diff options
Diffstat (limited to 'sound/soc')
27 files changed, 261 insertions, 124 deletions
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 30490a259148..594c6c5b7838 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; /* PCM hardware DMA capabilities - platform specific */ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, .formats = AU1XPSC_PCM_FMTS, .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, .period_bytes_max = 4096 * 1024 - 1, diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 030d2454725f..f2653803ede8 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,7 +56,6 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o -obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 921b205de28a..df7c8c281d2f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0); static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); /* + * Gain control for earpiece amplifier + * 0 dB to 12 dB in 6 dB steps (mute instead of -6) + */ +static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1); + +/* * Capture gain after the ADCs * from 0 dB to 31 dB in 1 dB steps */ @@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 4, 3, 0, output_tvl), SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume", - TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl), + TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl), /* Common capture gain controls */ SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume", diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3b1d0993bed9..0275321ff8ab 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream, * required for LRC in master mode. The DACs or ADCs need a * valid audio path i.e. pin -> ADC or DAC -> pin before * the LRC will be enabled in master mode. */ - if (!master && cmd != SNDRV_PCM_TRIGGER_START) + if (!master || cmd != SNDRV_PCM_TRIGGER_START) return 0; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 442ea6f160fc..9f6be3d31ac0 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); static int wm8580_out_vu(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 24) & 0xff; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; int ret; u16 val; @@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, return 0; } -#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \ +#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \ + xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \ - .private_value = (reg_left) | ((shift) << 8) | \ - ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } static const struct snd_kcontrol_new wm8580_snd_controls[] = { SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume", @@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, reg = wm8580_read(codec, WM8580_PLLA4 + offset); reg &= ~0x3f; reg |= pll_div.prescale | pll_div.postscale << 1 | - pll_div.freqmode << 4; + pll_div.freqmode << 3; wm8580_write(codec, WM8580_PLLA4 + offset, reg); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index c518c3e5aa3f..40cd274eb1ef 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, +SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, &wm8990_dapm_ainlmux_controls, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, +SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, &wm8990_dapm_ainrmux_controls, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = { {"LIN12 PGA", "LIN2 Switch", "LIN2"}, /* LIN34 PGA */ {"LIN34 PGA", "LIN3 Switch", "LIN3"}, - {"LIN34 PGA", "LIN4 Switch", "LIN4"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, /* INMIXL */ {"INMIXL", "Record Left Volume", "LOMIX"}, {"INMIXL", "LIN2 Volume", "LIN2"}, {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, - /* AILNMUX */ - {"AILNMUX", "INMIXL Mix", "INMIXL"}, - {"AILNMUX", "DIFFINL Mix", "LIN12PGA"}, - {"AILNMUX", "DIFFINL Mix", "LIN34PGA"}, - {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"}, - {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* AINLMUX */ + {"AINLMUX", "INMIXL Mix", "INMIXL"}, + {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"}, + {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"}, + {"AINLMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AINLMUX", "RXVOICE Mix", "RIN4/RXP"}, /* ADC */ - {"Left ADC", NULL, "AILNMUX"}, + {"Left ADC", NULL, "AINLMUX"}, /* RIN12 PGA */ {"RIN12 PGA", "RIN1 Switch", "RIN1"}, {"RIN12 PGA", "RIN2 Switch", "RIN2"}, /* RIN34 PGA */ {"RIN34 PGA", "RIN3 Switch", "RIN3"}, - {"RIN34 PGA", "RIN4 Switch", "RIN4"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, /* INMIXL */ {"INMIXR", "Record Right Volume", "ROMIX"}, {"INMIXR", "RIN2 Volume", "RIN2"}, {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, - /* AIRNMUX */ - {"AIRNMUX", "INMIXR Mix", "INMIXR"}, - {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"}, - {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"}, - {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"}, - {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* AINRMUX */ + {"AINRMUX", "INMIXR Mix", "INMIXR"}, + {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"}, + {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"}, + {"AINRMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AINRMUX", "RXVOICE Mix", "RIN4/RXP"}, /* ADC */ - {"Right ADC", NULL, "AIRNMUX"}, + {"Right ADC", NULL, "AINRMUX"}, /* LOMIX */ {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, @@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, /* OUT3MIX */ - {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"}, + {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"}, {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, /* OUT4MIX */ @@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Output Pins */ {"LON", NULL, "LONMIX"}, {"LOP", NULL, "LOPMIX"}, - {"OUT", NULL, "OUT3MIX"}, + {"OUT3", NULL, "OUT3MIX"}, {"LOUT", NULL, "LOUT PGA"}, {"SPKN", NULL, "SPKMIX"}, {"ROUT", NULL, "ROUT PGA"}, diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 6e23a81dba78..c2d1a7a18fa3 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -318,7 +318,7 @@ static int wm9705_reset(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm9705_soc_suspend(struct platform_device *pdev) +static int wm9705_soc_suspend(struct platform_device *pdev, pm_message_t msg) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index bd7392c9657e..411a710be660 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S tristate config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM + tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" + depends on SND_DAVINCI_SOC + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci EVM platform. + DaVinci DM6446 or DM355 EVM platforms. config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 9b90b347007c..58fd1cbedd88 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -20,7 +20,11 @@ #include <sound/soc-dapm.h> #include <asm/dma.h> -#include <mach/hardware.h> +#include <asm/mach-types.h> + +#include <mach/asp.h> +#include <mach/edma.h> +#include <mach/mux.h> #include "../codecs/tlv320aic3x.h" #include "davinci-pcm.h" @@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = { /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { - .i2c_bus = 0, + .i2c_bus = 1, .i2c_address = 0x1b, }; @@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ static struct resource evm_snd_resources[] = { { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, .flags = IORESOURCE_MEM, }, }; static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DM644X_DMACH_MCBSP_TX, - .rx_dma_ch = DM644X_DMACH_MCBSP_RX, + .tx_dma_ch = DAVINCI_DMA_ASP0_TX, + .rx_dma_ch = DAVINCI_DMA_ASP0_RX, +}; + +/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ +static struct resource dm335evm_snd_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm335evm_snd_data = { + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { + struct resource *resources; + unsigned num_resources; + struct evm_snd_platform_data *data; + int index; int ret; - evm_snd_device = platform_device_alloc("soc-audio", 0); + if (machine_is_davinci_evm()) { + davinci_cfg_reg(DM644X_MCBSP); + + resources = evm_snd_resources; + num_resources = ARRAY_SIZE(evm_snd_resources); + data = &evm_snd_data; + index = 0; + } else if (machine_is_davinci_dm355_evm()) { + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + + resources = dm335evm_snd_resources; + num_resources = ARRAY_SIZE(dm335evm_snd_resources); + data = &dm335evm_snd_data; + index = 1; + } else + return -EINVAL; + + evm_snd_device = platform_device_alloc("soc-audio", index); if (!evm_snd_device) return -ENOMEM; platform_set_drvdata(evm_snd_device, &evm_snd_devdata); evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, &evm_snd_data, - sizeof(evm_snd_data)); + platform_device_add_data(evm_snd_device, data, sizeof(*data)); - ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, - ARRAY_SIZE(evm_snd_resources)); + ret = platform_device_add_resources(evm_snd_device, resources, + num_resources); if (ret) { platform_device_put(evm_snd_device); return ret; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ffdb9439d3d8..b1ea52fc83c7 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -24,6 +24,26 @@ #include "davinci-pcm.h" + +/* + * NOTE: terminology here is confusing. + * + * - This driver supports the "Audio Serial Port" (ASP), + * found on dm6446, dm355, and other DaVinci chips. + * + * - But it labels it a "Multi-channel Buffered Serial Port" + * (McBSP) as on older chips like the dm642 ... which was + * backward-compatible, possibly explaining that confusion. + * + * - OMAP chips have a controller called McBSP, which is + * incompatible with the DaVinci flavor of McBSP. + * + * - Newer DaVinci chips have a controller called McASP, + * incompatible with ASP and with either McBSP. + * + * In short: this uses ASP to implement I2S, not McBSP. + * And it won't be the only DaVinci implemention of I2S. + */ #define DAVINCI_MCBSP_DRR_REG 0x00 #define DAVINCI_MCBSP_DXR_REG 0x04 #define DAVINCI_MCBSP_SPCR_REG 0x08 @@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, cpu_dai->private_data = dev; - dev->clk = clk_get(&pdev->dev, "McBSPCLK"); + dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev, { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 7af3b5b3a53d..a05996588489 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -22,6 +22,7 @@ #include <sound/soc.h> #include <asm/dma.h> +#include <mach/edma.h> #include "davinci-pcm.h" @@ -51,7 +52,7 @@ struct davinci_runtime_data { spinlock_t lock; int period; /* current DMA period */ int master_lch; /* Master DMA channel */ - int slave_lch; /* Slave DMA channel */ + int slave_lch; /* linked parameter RAM reload slot */ struct davinci_pcm_dma_params *params; /* DMA params */ }; @@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dst_bidx = data_type; } - davinci_set_dma_src_params(lch, src, INCR, W8BIT); - davinci_set_dma_dest_params(lch, dst, INCR, W8BIT); - davinci_set_dma_src_index(lch, src_bidx, 0); - davinci_set_dma_dest_index(lch, dst_bidx, 0); - davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC); + edma_set_src(lch, src, INCR, W8BIT); + edma_set_dest(lch, dst, INCR, W8BIT); + edma_set_src_index(lch, src_bidx, 0); + edma_set_dest_index(lch, dst_bidx, 0); + edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) prtd->period = 0; } -static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data) +static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) { struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; @@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; - int tcc = TCC_ANY; + struct edmacc_param p_ram; int ret; if (!dma_data) @@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) prtd->params = dma_data; /* Request master DMA channel */ - ret = davinci_request_dma(prtd->params->channel, prtd->params->name, + ret = edma_alloc_channel(prtd->params->channel, davinci_pcm_dma_irq, substream, - &prtd->master_lch, &tcc, EVENTQ_0); - if (ret) + EVENTQ_0); + if (ret < 0) return ret; + prtd->master_lch = ret; - /* Request slave DMA channel */ - ret = davinci_request_dma(PARAM_ANY, "Link", - NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0); - if (ret) { - davinci_free_dma(prtd->master_lch); + /* Request parameter RAM reload slot */ + ret = edma_alloc_slot(EDMA_SLOT_ANY); + if (ret < 0) { + edma_free_channel(prtd->master_lch); return ret; } - - /* Link slave DMA channel in loopback */ - davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch); + prtd->slave_lch = ret; + + /* Issue transfer completion IRQ when the channel completes a + * transfer, then always reload from the same slot (by a kind + * of loopback link). The completion IRQ handler will update + * the reload slot with a new buffer. + * + * REVISIT save p_ram here after setting up everything except + * the buffer and its length (ccnt) ... use it as a template + * so davinci_pcm_enqueue_dma() takes less time in IRQ. + */ + edma_read_slot(prtd->slave_lch, &p_ram); + p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch); + p_ram.link_bcntrld = prtd->slave_lch << 5; + edma_write_slot(prtd->slave_lch, &p_ram); return 0; } @@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - davinci_start_dma(prtd->master_lch); + edma_start(prtd->master_lch); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - davinci_stop_dma(prtd->master_lch); + edma_stop(prtd->master_lch); break; default: ret = -EINVAL; @@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static int davinci_pcm_prepare(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct paramentry_descriptor temp; + struct edmacc_param temp; prtd->period = 0; davinci_pcm_enqueue_dma(substream); - /* Get slave channel dma params for master channel startup */ - davinci_get_dma_params(prtd->slave_lch, &temp); - davinci_set_dma_params(prtd->master_lch, &temp); + /* Copy self-linked parameter RAM entry into master channel */ + edma_read_slot(prtd->slave_lch, &temp); + edma_write_slot(prtd->master_lch, &temp); return 0; } @@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) spin_lock(&prtd->lock); - davinci_dma_getposition(prtd->master_lch, &src, &dst); + edma_get_position(prtd->master_lch, &src, &dst); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) count = src - runtime->dma_addr; else @@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; - davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch); + edma_unlink(prtd->slave_lch); - davinci_free_dma(prtd->slave_lch); - davinci_free_dma(prtd->master_lch); + edma_free_slot(prtd->slave_lch); + edma_free_channel(prtd->master_lch); kfree(prtd); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 3aa729df27b5..1111c710118a 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = { static const struct snd_pcm_hardware psc_i2s_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, .rate_min = 8000, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index a6d1178ce128..91ef17992de5 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * Contact: Jarkko Nikula <jhnikula@gmail.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -417,6 +417,6 @@ static void __exit n810_soc_exit(void) module_init(n810_soc_init); module_exit(n810_soc_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>"); MODULE_DESCRIPTION("ALSA SoC Nokia N810"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 9c09b94f0cf8..912614283848 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Peter Ujfalusi <peter.ujfalusi@nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -283,7 +284,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_DSP_B: regs->srgr2 |= FPER(wlen * channels - 1); - regs->srgr1 |= FWID(wlen * channels - 2); + regs->srgr1 |= FWID(0); break; } @@ -302,6 +303,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; @@ -328,6 +330,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); + /* Invert FS polarity configuration */ + temp_fmt ^= SND_SOC_DAIFMT_NB_IF; break; default: /* Unsupported data format */ @@ -351,7 +355,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. @@ -529,6 +533,6 @@ static void __exit snd_omap_mcbsp_exit(void) } module_exit(snd_omap_mcbsp_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>"); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index df7ad13ba73d..c8147aace813 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Peter Ujfalusi <peter.ujfalusi@nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 1bdbb0427183..07cf7f46b584 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Peter Ujfalusi <peter.ujfalusi@nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -367,6 +368,6 @@ static void __exit omap_soc_platform_exit(void) } module_exit(omap_soc_platform_exit); -MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); +MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>"); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index e4369bdfd77d..8d9d26916b05 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * Contact: Jarkko Nikula <jhnikula@gmail.com> + * Peter Ujfalusi <peter.ujfalusi@nokia.com> * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index a952a4eb3361..a4e149b7f0eb 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -62,7 +62,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, /* Set codec DAI configuration */ err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (err < 0) { printk(KERN_ERR "can't set codec DAI configuration\n"); @@ -72,7 +72,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, /* Set cpu DAI configuration */ err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (err < 0) { printk(KERN_ERR "can't set cpu DAI configuration\n"); diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index f7c4544f7859..0625c342a1c9 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -27,8 +27,6 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> -#include <mach/pxa-regs.h> -#include <mach/hardware.h> #include <mach/magician.h> #include <asm/mach-types.h> #include "../codecs/uda1380.h" diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 48a73f64500b..44fcc4e01e08 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -200,7 +200,7 @@ static struct snd_soc_device palm27x_snd_devdata = { static struct platform_device *palm27x_snd_device; -static int __init palm27x_asoc_init(void) +static int palm27x_asoc_probe(struct platform_device *pdev) { int ret; @@ -208,6 +208,10 @@ static int __init palm27x_asoc_init(void) machine_is_palmld())) return -ENODEV; + if (pdev->dev.platform_data) + palm27x_ep_gpio = ((struct palm27x_asoc_info *) + (pdev->dev.platform_data))->jack_gpio; + ret = gpio_request(palm27x_ep_gpio, "Headphone Jack"); if (ret) return ret; @@ -245,16 +249,31 @@ err_alloc: return ret; } -static void __exit palm27x_asoc_exit(void) +static int __devexit palm27x_asoc_remove(struct platform_device *pdev) { free_irq(gpio_to_irq(palm27x_ep_gpio), NULL); gpio_free(palm27x_ep_gpio); platform_device_unregister(palm27x_snd_device); + return 0; } -void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) +static struct platform_driver palm27x_wm9712_driver = { + .probe = palm27x_asoc_probe, + .remove = __devexit_p(palm27x_asoc_remove), + .driver = { + .name = "palm27x-asoc", + .owner = THIS_MODULE, + }, +}; + +static int __init palm27x_asoc_init(void) +{ + return platform_driver_register(&palm27x_wm9712_driver); +} + +static void __exit palm27x_asoc_exit(void) { - palm27x_ep_gpio = data->jack_gpio; + platform_driver_unregister(&palm27x_wm9712_driver); } module_init(palm27x_asoc_init); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657928d2..286be31545df 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -280,12 +280,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_dev *dev, u32 div) +static void ssp_set_scr(struct ssp_device *ssp, u32 div) { - struct ssp_device *ssp = dev->ssp; - u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * ssp_get_clkdiv - get SSP clock divider + */ +static u32 ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 div; - ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; } /* @@ -326,7 +347,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(&priv->dev, 1); + ssp_set_scr(ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -387,7 +408,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(&priv->dev, div); + ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -674,8 +695,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_I2S: sspsp = ssp_read_reg(ssp, SSPSP); - if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && - (width == 16)) { + if ((ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -806,6 +826,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, goto err_priv; } + priv->dai_fmt = (unsigned int) -1; dai->private_data = priv; return 0; diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 2f3a21eee051..df494d1e346f 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,10 +1,10 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 || ARCH_S3C64XX + depends on ARCH_S3C2410 help Say Y or M if you want to add support for codecs attached to - the S3C24XX and S3C64XX AC97, I2S or SSP interface. You will - also need to select the audio interfaces to support below. + the S3C24XX AC97 or I2S interfaces. You will also need to + select the audio interfaces to support below. config SND_S3C24XX_SOC_I2S tristate diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 32063790d95b..93e6c87b7399 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -69,8 +69,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream, break; } - s3c_i2sv2_calc_rate(&div, NULL, params_rate(params), - s3c2412_get_iisclk()); + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c2412_get_iisclk()); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | @@ -145,8 +145,9 @@ static struct snd_soc_dai_link jive_dai = { }; /* jive audio machine driver */ -static struct snd_soc_machine snd_soc_machine_jive = { +static struct snd_soc_card snd_soc_machine_jive = { .name = "Jive", + .platform = &s3c24xx_soc_platform, .dai_link = &jive_dai, .num_links = 1, }; @@ -157,9 +158,8 @@ static struct wm8750_setup_data jive_wm8750_setup = { /* jive audio subsystem */ static struct snd_soc_device jive_snd_devdata = { - .machine = &snd_soc_machine_jive, - .platform = &s3c24xx_soc_platform, - .codec_dev = &soc_codec_dev_wm8750_spi, + .card = &snd_soc_machine_jive, + .codec_dev = &soc_codec_dev_wm8750, .codec_data = &jive_wm8750_setup, }; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 295a4c910262..ab680aac3fcb 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -473,9 +473,9 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, /* default table of all avaialable root fs divisors */ static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; -int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk) +int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk) { unsigned long clkrate = clk_get_rate(clk); unsigned int div; @@ -531,7 +531,7 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, return 0; } -EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); +EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); int s3c_i2sv2_probe(struct platform_device *pdev, struct snd_soc_dai *dai, @@ -624,15 +624,18 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai) int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) { - dai->ops.trigger = s3c2412_i2s_trigger; - dai->ops.hw_params = s3c2412_i2s_hw_params; - dai->ops.set_fmt = s3c2412_i2s_set_fmt; - dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv; + struct snd_soc_dai_ops *ops = dai->ops; + + ops->trigger = s3c2412_i2s_trigger; + ops->hw_params = s3c2412_i2s_hw_params; + ops->set_fmt = s3c2412_i2s_set_fmt; + ops->set_clkdiv = s3c2412_i2s_set_clkdiv; dai->suspend = s3c2412_i2s_suspend; dai->resume = s3c2412_i2s_resume; return snd_soc_register_dai(dai); } - EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 1ca3cdaa8213..b7e0b3f0bfc8 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -33,8 +33,8 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/regs-gpio.h> #include <plat/audio.h> +#include <mach/regs-gpio.h> #include <mach/dma.h> #include "s3c24xx-pcm.h" diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 0dad3a0bb920..baddb1242c71 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH), .formats = DMABRG_FMTS, .rates = DMABRG_RATES, .rate_min = 8000, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 99712f652d0d..1cd149b9ce69 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + if (!card->instantiated) + return 0; + run_delayed_work(&card->delayed_work); if (platform->remove) |