCF") Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/da7219.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 206d01c6eb7e..43c03e7b2f0e 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2357,7 +2357,9 @@ err_disable_reg: static void da7219_remove(struct snd_soc_component *component) { struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); +#ifdef CONFIG_COMMON_CLK int i; +#endif da7219_aad_exit(component); -- cgit v1.2.3-59-g8ed1b From c41d384c39f17ffb5326531da2374a1ab5859403 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2019 17:02:23 +0200 Subject: ASoC: SOF: Fix a compile warning with CONFIG_PCI=n MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A trivial fix for the randconfig build error: sound/soc/sof/ops.c:20:6: warning: ‘ret’ is used uninitialized in this function [-Wuninitialized] Fixes: d1d95fcb63e3 ("ASoC: SOF: Add DSP HW abstraction operations") Signed-off-by: Takashi Iwai Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ops.c b/sound/soc/sof/ops.c index 80f907740b82..7a27c3b719e7 100644 --- a/sound/soc/sof/ops.c +++ b/sound/soc/sof/ops.c @@ -17,7 +17,7 @@ bool snd_sof_pci_update_bits_unlocked(struct snd_sof_dev *sdev, u32 offset, { struct pci_dev *pci = to_pci_dev(sdev->dev); unsigned int old, new; - u32 ret; + u32 ret = 0; pci_read_config_dword(pci, offset, &ret); old = ret; -- cgit v1.2.3-59-g8ed1b From f153bf49ddf6169544b24ef5ea26fa3eb4996b95 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2019 17:02:24 +0200 Subject: ASoC: rockchip: Fix an uninitialized variable compile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Paper over a trivial case leading to an uninitialized variable compile warning: sound/soc/rockchip/rockchip_pdm.c:179:3: warning: ‘clk_out’ may be used uninitialized in this function [-Wmaybe-uninitialized] Fixes: 624e8e00acaf ("ASoC: rockchip: pdm: fixup pdm fractional div") Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_pdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 6c0f242db5ef..b9c1d8ad77c1 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -158,7 +158,7 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, struct rk_pdm_dev *pdm = to_info(dai); unsigned int val = 0; unsigned int clk_rate, clk_div, samplerate; - unsigned int clk_src, clk_out; + unsigned int clk_src, clk_out = 0; unsigned long m, n; bool change; int ret; -- cgit v1.2.3-59-g8ed1b From d3ba58bb895915f7f9105e0844441d2ca7d83340 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Mon, 6 May 2019 15:09:42 +0800 Subject: ALSA: hda/realtek - Support low power consumption for ALC295 Enter to close more power control widgets at suspend. Remove hp_pin check. Add the default pin 0x21 as headphone. Supported low power consumption, it must do depop procedure when headset jack was plugged or unplugged. So, alc225_init() and alc225_shutup() must run delay when headset jack was plugged or unplugged. If depop procedure not run with delay, it will have a chance to let power consumption raise high. [ A few compile fixes by tiwai ] Fixes: 8983eb602af5 ("ALSA: hda/realtek - Move to ACT_INIT state") Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++-------------- 1 file changed, 31 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d743f2b0e3cf..e733d323f1a7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -119,6 +119,7 @@ struct alc_spec { unsigned int no_depop_delay:1; unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; + unsigned int ultra_low_power:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -3269,8 +3270,7 @@ static void alc225_init(struct hda_codec *codec) bool hp1_pin_sense, hp2_pin_sense; if (!hp_pin) - return; - + hp_pin = 0x21; msleep(30); hp1_pin_sense = snd_hda_jack_detect(codec, hp_pin); @@ -3280,25 +3280,31 @@ static void alc225_init(struct hda_codec *codec) msleep(2); alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */ + if (spec->ultra_low_power) { + alc_update_coef_idx(codec, 0x08, 0x0f << 2, 3<<2); + alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6); + alc_update_coef_idx(codec, 0x33, 1<<11, 0); + msleep(30); + } - if (hp1_pin_sense) + if (hp1_pin_sense || spec->ultra_low_power) snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); if (hp2_pin_sense) snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - if (hp1_pin_sense || hp2_pin_sense) + if (hp1_pin_sense || hp2_pin_sense || spec->ultra_low_power) msleep(85); - if (hp1_pin_sense) + if (hp1_pin_sense || spec->ultra_low_power) snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); if (hp2_pin_sense) snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - if (hp1_pin_sense || hp2_pin_sense) + if (hp1_pin_sense || hp2_pin_sense || spec->ultra_low_power) msleep(100); alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); @@ -3311,11 +3317,8 @@ static void alc225_shutup(struct hda_codec *codec) hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp1_pin_sense, hp2_pin_sense; - if (!hp_pin) { - alc269_shutup(codec); - return; - } - + if (!hp_pin) + hp_pin = 0x21; /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3325,28 +3328,36 @@ static void alc225_shutup(struct hda_codec *codec) if (hp1_pin_sense || hp2_pin_sense) msleep(2); - if (hp1_pin_sense) + if (hp1_pin_sense || spec->ultra_low_power) snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); if (hp2_pin_sense) snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - if (hp1_pin_sense || hp2_pin_sense) + if (hp1_pin_sense || hp2_pin_sense || spec->ultra_low_power) msleep(85); - if (hp1_pin_sense) + if (hp1_pin_sense || spec->ultra_low_power) snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); if (hp2_pin_sense) snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); - if (hp1_pin_sense || hp2_pin_sense) + if (hp1_pin_sense || hp2_pin_sense || spec->ultra_low_power) msleep(100); alc_auto_setup_eapd(codec, false); alc_shutup_pins(codec); + if (spec->ultra_low_power) { + msleep(50); + alc_update_coef_idx(codec, 0x08, 0x0f << 2, 0x0c << 2); + alc_update_coef_idx(codec, 0x0e, 7<<6, 0); + alc_update_coef_idx(codec, 0x33, 1<<11, 1<<11); + alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); + msleep(30); + } } static void alc_default_init(struct hda_codec *codec) @@ -5526,7 +5537,12 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { + struct alc_spec *spec = codec->spec; + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->ultra_low_power = true; + break; case HDA_FIXUP_ACT_INIT: switch (codec->core.vendor_id) { case 0x10ec0295: -- cgit v1.2.3-59-g8ed1b From d4c2ccdb5855ce8786ebc66f7405096065d0c198 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 May 2019 15:25:00 -0500 Subject: ALSA: hda/intel: add CometLake PCI IDs Add PCI IDs for LP and H skews. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 789308f54785..0741eae23f10 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2378,6 +2378,12 @@ static const struct pci_device_id azx_ids[] = { /* Cannonlake */ { PCI_DEVICE(0x8086, 0x9dc8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* CometLake-LP */ + { PCI_DEVICE(0x8086, 0x02C8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* CometLake-H */ + { PCI_DEVICE(0x8086, 0x06C8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- cgit v1.2.3-59-g8ed1b From 80a5052db75131423b67f38b21958555d7d970e4 Mon Sep 17 00:00:00 2001 From: Jeremy Soller Date: Tue, 7 May 2019 17:11:08 -0400 Subject: ALSA: hdea/realtek - Headset fixup for System76 Gazelle (gaze14) On the System76 Gazelle (gaze14), there is a headset microphone input attached to 0x1a that does not have a jack detect. In order to get it working, the pin configuration needs to be set correctly, and the ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied. This is identical to the patch already applied for the System76 Darter Pro (darp5). Signed-off-by: Jeremy Soller Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e733d323f1a7..dacccacb5fe0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6948,6 +6948,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8550, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), -- cgit v1.2.3-59-g8ed1b From 54d198d5019dd98b9bcb9099a389608d7e2cccad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2019 21:39:10 +0200 Subject: ASoC: SOF: Propagate sof_get_ctrl_copy_params() error properly This fixes a compile warning below by properly handling the error code from sof_get_ctrl_copy_params(): include/linux/kernel.h:843:43: warning: 'sparams.pl_size' may be used uninitialized in this function [-Wmaybe-uninitialized] sound/soc/sof/ipc.c:639:34: note: 'sparams.pl_size' was declared here The function returns an error before setting sparams.pl_size, so it'd assign an uninitialized value at a later point. Fixes: 53e0c72d98ba ("ASoC: SOF: Add support for IPC IO between DSP and Host") Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/ipc.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index ba1bb17a8d1e..f0b9d3c53f6f 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -567,7 +567,7 @@ static int sof_set_get_large_ctrl_data(struct snd_sof_dev *sdev, size_t offset = 0; size_t msg_bytes; size_t pl_size; - int err = 0; + int err; int i; /* allocate max ipc size because we have at least one */ @@ -576,9 +576,13 @@ static int sof_set_get_large_ctrl_data(struct snd_sof_dev *sdev, return -ENOMEM; if (send) - sof_get_ctrl_copy_params(cdata->type, cdata, partdata, sparams); + err = sof_get_ctrl_copy_params(cdata->type, cdata, partdata, + sparams); else - sof_get_ctrl_copy_params(cdata->type, partdata, cdata, sparams); + err = sof_get_ctrl_copy_params(cdata->type, partdata, cdata, + sparams); + if (err < 0) + return err; msg_bytes = sparams->msg_bytes; pl_size = sparams->pl_size; -- cgit v1.2.3-59-g8ed1b From 78989ff8aeaddc183237da558ee07deee35fbbbd Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 May 2019 11:32:35 -0500 Subject: ASoC: SOF: xtensa: fix undefined references The SND_SOC_INTEL_COMMON Kconfig was removed months ago from SOF Kconfig files but is still selected instead of the correct SND_SOC_SOF_INTEL_COMMON kconfig which does select xtensa stuff, leading to the following errors. ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x120): undefined reference to `sof_xtensa_arch_ops' ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x180): undefined reference to `sof_xtensa_arch_ops' ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x1e0): undefined reference to `sof_xtensa_arch_ops' Reported-by: kbuild test robot Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 32ee0fabab92..603e0db4f012 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -36,7 +36,7 @@ config SND_SOC_SOF_INTEL_HIFI_EP_IPC config SND_SOC_SOF_INTEL_ATOM_HIFI_EP tristate - select SND_SOC_INTEL_COMMON + select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_INTEL_HIFI_EP_IPC help This option is not user-selectable but automagically handled by -- cgit v1.2.3-59-g8ed1b From ce38a75089f70f6380fc63a5478a7659b4eb3f47 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 May 2019 11:32:36 -0500 Subject: ASoC: SOF: core: fix undefined nocodec reference The existing code mistakenly uses IS_ENABLED in C code instead of as in conditional compilation, leading to the following error: ld: sound/soc/sof/core.o: in function `sof_machine_check': sound/soc/sof/core.c:279: undefined reference to `sof_nocodec_setup' Fix by using #if !IS_ENABLED() Reported-by: kbuild test robot Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 39cbd84ff9c8..5ddbfa8f1a28 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -265,11 +265,10 @@ static int sof_machine_check(struct snd_sof_dev *sdev) if (plat_data->machine) return 0; - if (!IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC)) { - dev_err(sdev->dev, "error: no matching ASoC machine driver found - aborting probe\n"); - return -ENODEV; - } - +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC) + dev_err(sdev->dev, "error: no matching ASoC machine driver found - aborting probe\n"); + return -ENODEV; +#else /* fallback to nocodec mode */ dev_warn(sdev->dev, "No ASoC machine driver found - using nocodec\n"); machine = devm_kzalloc(sdev->dev, sizeof(*machine), GFP_KERNEL); @@ -284,6 +283,7 @@ static int sof_machine_check(struct snd_sof_dev *sdev) plat_data->machine = machine; return 0; +#endif } static int sof_probe_continue(struct snd_sof_dev *sdev) -- cgit v1.2.3-59-g8ed1b From bdb8fa6b09a801a6cbe3c2487f0228860f9c5aba Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Tue, 7 May 2019 18:12:18 +0800 Subject: ASoC: mediatek: mt6358: initialize setting when ramping volume Mt6358 ramps up from the smallest volume (i.e. -10dB) to target dB when opening and ramps down from target dB to mute (i.e. -40dB) when closing. If target is equal to -10dB when opening, headset_volume_ramp() simply leaves current setting (which may not be -10dB) unchanged. Execute the loop at least once to initialize the setting to the starting point (i.e. from). Signed-off-by: Tzung-Bi Shih Signed-off-by: Mark Brown --- sound/soc/codecs/mt6358.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index d4c4fee6d3d9..96547ae50a5d 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -405,10 +405,9 @@ static bool is_valid_hp_pga_idx(int reg_idx) reg_idx == DL_GAIN_N_40DB; } -static void headset_volume_ramp(struct mt6358_priv *priv, - int from, int to) +static void headset_volume_ramp(struct mt6358_priv *priv, int from, int to) { - int offset = 0, count = 1, reg_idx; + int offset = 0, count = 0, reg_idx; if (!is_valid_hp_pga_idx(from) || !is_valid_hp_pga_idx(to)) dev_warn(priv->dev, "%s(), volume index is not valid, from %d, to %d\n", @@ -422,7 +421,7 @@ static void headset_volume_ramp(struct mt6358_priv *priv, else offset = from - to; - while (offset > 0) { + while (offset >= 0) { if (to > from) reg_idx = from + count; else -- cgit v1.2.3-59-g8ed1b From bbb56537a3279f7221e253afc29e834a3d99662f Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Tue, 7 May 2019 18:12:19 +0800 Subject: ASoC: mediatek: mt6358: save output volume for mixer controls Output volume settings from mixer controls would be lost. Imagine that "Headphone Volume" has set to -10dB via amixer: - in mtk_hp_enable() - hp_store_gain() saves the volume setting -10dB from regmap_read() to ana_gain[AUDIO_ANALOG_VOLUME_HPOUTL] - headset_volume_ramp() ramps up from -10dB to -10dB - in mtk_hp_disable() - headset_volume_ramp() ramps down from -10dB to -40dB Next time in mtk_hp_enable(), hp_store_gain() would save -40dB but not -10dB. As a result, headset_volume_ramp() would ramp from -10dB to -40dB (which is mute). Signed-off-by: Tzung-Bi Shih Signed-off-by: Mark Brown --- sound/soc/codecs/mt6358.c | 91 ++++++++++++++++++++++++++--------------------- 1 file changed, 51 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index 96547ae50a5d..44edbda8afcc 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -320,32 +320,6 @@ enum { #define DL_GAIN_N_40DB_REG (DL_GAIN_N_40DB << 7 | DL_GAIN_N_40DB) #define DL_GAIN_REG_MASK 0x0f9f -static void lo_store_gain(struct mt6358_priv *priv) -{ - unsigned int reg; - unsigned int gain_l, gain_r; - - regmap_read(priv->regmap, MT6358_ZCD_CON1, ®); - gain_l = (reg >> RG_AUDLOLGAIN_SFT) & RG_AUDLOLGAIN_MASK; - gain_r = (reg >> RG_AUDLORGAIN_SFT) & RG_AUDLORGAIN_MASK; - - priv->ana_gain[AUDIO_ANALOG_VOLUME_LINEOUTL] = gain_l; - priv->ana_gain[AUDIO_ANALOG_VOLUME_LINEOUTR] = gain_r; -} - -static void hp_store_gain(struct mt6358_priv *priv) -{ - unsigned int reg; - unsigned int gain_l, gain_r; - - regmap_read(priv->regmap, MT6358_ZCD_CON2, ®); - gain_l = (reg >> RG_AUDHPLGAIN_SFT) & RG_AUDHPLGAIN_MASK; - gain_r = (reg >> RG_AUDHPRGAIN_SFT) & RG_AUDHPRGAIN_MASK; - - priv->ana_gain[AUDIO_ANALOG_VOLUME_HPOUTL] = gain_l; - priv->ana_gain[AUDIO_ANALOG_VOLUME_HPOUTR] = gain_r; -} - static void hp_zcd_disable(struct mt6358_priv *priv) { regmap_write(priv->regmap, MT6358_ZCD_CON0, 0x0000); @@ -439,20 +413,62 @@ static void headset_volume_ramp(struct mt6358_priv *priv, int from, int to) } } +static int mt6358_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6358_priv *priv = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg; + int ret; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + switch (mc->reg) { + case MT6358_ZCD_CON2: + regmap_read(priv->regmap, MT6358_ZCD_CON2, ®); + priv->ana_gain[AUDIO_ANALOG_VOLUME_HPOUTL] = + (reg >> RG_AUDHPLGAIN_SFT) & RG_AUDHPLGAIN_MASK; + priv->ana_gain[AUDIO_ANALOG_VOLUME_HPOUTR] = + (reg >> RG_AUDHPRGAIN_SFT) & RG_AUDHPRGAIN_MASK; + break; + case MT6358_ZCD_CON1: + regmap_read(priv->regmap, MT6358_ZCD_CON1, ®); + priv->ana_gain[AUDIO_ANALOG_VOLUME_LINEOUTL] = + (reg >> RG_AUDLOLGAIN_SFT) & RG_AUDLOLGAIN_MASK; + priv->ana_gain[AUDIO_ANALOG_VOLUME_LINEOUTR] = + (reg >> RG_AUDLORGAIN_SFT) & RG_AUDLORGAIN_MASK; + break; + case MT6358_ZCD_CON3: + regmap_read(priv->regmap, MT6358_ZCD_CON3, ®); + priv->ana_gain[AUDIO_ANALOG_VOLUME_HSOUTL] = + (reg >> RG_AUDHSGAIN_SFT) & RG_AUDHSGAIN_MASK; + priv->ana_gain[AUDIO_ANALOG_VOLUME_HSOUTR] = + (reg >> RG_AUDHSGAIN_SFT) & RG_AUDHSGAIN_MASK; + break; + } + + return ret; +} + static const DECLARE_TLV_DB_SCALE(playback_tlv, -1000, 100, 0); static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 600, 0); static const struct snd_kcontrol_new mt6358_snd_controls[] = { /* dl pga gain */ - SOC_DOUBLE_TLV("Headphone Volume", - MT6358_ZCD_CON2, 0, 7, 0x12, 1, - playback_tlv), - SOC_DOUBLE_TLV("Lineout Volume", - MT6358_ZCD_CON1, 0, 7, 0x12, 1, - playback_tlv), - SOC_SINGLE_TLV("Handset Volume", - MT6358_ZCD_CON3, 0, 0x12, 1, - playback_tlv), + SOC_DOUBLE_EXT_TLV("Headphone Volume", + MT6358_ZCD_CON2, 0, 7, 0x12, 1, + snd_soc_get_volsw, mt6358_put_volsw, playback_tlv), + SOC_DOUBLE_EXT_TLV("Lineout Volume", + MT6358_ZCD_CON1, 0, 7, 0x12, 1, + snd_soc_get_volsw, mt6358_put_volsw, playback_tlv), + SOC_SINGLE_EXT_TLV("Handset Volume", + MT6358_ZCD_CON3, 0, 0x12, 1, + snd_soc_get_volsw, mt6358_put_volsw, playback_tlv), /* ul pga gain */ SOC_DOUBLE_R_TLV("PGA Volume", MT6358_AUDENC_ANA_CON0, MT6358_AUDENC_ANA_CON1, @@ -831,8 +847,6 @@ static int mtk_hp_enable(struct mt6358_priv *priv) /* Reduce ESD resistance of AU_REFN */ regmap_write(priv->regmap, MT6358_AUDDEC_ANA_CON2, 0x4000); - /* save target gain to restore after hardware open complete */ - hp_store_gain(priv); /* Set HPR/HPL gain as minimum (~ -40dB) */ regmap_write(priv->regmap, MT6358_ZCD_CON2, DL_GAIN_N_40DB_REG); @@ -1042,8 +1056,6 @@ static int mtk_hp_spk_enable(struct mt6358_priv *priv) /* Reduce ESD resistance of AU_REFN */ regmap_write(priv->regmap, MT6358_AUDDEC_ANA_CON2, 0x4000); - /* save target gain to restore after hardware open complete */ - hp_store_gain(priv); /* Set HPR/HPL gain to -10dB */ regmap_write(priv->regmap, MT6358_ZCD_CON2, DL_GAIN_N_10DB_REG); @@ -1103,7 +1115,6 @@ static int mtk_hp_spk_enable(struct mt6358_priv *priv) hp_main_output_ramp(priv, true); /* Set LO gain as minimum (~ -40dB) */ - lo_store_gain(priv); regmap_write(priv->regmap, MT6358_ZCD_CON1, DL_GAIN_N_40DB_REG); /* apply volume setting */ headset_volume_ramp(priv, -- cgit v1.2.3-59-g8ed1b From ec0574a6ce31f2d6640bb8a5edf98973356dd3b4 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Tue, 7 May 2019 18:12:20 +0800 Subject: ASoC: mediatek: mt6358: save PGA for mixer control mt6358_amic_disable() resets PGA to 0. Save the gain settings from mixer control and restore them when using the microphone. Signed-off-by: Tzung-Bi Shih Signed-off-by: Mark Brown --- sound/soc/codecs/mt6358.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index 44edbda8afcc..50b3fc5457ea 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -450,6 +450,15 @@ static int mt6358_put_volsw(struct snd_kcontrol *kcontrol, priv->ana_gain[AUDIO_ANALOG_VOLUME_HSOUTR] = (reg >> RG_AUDHSGAIN_SFT) & RG_AUDHSGAIN_MASK; break; + case MT6358_AUDENC_ANA_CON0: + case MT6358_AUDENC_ANA_CON1: + regmap_read(priv->regmap, MT6358_AUDENC_ANA_CON0, ®); + priv->ana_gain[AUDIO_ANALOG_VOLUME_MICAMP1] = + (reg >> RG_AUDPREAMPLGAIN_SFT) & RG_AUDPREAMPLGAIN_MASK; + regmap_read(priv->regmap, MT6358_AUDENC_ANA_CON1, ®); + priv->ana_gain[AUDIO_ANALOG_VOLUME_MICAMP2] = + (reg >> RG_AUDPREAMPRGAIN_SFT) & RG_AUDPREAMPRGAIN_MASK; + break; } return ret; @@ -470,10 +479,10 @@ static const struct snd_kcontrol_new mt6358_snd_controls[] = { MT6358_ZCD_CON3, 0, 0x12, 1, snd_soc_get_volsw, mt6358_put_volsw, playback_tlv), /* ul pga gain */ - SOC_DOUBLE_R_TLV("PGA Volume", - MT6358_AUDENC_ANA_CON0, MT6358_AUDENC_ANA_CON1, - 8, 4, 0, - pga_tlv), + SOC_DOUBLE_R_EXT_TLV("PGA Volume", + MT6358_AUDENC_ANA_CON0, MT6358_AUDENC_ANA_CON1, + 8, 4, 0, + snd_soc_get_volsw, mt6358_put_volsw, pga_tlv), }; /* MUX */ @@ -1750,6 +1759,21 @@ static void mt6358_dmic_disable(struct mt6358_priv *priv) regmap_write(priv->regmap, MT6358_AUDENC_ANA_CON9, 0x0000); } +static void mt6358_restore_pga(struct mt6358_priv *priv) +{ + unsigned int gain_l, gain_r; + + gain_l = priv->ana_gain[AUDIO_ANALOG_VOLUME_MICAMP1]; + gain_r = priv->ana_gain[AUDIO_ANALOG_VOLUME_MICAMP2]; + + regmap_update_bits(priv->regmap, MT6358_AUDENC_ANA_CON0, + RG_AUDPREAMPLGAIN_MASK_SFT, + gain_l << RG_AUDPREAMPLGAIN_SFT); + regmap_update_bits(priv->regmap, MT6358_AUDENC_ANA_CON1, + RG_AUDPREAMPRGAIN_MASK_SFT, + gain_r << RG_AUDPREAMPRGAIN_SFT); +} + static int mt_mic_type_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) @@ -1774,6 +1798,7 @@ static int mt_mic_type_event(struct snd_soc_dapm_widget *w, mt6358_amic_enable(priv); break; } + mt6358_restore_pga(priv); break; case SND_SOC_DAPM_POST_PMD: -- cgit v1.2.3-59-g8ed1b From 939507f2779836919271ba0311723d1001d3993b Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Tue, 7 May 2019 14:33:11 +0800 Subject: ASoC: codec: hdac_hdmi: no checking monitor in hw_params Some userspace apps, like pulseaudio, may call open, hw_params, prepare to judge whether the pcm is ready or not. Current hdac_hdmi will return -ENODEV if monitor is not connected, which will cause the apps believe the pcm is not ready. Actually PCM for hdmi is ready, even the monitor is not connected. This patch removes the check of monitor presence in hw_params, just like what the legacy HD-Audio driver does. Signed-off-by: Libin Yang Acked-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hdmi.c | 13 ------------- 1 file changed, 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4de1fbfa8827..660e0587f399 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -455,24 +455,11 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hparams, struct snd_soc_dai *dai) { struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); - struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; - struct hdac_hdmi_port *port; struct hdac_hdmi_pcm *pcm; int format; dai_map = &hdmi->dai_map[dai->id]; - port = dai_map->port; - - if (!port) - return -ENODEV; - - if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { - dev_err(&hdev->dev, - "device is not configured for this pin:port%d:%d\n", - port->pin->nid, port->id); - return -ENODEV; - } format = snd_hdac_calc_stream_format(params_rate(hparams), params_channels(hparams), params_format(hparams), -- cgit v1.2.3-59-g8ed1b From 4fc19fffaaf87335aafaeb059a561ef91aa6031c Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Mon, 6 May 2019 14:54:11 +0200 Subject: ASoC: stm32: i2s: update pcm hardware constraints - Set period minimum size. Ensure at least 5ms period up to 48kHz/16 bits to prevent underrun/overrun. - Remove MDMA constraints on period maximum size and set period maximum to half the buffer maximum size. Signed-off-by: Olivier Moysan Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 97d5e9901a0e..8ee697ff1f86 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -731,7 +731,8 @@ static const struct snd_soc_dai_ops stm32_i2s_pcm_dai_ops = { static const struct snd_pcm_hardware stm32_i2s_pcm_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP, .buffer_bytes_max = 8 * PAGE_SIZE, - .period_bytes_max = 2048, + .period_bytes_min = 1024, + .period_bytes_max = 4 * PAGE_SIZE, .periods_min = 2, .periods_max = 8, }; -- cgit v1.2.3-59-g8ed1b From 6447c962bc47a58c5ddb27d1c7c7dc45ca90bad0 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 8 May 2019 16:27:03 +0800 Subject: ALSA: hda/realtek - Support low power consumption for ALC256 Remove hp_pin = 0 return. Add the default pin 0x21 as headphone. Supported low power consumption, it must do depop procedure when headset jack was plugged or unplugged. So, alc256_init() and alc256_shutup() must run delay when headset jack was plugged or unplugged. If depop procedure not run with delay, it will have a chance to let power consumption raise high. [ NOTE: this is a quite quite similar change for ALC295 by commit d3ba58bb8959 ("ALSA: hda/realtek - Support low power consumption for ALC295"), but applied to ALC256 codec instead -- tiwai ] Fixes: 8983eb602af5 ("ALSA: hda/realtek - Move to ACT_INIT state") Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 33 ++++++++++++++++++++++++--------- 1 file changed, 24 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dacccacb5fe0..c53ca589c930 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3197,7 +3197,7 @@ static void alc256_init(struct hda_codec *codec) bool hp_pin_sense; if (!hp_pin) - return; + hp_pin = 0x21; msleep(30); @@ -3207,17 +3207,25 @@ static void alc256_init(struct hda_codec *codec) msleep(2); alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */ + if (spec->ultra_low_power) { + alc_update_coef_idx(codec, 0x03, 1<<1, 1<<1); + alc_update_coef_idx(codec, 0x08, 3<<2, 3<<2); + alc_update_coef_idx(codec, 0x08, 7<<4, 0); + alc_update_coef_idx(codec, 0x3b, 1<<15, 0); + alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6); + msleep(30); + } snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - if (hp_pin_sense) + if (hp_pin_sense || spec->ultra_low_power) msleep(85); snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - if (hp_pin_sense) + if (hp_pin_sense || spec->ultra_low_power) msleep(100); alc_update_coef_idx(codec, 0x46, 3 << 12, 0); @@ -3232,10 +3240,8 @@ static void alc256_shutup(struct hda_codec *codec) hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!hp_pin) { - alc269_shutup(codec); - return; - } + if (!hp_pin) + hp_pin = 0x21; hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); @@ -3245,7 +3251,7 @@ static void alc256_shutup(struct hda_codec *codec) snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - if (hp_pin_sense) + if (hp_pin_sense || spec->ultra_low_power) msleep(85); /* 3k pull low control for Headset jack. */ @@ -3256,11 +3262,20 @@ static void alc256_shutup(struct hda_codec *codec) snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); - if (hp_pin_sense) + if (hp_pin_sense || spec->ultra_low_power) msleep(100); alc_auto_setup_eapd(codec, false); alc_shutup_pins(codec); + if (spec->ultra_low_power) { + msleep(50); + alc_update_coef_idx(codec, 0x03, 1<<1, 0); + alc_update_coef_idx(codec, 0x08, 7<<4, 7<<4); + alc_update_coef_idx(codec, 0x08, 3<<2, 0); + alc_update_coef_idx(codec, 0x3b, 1<<15, 1<<15); + alc_update_coef_idx(codec, 0x0e, 7<<6, 0); + msleep(30); + } } static void alc225_init(struct hda_codec *codec) -- cgit v1.2.3-59-g8ed1b From 534420c6ff87d3052540f1fd346e0adcff440819 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2019 14:45:38 +0200 Subject: ALSA: aica: Fix a long-time build breakage The build of aica sound driver has been broken since the timer API conversion and some code rewrite. This patch fixes the breakage by using the common substream field, as well as a bit cleaning up wrt the timer handling in the code. Fixes: d522bb6a105f ("ALSA: sh: aica: Convert timers to use timer_setup()") Signed-off-by: Takashi Iwai --- sound/sh/aica.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/sh/aica.c b/sound/sh/aica.c index e7fef3fce44a..a24e486d9d83 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -303,7 +303,7 @@ static void aica_period_elapsed(struct timer_list *t) { struct snd_card_aica *dreamcastcard = from_timer(dreamcastcard, t, timer); - struct snd_pcm_substream *substream = dreamcastcard->timer_substream; + struct snd_pcm_substream *substream = dreamcastcard->substream; /*timer function - so cannot sleep */ int play_period; struct snd_pcm_runtime *runtime; @@ -335,13 +335,6 @@ static void spu_begin_dma(struct snd_pcm_substream *substream) dreamcastcard = substream->pcm->private_data; /*get the queue to do the work */ schedule_work(&(dreamcastcard->spu_dma_work)); - /* Timer may already be running */ - if (unlikely(dreamcastcard->timer_substream)) { - mod_timer(&dreamcastcard->timer, jiffies + 4); - return; - } - timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); - dreamcastcard->timer_substream = substream; mod_timer(&dreamcastcard->timer, jiffies + 4); } @@ -379,8 +372,8 @@ static int snd_aicapcm_pcm_close(struct snd_pcm_substream { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; flush_work(&(dreamcastcard->spu_dma_work)); - if (dreamcastcard->timer_substream) - del_timer(&dreamcastcard->timer); + del_timer(&dreamcastcard->timer); + dreamcastcard->substream = NULL; kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -613,6 +606,7 @@ static int snd_aica_probe(struct platform_device *devptr) "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); /* Prepare to use the queue */ INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma); + timer_setup(&dreamcastcard->timer, aica_period_elapsed, 0); /* Load the PCM 'chip' */ err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) -- cgit v1.2.3-59-g8ed1b From 7f84ff68be05ec7a5d2acf8fdc734fe5897af48f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2019 15:01:24 +0200 Subject: ALSA: line6: toneport: Fix broken usage of timer for delayed execution The line6 toneport driver has code for some delayed initialization, and this hits the kernel Oops because mutex and other sleepable functions are used in the timer callback. Fix the abuse by a delayed work instead so that everything works gracefully. Reported-by: syzbot+a07d0142e74fdd595cfb@syzkaller.appspotmail.com Cc: Signed-off-by: Takashi Iwai --- sound/usb/line6/toneport.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c index 19bee725de00..325b07b98b3c 100644 --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -54,8 +54,8 @@ struct usb_line6_toneport { /* Firmware version (x 100) */ u8 firmware_version; - /* Timer for delayed PCM startup */ - struct timer_list timer; + /* Work for delayed PCM startup */ + struct delayed_work pcm_work; /* Device type */ enum line6_device_type type; @@ -241,9 +241,10 @@ static int snd_toneport_source_put(struct snd_kcontrol *kcontrol, return 1; } -static void toneport_start_pcm(struct timer_list *t) +static void toneport_start_pcm(struct work_struct *work) { - struct usb_line6_toneport *toneport = from_timer(toneport, t, timer); + struct usb_line6_toneport *toneport = + container_of(work, struct usb_line6_toneport, pcm_work.work); struct usb_line6 *line6 = &toneport->line6; line6_pcm_acquire(line6->line6pcm, LINE6_STREAM_MONITOR, true); @@ -393,7 +394,8 @@ static int toneport_setup(struct usb_line6_toneport *toneport) if (toneport_has_led(toneport)) toneport_update_led(toneport); - mod_timer(&toneport->timer, jiffies + TONEPORT_PCM_DELAY * HZ); + schedule_delayed_work(&toneport->pcm_work, + msecs_to_jiffies(TONEPORT_PCM_DELAY * 1000)); return 0; } @@ -405,7 +407,7 @@ static void line6_toneport_disconnect(struct usb_line6 *line6) struct usb_line6_toneport *toneport = (struct usb_line6_toneport *)line6; - del_timer_sync(&toneport->timer); + cancel_delayed_work_sync(&toneport->pcm_work); if (toneport_has_led(toneport)) toneport_remove_leds(toneport); @@ -422,7 +424,7 @@ static int toneport_init(struct usb_line6 *line6, struct usb_line6_toneport *toneport = (struct usb_line6_toneport *) line6; toneport->type = id->driver_info; - timer_setup(&toneport->timer, toneport_start_pcm, 0); + INIT_DELAYED_WORK(&toneport->pcm_work, toneport_start_pcm); line6->disconnect = line6_toneport_disconnect; -- cgit v1.2.3-59-g8ed1b From 7f6647ced298d56027218e47619b7f0d925fc83e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2019 10:50:37 +0200 Subject: ASoC: SOF: Fix unused variable warnings MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The recent fix for the build fix caused a couple of unused variable compiler warnings when CONFIG_SND_SOC_SOF_NOCODEC isn't set: sound/soc/sof/core.c:263:6: warning: unused variable ‘ret’ [-Wunused-variable] sound/soc/sof/core.c:262:28: warning: unused variable ‘machine’ [-Wunused-variable] Fix them by adding another ifdef. Fixes: ce38a75089f7 ("ASoC: SOF: core: fix undefined nocodec reference") Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 5ddbfa8f1a28..32105e0fabe8 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -259,8 +259,10 @@ int snd_sof_create_page_table(struct snd_sof_dev *sdev, static int sof_machine_check(struct snd_sof_dev *sdev) { struct snd_sof_pdata *plat_data = sdev->pdata; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC) struct snd_soc_acpi_mach *machine; int ret; +#endif if (plat_data->machine) return 0; -- cgit v1.2.3-59-g8ed1b