diff options
Diffstat (limited to 'sound/soc')
55 files changed, 2195 insertions, 480 deletions
diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 9ca9214cb7fb..5f40517717c4 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -10,7 +10,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_MAX98357A select SND_SOC_ADAU7002 select REGULATOR - depends on SND_SOC_AMD_ACP && I2C + depends on SND_SOC_AMD_ACP && I2C && GPIOLIB help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index bc4dfafdfcd1..ea57088d50ce 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -631,7 +631,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); - return -ENODEV; + return -ENODEV; } adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 9a406144b18f..befc2a3a05b0 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -674,8 +674,13 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, dev->channels = channels; ret = regmap_write(dev->regmap, MCHP_I2SMCC_MRA, mra); - if (ret < 0) + if (ret < 0) { + if (dev->gclk_use) { + clk_unprepare(dev->gclk); + dev->gclk_use = 0; + } return ret; + } return regmap_write(dev->regmap, MCHP_I2SMCC_MRB, mrb); } @@ -690,31 +695,37 @@ static int mchp_i2s_mcc_hw_free(struct snd_pcm_substream *substream, err = wait_event_interruptible_timeout(dev->wq_txrdy, dev->tx_rdy, msecs_to_jiffies(500)); + if (err == 0) { + dev_warn_once(dev->dev, + "Timeout waiting for Tx ready\n"); + regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, + MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels)); + dev->tx_rdy = 1; + } } 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 (err == 0) { + dev_warn_once(dev->dev, + "Timeout waiting for Rx ready\n"); + regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, + MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels)); + dev->rx_rdy = 1; + } } 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); + clk_disable(dev->gclk); dev->gclk_running = 0; } + if (dev->gclk_use) { + clk_unprepare(dev->gclk); + dev->gclk_use = 0; + } } return 0; @@ -813,6 +824,8 @@ static int mchp_i2s_mcc_dai_probe(struct snd_soc_dai *dai) init_waitqueue_head(&dev->wq_txrdy); init_waitqueue_head(&dev->wq_rxrdy); + dev->tx_rdy = 1; + dev->rx_rdy = 1; snd_soc_dai_init_dma_data(dai, &dev->playback, &dev->capture); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 89238343e34d..bcac95785493 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -51,7 +51,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_BD28623 select SND_SOC_CQ0093VC - select SND_SOC_CROS_EC_CODEC if MFD_CROS_EC + select SND_SOC_CROS_EC_CODEC if CROS_EC select SND_SOC_CS35L32 if I2C select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C @@ -179,6 +179,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_STI_SAS select SND_SOC_TAS2552 if I2C + select SND_SOC_TAS2770 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C select SND_SOC_TAS5720 if I2C @@ -477,7 +478,7 @@ config SND_SOC_CQ0093VC config SND_SOC_CROS_EC_CODEC tristate "codec driver for ChromeOS EC" - depends on MFD_CROS_EC + depends on CROS_EC help If you say yes here you will get support for the ChromeOS Embedded Controller's Audio Codec. @@ -1104,6 +1105,10 @@ config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" depends on I2C +config SND_SOC_TAS2770 + tristate "Texas Instruments TAS2770 speaker amplifier" + depends on I2C + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c498373dcc5f..73b2d5982dcb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -196,6 +196,7 @@ snd-soc-tas571x-objs := tas571x.o snd-soc-tas5720-objs := tas5720.o snd-soc-tas6424-objs := tas6424.o snd-soc-tda7419-objs := tda7419.o +snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o @@ -479,6 +480,7 @@ obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o obj-$(CONFIG_SND_SOC_TAS6424) += snd-soc-tas6424.o obj-$(CONFIG_SND_SOC_TDA7419) += snd-soc-tda7419.o +obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 977f5a63be3f..5ca9b744b7d8 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -28,6 +28,10 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 +#define ADAU1761_ALC_CTRL1 0x4012 +#define ADAU1761_ALC_CTRL2 0x4013 +#define ADAU1761_ALC_CTRL3 0x4014 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -71,6 +75,10 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, + { ADAU1761_ALC_CTRL1, 0x00 }, + { ADAU1761_ALC_CTRL2, 0x00 }, + { ADAU1761_ALC_CTRL3, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -121,6 +129,10 @@ static const DECLARE_TLV_DB_SCALE(adau1761_sidetone_tlv, -1800, 300, 1); static const DECLARE_TLV_DB_SCALE(adau1761_boost_tlv, -600, 600, 1); static const DECLARE_TLV_DB_SCALE(adau1761_pga_boost_tlv, -2000, 2000, 1); +static const DECLARE_TLV_DB_SCALE(adau1761_alc_max_gain_tlv, -1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(adau1761_alc_target_tlv, -2850, 150, 0); +static const DECLARE_TLV_DB_SCALE(adau1761_alc_ng_threshold_tlv, -7650, 150, 0); + static const unsigned int adau1761_bias_select_values[] = { 0, 2, 3, }; @@ -147,6 +159,103 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static const char * const adau1761_alc_function_text[] = { + "Off", + "Right", + "Left", + "Stereo", + "DSP control", +}; + +static const char * const adau1761_alc_hold_time_text[] = { + "2.67 ms", + "5.34 ms", + "10.68 ms", + "21.36 ms", + "42.72 ms", + "85.44 ms", + "170.88 ms", + "341.76 ms", + "683.52 ms", + "1367 ms", + "2734.1 ms", + "5468.2 ms", + "10936 ms", + "21873 ms", + "43745 ms", + "87491 ms", +}; + +static const char * const adau1761_alc_attack_time_text[] = { + "6 ms", + "12 ms", + "24 ms", + "48 ms", + "96 ms", + "192 ms", + "384 ms", + "768 ms", + "1540 ms", + "3070 ms", + "6140 ms", + "12290 ms", + "24580 ms", + "49150 ms", + "98300 ms", + "196610 ms", +}; + +static const char * const adau1761_alc_decay_time_text[] = { + "24 ms", + "48 ms", + "96 ms", + "192 ms", + "384 ms", + "768 ms", + "15400 ms", + "30700 ms", + "61400 ms", + "12290 ms", + "24580 ms", + "49150 ms", + "98300 ms", + "196610 ms", + "393220 ms", + "786430 ms", +}; + +static const char * const adau1761_alc_ng_type_text[] = { + "Hold", + "Mute", + "Fade", + "Fade + Mute", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); +static SOC_ENUM_SINGLE_DECL(adau1761_alc_function_enum, + ADAU1761_ALC_CTRL0, 0, adau1761_alc_function_text); +static SOC_ENUM_SINGLE_DECL(adau1761_alc_hold_time_enum, + ADAU1761_ALC_CTRL1, 4, adau1761_alc_hold_time_text); +static SOC_ENUM_SINGLE_DECL(adau1761_alc_attack_time_enum, + ADAU1761_ALC_CTRL2, 4, adau1761_alc_attack_time_text); +static SOC_ENUM_SINGLE_DECL(adau1761_alc_decay_time_enum, + ADAU1761_ALC_CTRL2, 0, adau1761_alc_decay_time_text); +static SOC_ENUM_SINGLE_DECL(adau1761_alc_ng_type_enum, + ADAU1761_ALC_CTRL3, 6, adau1761_alc_ng_type_text); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -161,6 +270,22 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), + + SOC_SINGLE_TLV("ALC Capture Max Gain Volume", ADAU1761_ALC_CTRL0, + 3, 7, 0, adau1761_alc_max_gain_tlv), + SOC_ENUM("ALC Capture Function", adau1761_alc_function_enum), + SOC_ENUM("ALC Capture Hold Time", adau1761_alc_hold_time_enum), + SOC_SINGLE_TLV("ALC Capture Target Volume", ADAU1761_ALC_CTRL1, + 0, 15, 0, adau1761_alc_target_tlv), + SOC_ENUM("ALC Capture Attack Time", adau1761_alc_decay_time_enum), + SOC_ENUM("ALC Capture Decay Time", adau1761_alc_attack_time_enum), + SOC_ENUM("ALC Capture Noise Gate Type", adau1761_alc_ng_type_enum), + SOC_SINGLE("ALC Capture Noise Gate Switch", + ADAU1761_ALC_CTRL3, 5, 1, 0), + SOC_SINGLE_TLV("ALC Capture Noise Gate Threshold Volume", + ADAU1761_ALC_CTRL3, 0, 31, 0, adau1761_alc_ng_threshold_tlv), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -632,6 +757,10 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: + case ADAU1761_ALC_CTRL1: + case ADAU1761_ALC_CTRL2: + case ADAU1761_ALC_CTRL3: return true; default: break; diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 85beef265cc8..3c1bd24a1057 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -9,9 +9,9 @@ #include <linux/delay.h> #include <linux/device.h> #include <linux/kernel.h> -#include <linux/mfd/cros_ec.h> -#include <linux/mfd/cros_ec_commands.h> #include <linux/module.h> +#include <linux/platform_data/cros_ec_commands.h> +#include <linux/platform_data/cros_ec_proto.h> #include <linux/platform_device.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 9150e7068467..36eef1fb3d18 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -53,7 +53,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0); -static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv, + 0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0), + 8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0), +); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0), @@ -91,7 +94,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL, 4, 0, 3, 1, hpout_vol_tlv), SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL, - 0, 4, 7, 0, hpmixer_gain_tlv), + 4, 0, 11, 0, hpmixer_gain_tlv), SOC_ENUM("Playback Polarity", dacpol), SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, diff --git a/sound/soc/codecs/madera.h b/sound/soc/codecs/madera.h index 1f3e8e230cf2..6d8938a3fb64 100644 --- a/sound/soc/codecs/madera.h +++ b/sound/soc/codecs/madera.h @@ -27,6 +27,7 @@ #define MADERA_FLL_SRC_NONE -1 #define MADERA_FLL_SRC_MCLK1 0 #define MADERA_FLL_SRC_MCLK2 1 +#define MADERA_FLL_SRC_MCLK3 2 #define MADERA_FLL_SRC_SLIMCLK 3 #define MADERA_FLL_SRC_FLL1 4 #define MADERA_FLL_SRC_FLL2 5 @@ -51,6 +52,7 @@ #define MADERA_CLK_SRC_MCLK1 0x0 #define MADERA_CLK_SRC_MCLK2 0x1 +#define MADERA_CLK_SRC_MCLK3 0x2 #define MADERA_CLK_SRC_FLL1 0x4 #define MADERA_CLK_SRC_FLL2 0x5 #define MADERA_CLK_SRC_FLL3 0x6 diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index a92a0bacd812..be1e276e3631 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1628,14 +1628,18 @@ static int rt1011_hw_params(struct snd_pcm_substream *substream, static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); unsigned int reg_val = 0, reg_bclk_inv = 0; + int ret = 0; + snd_soc_dapm_mutex_lock(dapm); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: reg_val |= RT1011_I2S_TDM_MS_S; break; default: - return -EINVAL; + ret = -EINVAL; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -1645,7 +1649,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) reg_bclk_inv |= RT1011_TDM_INV_BCLK; break; default: - return -EINVAL; + ret = -EINVAL; } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -1661,7 +1665,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) reg_val |= RT1011_I2S_TDM_DF_PCM_B; break; default: - return -EINVAL; + ret = -EINVAL; } switch (dai->id) { @@ -1676,9 +1680,11 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); - return -EINVAL; + ret = -EINVAL; } - return 0; + + snd_soc_dapm_mutex_unlock(dapm); + return ret; } static int rt1011_set_component_sysclk(struct snd_soc_component *component, @@ -1797,8 +1803,12 @@ static int rt1011_set_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 snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); unsigned int val = 0, tdm_en = 0; + int ret = 0; + snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) tdm_en = RT1011_TDM_I2S_DOCK_EN_1; @@ -1818,7 +1828,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, case 2: break; default: - return -EINVAL; + ret = -EINVAL; } switch (slot_width) { @@ -1837,7 +1847,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, case 16: break; default: - return -EINVAL; + ret = -EINVAL; } snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, @@ -1854,7 +1864,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); - return 0; + snd_soc_dapm_mutex_unlock(dapm); + return ret; } static int rt1011_probe(struct snd_soc_component *component) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 2943692f66ed..e6c1ec6c426e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3644,7 +3644,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_PWR_ANLG_1, RT5663_LDO1_DVO_MASK | RT5663_AMP_HP_MASK, RT5663_LDO1_DVO_0_9V | RT5663_AMP_HP_3X); - break; + break; case CODEC_VER_0: regmap_update_bits(rt5663->regmap, RT5663_DIG_MISC, RT5663_DIG_GATE_CTRL_MASK, RT5663_DIG_GATE_CTRL_EN); @@ -3663,7 +3663,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_TDM_2, RT5663_DATA_SWAP_ADCDAT1_MASK, RT5663_DATA_SWAP_ADCDAT1_LL); - break; + break; default: dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); } diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c new file mode 100644 index 000000000000..9da88ccb1d51 --- /dev/null +++ b/sound/soc/codecs/tas2770.c @@ -0,0 +1,808 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2770 20-W Digital Input Mono Class-D +// Audio Amplifier with Speaker I/V Sense +// +// Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ +// Author: Tracy Yi <tracy-yi@ti.com> +// Frank Shi <shifu0704@thundersoft.com> + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/err.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/firmware.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "tas2770.h" + +#define TAS2770_MDELAY 0xFFFFFFFE + +static void tas2770_reset(struct tas2770_priv *tas2770) +{ + if (tas2770->reset_gpio) { + gpiod_set_value_cansleep(tas2770->reset_gpio, 0); + msleep(20); + gpiod_set_value_cansleep(tas2770->reset_gpio, 1); + } + snd_soc_component_write(tas2770->component, TAS2770_SW_RST, + TAS2770_RST); +} + +static int tas2770_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_component_update_bits(component, + TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_ACTIVE); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_component_update_bits(component, + TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_SHUTDOWN); + break; + + default: + dev_err(tas2770->dev, + "wrong power level setting %d\n", level); + return -EINVAL; + } + + return 0; +} + +#ifdef CONFIG_PM +static int tas2770_codec_suspend(struct snd_soc_component *component) +{ + int ret; + + ret = snd_soc_component_update_bits(component, + TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_SHUTDOWN); + if (ret) + return ret; + + return 0; +} + +static int tas2770_codec_resume(struct snd_soc_component *component) +{ + int ret; + + ret = snd_soc_component_update_bits(component, + TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_ACTIVE); + if (ret) + return -EINVAL; + + return 0; +} +#else +#define tas2770_codec_suspend NULL +#define tas2770_codec_resume NULL +#endif + +static const char * const tas2770_ASI1_src[] = { + "I2C offset", "Left", "Right", "LeftRightDiv2", +}; + +static SOC_ENUM_SINGLE_DECL( + tas2770_ASI1_src_enum, TAS2770_TDM_CFG_REG2, + 4, tas2770_ASI1_src); + +static const struct snd_kcontrol_new tas2770_asi1_mux = + SOC_DAPM_ENUM("ASI1 Source", tas2770_ASI1_src_enum); + +static int tas2770_dac_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 tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + ret = snd_soc_component_update_bits(component, + TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_MUTE); + if (ret) + goto end; + break; + case SND_SOC_DAPM_PRE_PMD: + ret = snd_soc_component_update_bits(component, + TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_SHUTDOWN); + if (ret) + goto end; + break; + default: + dev_err(tas2770->dev, "Not supported evevt\n"); + return -EINVAL; + } + +end: + return ret; +} + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 3, 1, 1); +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 2, 1, 1); + +static const struct snd_soc_dapm_widget tas2770_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, + &tas2770_asi1_mux), + SND_SOC_DAPM_SWITCH("ISENSE", TAS2770_PWR_CTRL, 3, 1, + &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", TAS2770_PWR_CTRL, 2, 1, + &vsense_switch), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2770_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON") +}; + +static const struct snd_soc_dapm_route tas2770_audio_map[] = { + {"ASI1 Sel", "I2C offset", "ASI1"}, + {"ASI1 Sel", "Left", "ASI1"}, + {"ASI1 Sel", "Right", "ASI1"}, + {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, + {"DAC", NULL, "ASI1 Sel"}, + {"OUT", NULL, "DAC"}, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + +static int tas2770_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + int ret; + + if (mute) + ret = snd_soc_component_update_bits(component, + TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_MUTE); + else + ret = snd_soc_component_update_bits(component, + TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_ACTIVE); + + return ret; +} + +static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) +{ + int ret; + struct snd_soc_component *component = tas2770->component; + + switch (bitwidth) { + case SNDRV_PCM_FORMAT_S16_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_16BITS); + tas2770->v_sense_slot = tas2770->i_sense_slot + 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_24BITS); + tas2770->v_sense_slot = tas2770->i_sense_slot + 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_32BITS); + tas2770->v_sense_slot = tas2770->i_sense_slot + 4; + break; + + default: + return -EINVAL; + } + + tas2770->channel_size = bitwidth; + + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG5, + TAS2770_TDM_CFG_REG5_VSNS_MASK | + TAS2770_TDM_CFG_REG5_50_MASK, + TAS2770_TDM_CFG_REG5_VSNS_ENABLE | + tas2770->v_sense_slot); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG6, + TAS2770_TDM_CFG_REG6_ISNS_MASK | + TAS2770_TDM_CFG_REG6_50_MASK, + TAS2770_TDM_CFG_REG6_ISNS_ENABLE | + tas2770->i_sense_slot); + +end: + return ret; +} + +static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) +{ + int ret; + struct snd_soc_component *component = tas2770->component; + + switch (samplerate) { + case 48000: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); + if (ret) + goto end; + break; + case 44100: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); + if (ret) + goto end; + break; + case 96000: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); + break; + case 88200: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); + break; + case 19200: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); + if (ret) + goto end; + break; + case 17640: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); + break; + default: + ret = -EINVAL; + } + +end: + if (!ret) + tas2770->sampling_rate = samplerate; + return ret; +} + +static int tas2770_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 tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + ret = tas2770_set_bitwidth(tas2770, params_format(params)); + if (ret < 0) + goto end; + + + ret = tas2770_set_samplerate(tas2770, params_rate(params)); + +end: + return ret; +} + +static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + int ret; + int value = 0; + struct snd_soc_component *component = dai->component; + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(tas2770->dev, "ASI format master is not found\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_RSING; + break; + case SND_SOC_DAIFMT_IB_NF: + asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_FALING; + break; + default: + dev_err(tas2770->dev, "ASI format Inverse is not found\n"); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, + TAS2770_TDM_CFG_REG1_RX_MASK, + asi_cfg_1); + if (ret) + return ret; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + tdm_rx_start_slot = 1; + break; + case SND_SOC_DAIFMT_DSP_A: + tdm_rx_start_slot = 0; + break; + case SND_SOC_DAIFMT_DSP_B: + tdm_rx_start_slot = 1; + break; + case SND_SOC_DAIFMT_LEFT_J: + tdm_rx_start_slot = 0; + break; + default: + dev_err(tas2770->dev, + "DAI Format is not found, fmt=0x%x\n", fmt); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, + TAS2770_TDM_CFG_REG1_MASK, + (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); + if (ret) + return ret; + + value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); + + tas2770->asi_format = fmt; + + return 0; +} + +static int tas2770_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 tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + int left_slot, right_slot; + int ret; + + if (tx_mask == 0 || rx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, + TAS2770_TDM_CFG_REG3_30_MASK, + (left_slot << TAS2770_TDM_CFG_REG3_30_SHIFT)); + if (ret) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, + TAS2770_TDM_CFG_REG3_RXS_MASK, + (right_slot << TAS2770_TDM_CFG_REG3_RXS_SHIFT)); + if (ret) + return ret; + + switch (slot_width) { + case 16: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_16BITS); + break; + + case 24: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_24BITS); + break; + + case 32: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_32BITS); + break; + + case 0: + /* Do not change slot width */ + ret = 0; + break; + + default: + ret = -EINVAL; + } + + if (!ret) + tas2770->slot_width = slot_width; + + return ret; +} + +static struct snd_soc_dai_ops tas2770_dai_ops = { + .digital_mute = tas2770_mute, + .hw_params = tas2770_hw_params, + .set_fmt = tas2770_set_fmt, + .set_tdm_slot = tas2770_set_dai_tdm_slot, +}; + +#define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +#define TAS2770_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_192000\ + ) + +static struct snd_soc_dai_driver tas2770_dai_driver[] = { + { + .name = "tas2770 ASI1", + .id = 0, + .playback = { + .stream_name = "ASI1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TAS2770_RATES, + .formats = TAS2770_FORMATS, + }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 0, + .channels_max = 2, + .rates = TAS2770_RATES, + .formats = TAS2770_FORMATS, + }, + .ops = &tas2770_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int tas2770_codec_probe(struct snd_soc_component *component) +{ + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + + tas2770->component = component; + + return 0; +} + +static DECLARE_TLV_DB_SCALE(tas2770_digital_tlv, 1100, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -12750, 50, 0); + +static const struct snd_kcontrol_new tas2770_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Playback Volume", TAS2770_PLAY_CFG_REG2, + 0, TAS2770_PLAY_CFG_REG2_VMAX, 1, + tas2770_playback_volume), + SOC_SINGLE_TLV("Amp Gain Volume", TAS2770_PLAY_CFG_REG0, + 0, 0x14, 0, + tas2770_digital_tlv), +}; + +static const struct snd_soc_component_driver soc_component_driver_tas2770 = { + .probe = tas2770_codec_probe, + .suspend = tas2770_codec_suspend, + .resume = tas2770_codec_resume, + .set_bias_level = tas2770_set_bias_level, + .controls = tas2770_snd_controls, + .num_controls = ARRAY_SIZE(tas2770_snd_controls), + .dapm_widgets = tas2770_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2770_dapm_widgets), + .dapm_routes = tas2770_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2770_audio_map), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int tas2770_register_codec(struct tas2770_priv *tas2770) +{ + return devm_snd_soc_register_component(tas2770->dev, + &soc_component_driver_tas2770, + tas2770_dai_driver, ARRAY_SIZE(tas2770_dai_driver)); +} + +static const struct reg_default tas2770_reg_defaults[] = { + { TAS2770_PAGE, 0x00 }, + { TAS2770_SW_RST, 0x00 }, + { TAS2770_PWR_CTRL, 0x0e }, + { TAS2770_PLAY_CFG_REG0, 0x10 }, + { TAS2770_PLAY_CFG_REG1, 0x01 }, + { TAS2770_PLAY_CFG_REG2, 0x00 }, + { TAS2770_MSC_CFG_REG0, 0x07 }, + { TAS2770_TDM_CFG_REG1, 0x02 }, + { TAS2770_TDM_CFG_REG2, 0x0a }, + { TAS2770_TDM_CFG_REG3, 0x10 }, + { TAS2770_INT_MASK_REG0, 0xfc }, + { TAS2770_INT_MASK_REG1, 0xb1 }, + { TAS2770_INT_CFG, 0x05 }, + { TAS2770_MISC_IRQ, 0x81 }, + { TAS2770_CLK_CGF, 0x0c }, + +}; + +static bool tas2770_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2770_PAGE: /* regmap implementation requires this */ + case TAS2770_SW_RST: /* always clears after write */ + case TAS2770_BO_PRV_REG0:/* has a self clearing bit */ + case TAS2770_LVE_INT_REG0: + case TAS2770_LVE_INT_REG1: + case TAS2770_LAT_INT_REG0:/* Sticky interrupt flags */ + case TAS2770_LAT_INT_REG1:/* Sticky interrupt flags */ + case TAS2770_VBAT_MSB: + case TAS2770_VBAT_LSB: + case TAS2770_TEMP_MSB: + case TAS2770_TEMP_LSB: + return true; + } + return false; +} + +static bool tas2770_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2770_LVE_INT_REG0: + case TAS2770_LVE_INT_REG1: + case TAS2770_LAT_INT_REG0: + case TAS2770_LAT_INT_REG1: + case TAS2770_VBAT_MSB: + case TAS2770_VBAT_LSB: + case TAS2770_TEMP_MSB: + case TAS2770_TEMP_LSB: + case TAS2770_TDM_CLK_DETC: + case TAS2770_REV_AND_GPID: + return false; + } + return true; +} + +static const struct regmap_range_cfg tas2770_regmap_ranges[] = { + { + .range_min = 0, + .range_max = 1 * 128, + .selector_reg = TAS2770_PAGE, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tas2770_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = tas2770_writeable, + .volatile_reg = tas2770_volatile, + .reg_defaults = tas2770_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2770_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = tas2770_regmap_ranges, + .num_ranges = ARRAY_SIZE(tas2770_regmap_ranges), + .max_register = 1 * 128, +}; + +static int tas2770_parse_dt(struct device *dev, struct tas2770_priv *tas2770) +{ + int rc = 0; + + rc = fwnode_property_read_u32(dev->fwnode, "ti,asi-format", + &tas2770->asi_format); + if (rc) { + dev_err(tas2770->dev, "Looking up %s property failed %d\n", + "ti,asi-format", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", + &tas2770->i_sense_slot); + if (rc) { + dev_err(tas2770->dev, "Looking up %s property failed %d\n", + "ti,imon-slot-no", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,vmon-slot-no", + &tas2770->v_sense_slot); + if (rc) { + dev_err(tas2770->dev, "Looking up %s property failed %d\n", + "ti,vmon-slot-no", rc); + goto end; + } + +end: + return rc; +} + +static int tas2770_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tas2770_priv *tas2770; + int result; + + tas2770 = devm_kzalloc(&client->dev, + sizeof(struct tas2770_priv), GFP_KERNEL); + if (!tas2770) + return -ENOMEM; + tas2770->dev = &client->dev; + + i2c_set_clientdata(client, tas2770); + dev_set_drvdata(&client->dev, tas2770); + tas2770->power_state = TAS2770_POWER_SHUTDOWN; + + tas2770->regmap = devm_regmap_init_i2c(client, &tas2770_i2c_regmap); + if (IS_ERR(tas2770->regmap)) { + result = PTR_ERR(tas2770->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + result); + goto end; + } + + if (client->dev.of_node) { + result = tas2770_parse_dt(&client->dev, tas2770); + if (result) { + dev_err(tas2770->dev, "%s: Failed to parse devicetree\n", + __func__); + goto end; + } + } + + tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, + "reset-gpio", + GPIOD_OUT_HIGH); + if (IS_ERR(tas2770->reset_gpio)) { + if (PTR_ERR(tas2770->reset_gpio) == -EPROBE_DEFER) { + tas2770->reset_gpio = NULL; + return -EPROBE_DEFER; + } + } + + tas2770->channel_size = 0; + tas2770->slot_width = 0; + + tas2770_reset(tas2770); + + result = tas2770_register_codec(tas2770); + if (result) + dev_err(tas2770->dev, "Register codec failed.\n"); + +end: + return result; +} + +static int tas2770_i2c_remove(struct i2c_client *client) +{ + pm_runtime_disable(&client->dev); + return 0; +} + + +static const struct i2c_device_id tas2770_i2c_id[] = { + { "tas2770", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2770_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id tas2770_of_match[] = { + { .compatible = "ti,tas2770" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2770_of_match); +#endif + +static struct i2c_driver tas2770_i2c_driver = { + .driver = { + .name = "tas2770", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2770_of_match), + }, + .probe = tas2770_i2c_probe, + .remove = tas2770_i2c_remove, + .id_table = tas2770_i2c_id, +}; + +module_i2c_driver(tas2770_i2c_driver); + +MODULE_AUTHOR("Shi Fu <shifu0704@thundersoft.com>"); +MODULE_DESCRIPTION("TAS2770 I2C Smart Amplifier driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tas2770.h b/sound/soc/codecs/tas2770.h new file mode 100644 index 000000000000..d597a8280707 --- /dev/null +++ b/sound/soc/codecs/tas2770.h @@ -0,0 +1,164 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC TAS2770 codec driver + * + * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ + */ +#ifndef __TAS2770__ +#define __TAS2770__ + +/* Book Control Register (available in page0 of each book) */ +#define TAS2770_BOOKCTL_PAGE 0 +#define TAS2770_BOOKCTL_REG 127 +#define TAS2770_REG(page, reg) ((page * 128) + reg) + /* Page */ +#define TAS2770_PAGE TAS2770_REG(0X0, 0x00) +#define TAS2770_PAGE_PAGE_MASK 255 + /* Software Reset */ +#define TAS2770_SW_RST TAS2770_REG(0X0, 0x01) +#define TAS2770_RST BIT(0) + /* Power Control */ +#define TAS2770_PWR_CTRL TAS2770_REG(0X0, 0x02) +#define TAS2770_PWR_CTRL_MASK 0x3 +#define TAS2770_PWR_CTRL_ACTIVE 0x0 +#define TAS2770_PWR_CTRL_MUTE BIT(0) +#define TAS2770_PWR_CTRL_SHUTDOWN 0x2 + /* Playback Configuration Reg0 */ +#define TAS2770_PLAY_CFG_REG0 TAS2770_REG(0X0, 0x03) + /* Playback Configuration Reg1 */ +#define TAS2770_PLAY_CFG_REG1 TAS2770_REG(0X0, 0x04) + /* Playback Configuration Reg2 */ +#define TAS2770_PLAY_CFG_REG2 TAS2770_REG(0X0, 0x05) +#define TAS2770_PLAY_CFG_REG2_VMAX 0xc9 + /* Misc Configuration Reg0 */ +#define TAS2770_MSC_CFG_REG0 TAS2770_REG(0X0, 0x07) + /* TDM Configuration Reg0 */ +#define TAS2770_TDM_CFG_REG0 TAS2770_REG(0X0, 0x0A) +#define TAS2770_TDM_CFG_REG0_SMP_MASK BIT(5) +#define TAS2770_TDM_CFG_REG0_SMP_48KHZ 0x0 +#define TAS2770_TDM_CFG_REG0_SMP_44_1KHZ BIT(5) +#define TAS2770_TDM_CFG_REG0_31_MASK 0xe +#define TAS2770_TDM_CFG_REG0_31_44_1_48KHZ 0x6 +#define TAS2770_TDM_CFG_REG0_31_88_2_96KHZ 0x8 +#define TAS2770_TDM_CFG_REG0_31_176_4_192KHZ 0xa + /* TDM Configuration Reg1 */ +#define TAS2770_TDM_CFG_REG1 TAS2770_REG(0X0, 0x0B) +#define TAS2770_TDM_CFG_REG1_MASK 0x3e +#define TAS2770_TDM_CFG_REG1_51_SHIFT 1 +#define TAS2770_TDM_CFG_REG1_RX_MASK BIT(0) +#define TAS2770_TDM_CFG_REG1_RX_RSING 0x0 +#define TAS2770_TDM_CFG_REG1_RX_FALING BIT(0) + /* TDM Configuration Reg2 */ +#define TAS2770_TDM_CFG_REG2 TAS2770_REG(0X0, 0x0C) +#define TAS2770_TDM_CFG_REG2_RXW_MASK 0xc +#define TAS2770_TDM_CFG_REG2_RXW_16BITS 0x0 +#define TAS2770_TDM_CFG_REG2_RXW_24BITS 0x8 +#define TAS2770_TDM_CFG_REG2_RXW_32BITS 0xc +#define TAS2770_TDM_CFG_REG2_RXS_MASK 0x3 +#define TAS2770_TDM_CFG_REG2_RXS_16BITS 0x0 +#define TAS2770_TDM_CFG_REG2_RXS_24BITS BIT(0) +#define TAS2770_TDM_CFG_REG2_RXS_32BITS 0x2 + /* TDM Configuration Reg3 */ +#define TAS2770_TDM_CFG_REG3 TAS2770_REG(0X0, 0x0D) +#define TAS2770_TDM_CFG_REG3_RXS_MASK 0xf0 +#define TAS2770_TDM_CFG_REG3_RXS_SHIFT 0x4 +#define TAS2770_TDM_CFG_REG3_30_MASK 0xf +#define TAS2770_TDM_CFG_REG3_30_SHIFT 0 + /* TDM Configuration Reg5 */ +#define TAS2770_TDM_CFG_REG5 TAS2770_REG(0X0, 0x0F) +#define TAS2770_TDM_CFG_REG5_VSNS_MASK BIT(6) +#define TAS2770_TDM_CFG_REG5_VSNS_ENABLE BIT(6) +#define TAS2770_TDM_CFG_REG5_50_MASK 0x3f + /* TDM Configuration Reg6 */ +#define TAS2770_TDM_CFG_REG6 TAS2770_REG(0X0, 0x10) +#define TAS2770_TDM_CFG_REG6_ISNS_MASK BIT(6) +#define TAS2770_TDM_CFG_REG6_ISNS_ENABLE BIT(6) +#define TAS2770_TDM_CFG_REG6_50_MASK 0x3f + /* Brown Out Prevention Reg0 */ +#define TAS2770_BO_PRV_REG0 TAS2770_REG(0X0, 0x1B) + /* Interrupt MASK Reg0 */ +#define TAS2770_INT_MASK_REG0 TAS2770_REG(0X0, 0x20) +#define TAS2770_INT_REG0_DEFAULT 0xfc +#define TAS2770_INT_MASK_REG0_DISABLE 0xff + /* Interrupt MASK Reg1 */ +#define TAS2770_INT_MASK_REG1 TAS2770_REG(0X0, 0x21) +#define TAS2770_INT_REG1_DEFAULT 0xb1 +#define TAS2770_INT_MASK_REG1_DISABLE 0xff + /* Live-Interrupt Reg0 */ +#define TAS2770_LVE_INT_REG0 TAS2770_REG(0X0, 0x22) + /* Live-Interrupt Reg1 */ +#define TAS2770_LVE_INT_REG1 TAS2770_REG(0X0, 0x23) + /* Latched-Interrupt Reg0 */ +#define TAS2770_LAT_INT_REG0 TAS2770_REG(0X0, 0x24) +#define TAS2770_LAT_INT_REG0_OCE_FLG BIT(1) +#define TAS2770_LAT_INT_REG0_OTE_FLG BIT(0) + /* Latched-Interrupt Reg1 */ +#define TAS2770_LAT_INT_REG1 TAS2770_REG(0X0, 0x25) +#define TAS2770_LAT_INT_REG1_VBA_TOV BIT(3) +#define TAS2770_LAT_INT_REG1_VBA_TUV BIT(2) +#define TAS2770_LAT_INT_REG1_BOUT_FLG BIT(1) + /* VBAT MSB */ +#define TAS2770_VBAT_MSB TAS2770_REG(0X0, 0x27) + /* VBAT LSB */ +#define TAS2770_VBAT_LSB TAS2770_REG(0X0, 0x28) + /* TEMP MSB */ +#define TAS2770_TEMP_MSB TAS2770_REG(0X0, 0x29) + /* TEMP LSB */ +#define TAS2770_TEMP_LSB TAS2770_REG(0X0, 0x2A) + /* Interrupt Configuration */ +#define TAS2770_INT_CFG TAS2770_REG(0X0, 0x30) + /* Misc IRQ */ +#define TAS2770_MISC_IRQ TAS2770_REG(0X0, 0x32) + /* Clock Configuration */ +#define TAS2770_CLK_CGF TAS2770_REG(0X0, 0x3C) + /* TDM Clock detection monitor */ +#define TAS2770_TDM_CLK_DETC TAS2770_REG(0X0, 0x77) + /* Revision and PG ID */ +#define TAS2770_REV_AND_GPID TAS2770_REG(0X0, 0x7D) + +#define TAS2770_POWER_ACTIVE 0 +#define TAS2770_POWER_MUTE 1 +#define TAS2770_POWER_SHUTDOWN 2 +#define ERROR_OVER_CURRENT 0x0000001 +#define ERROR_DIE_OVERTEMP 0x0000002 +#define ERROR_OVER_VOLTAGE 0x0000004 +#define ERROR_UNDER_VOLTAGE 0x0000008 +#define ERROR_BROWNOUT 0x0000010 +#define ERROR_CLASSD_PWR 0x0000020 +#define TAS2770_SLOT_16BIT 16 +#define TAS2770_SLOT_32BIT 32 +#define TAS2770_I2C_RETRY_COUNT 3 + +struct tas2770_register { + int book; + int page; + int reg; +}; + +struct tas2770_dai_cfg { + unsigned int dai_fmt; + unsigned int tdm_delay; +}; + +struct tas2770_priv { + struct device *dev; + struct regmap *regmap; + struct snd_soc_codec *codec; + struct snd_soc_component *component; + struct mutex dev_lock; + struct hrtimer mtimer; + int power_state; + int asi_format; + struct gpio_desc *reset_gpio; + int sampling_rate; + int frame_size; + int channel_size; + int slot_width; + int v_sense_slot; + int i_sense_slot; + bool runtime_suspend; + unsigned int err_code; + struct mutex codec_lock; +}; + +#endif /* __TAS2770__ */ diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f318403133e9..f11ffa28683b 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2837,11 +2837,11 @@ static int wcd9335_codec_enable_dec(struct snd_soc_dapm_widget *w, TX_HPF_CUT_OFF_FREQ_MASK) >> 5; snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x10); snd_soc_component_update_bits(comp, dec_cfg_reg, 0x08, 0x00); - if (hpf_coff_freq != CF_MIN_3DB_150HZ) { - snd_soc_component_update_bits(comp, dec_cfg_reg, - TX_HPF_CUT_OFF_FREQ_MASK, - hpf_coff_freq << 5); - } + if (hpf_coff_freq != CF_MIN_3DB_150HZ) { + snd_soc_component_update_bits(comp, dec_cfg_reg, + TX_HPF_CUT_OFF_FREQ_MASK, + hpf_coff_freq << 5); + } break; case SND_SOC_DAPM_POST_PMD: snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x00); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d5fb7f5dd551..15ce64a48a87 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -167,12 +167,12 @@ static int configure_aif_clock(struct snd_soc_component *component, int aif) switch (wm8994->sysclk[aif]) { case WM8994_SYSCLK_MCLK1: - rate = wm8994->mclk[0]; + rate = wm8994->mclk_rate[0]; break; case WM8994_SYSCLK_MCLK2: reg1 |= 0x8; - rate = wm8994->mclk[1]; + rate = wm8994->mclk_rate[1]; break; case WM8994_SYSCLK_FLL1: @@ -1038,6 +1038,45 @@ static bool wm8994_check_class_w_digital(struct snd_soc_component *component) return true; } +static int aif_mclk_set(struct snd_soc_component *component, int aif, bool enable) +{ + struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); + unsigned int offset, val, clk_idx; + int ret; + + if (aif) + offset = 4; + else + offset = 0; + + val = snd_soc_component_read32(component, WM8994_AIF1_CLOCKING_1 + offset); + val &= WM8994_AIF1CLK_SRC_MASK; + + switch (val) { + case 0: + clk_idx = WM8994_MCLK1; + break; + case 1: + clk_idx = WM8994_MCLK2; + break; + default: + return 0; + } + + if (enable) { + ret = clk_prepare_enable(wm8994->mclk[clk_idx].clk); + if (ret < 0) { + dev_err(component->dev, "Failed to enable MCLK%d\n", + clk_idx); + return ret; + } + } else { + clk_disable_unprepare(wm8994->mclk[clk_idx].clk); + } + + return 0; +} + static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1045,7 +1084,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; - int i; + int ret, i; int dac; int adc; int val; @@ -1061,6 +1100,10 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: + ret = aif_mclk_set(component, 0, true); + if (ret < 0) + return ret; + /* Don't enable timeslot 2 if not in use */ if (wm8994->channels[0] <= 2) mask &= ~(WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA); @@ -1133,6 +1176,12 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, break; } + switch (event) { + case SND_SOC_DAPM_POST_PMD: + aif_mclk_set(component, 0, false); + break; + } + return 0; } @@ -1140,13 +1189,17 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - int i; + int ret, i; int dac; int adc; int val; switch (event) { case SND_SOC_DAPM_PRE_PMU: + ret = aif_mclk_set(component, 1, true); + if (ret < 0) + return ret; + val = snd_soc_component_read32(component, WM8994_AIF2_CONTROL_1); if ((val & WM8994_AIF2ADCL_SRC) && (val & WM8994_AIF2ADCR_SRC)) @@ -1218,6 +1271,12 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, break; } + switch (event) { + case SND_SOC_DAPM_POST_PMD: + aif_mclk_set(component, 1, false); + break; + } + return 0; } @@ -1623,10 +1682,10 @@ SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, aif1clk_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, aif2clk_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), @@ -2141,6 +2200,7 @@ static int _wm8994_set_fll(struct snd_soc_component *component, int id, int src, u16 reg, clk1, aif_reg, aif_src; unsigned long timeout; bool was_enabled; + struct clk *mclk; switch (id) { case WM8994_FLL1: @@ -2216,6 +2276,27 @@ static int _wm8994_set_fll(struct snd_soc_component *component, int id, int src, snd_soc_component_update_bits(component, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA, 0); + /* Disable MCLK if needed before we possibly change to new clock parent */ + if (was_enabled) { + reg = snd_soc_component_read32(component, WM8994_FLL1_CONTROL_5 + + reg_offset); + reg = ((reg & WM8994_FLL1_REFCLK_SRC_MASK) + >> WM8994_FLL1_REFCLK_SRC_SHIFT) + 1; + + switch (reg) { + case WM8994_FLL_SRC_MCLK1: + mclk = wm8994->mclk[WM8994_MCLK1].clk; + break; + case WM8994_FLL_SRC_MCLK2: + mclk = wm8994->mclk[WM8994_MCLK2].clk; + break; + default: + mclk = NULL; + } + + clk_disable_unprepare(mclk); + } + if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK && freq_in == freq_out && freq_out) { dev_dbg(component->dev, "Bypassing FLL%d\n", id + 1); @@ -2260,10 +2341,29 @@ static int _wm8994_set_fll(struct snd_soc_component *component, int id, int src, /* Clear any pending completion from a previous failure */ try_wait_for_completion(&wm8994->fll_locked[id]); + switch (src) { + case WM8994_FLL_SRC_MCLK1: + mclk = wm8994->mclk[WM8994_MCLK1].clk; + break; + case WM8994_FLL_SRC_MCLK2: + mclk = wm8994->mclk[WM8994_MCLK2].clk; + break; + default: + mclk = NULL; + } + /* Enable (with fractional mode if required) */ if (freq_out) { + ret = clk_prepare_enable(mclk); + if (ret < 0) { + dev_err(component->dev, "Failed to enable MCLK for FLL%d\n", + id + 1); + return ret; + } + /* Enable VMID if we need it */ if (!was_enabled) { + active_reference(component); switch (control->type) { @@ -2372,12 +2472,29 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src, return _wm8994_set_fll(dai->component, id, src, freq_in, freq_out); } +static int wm8994_set_mclk_rate(struct wm8994_priv *wm8994, unsigned int id, + unsigned int *freq) +{ + int ret; + + if (!wm8994->mclk[id].clk || *freq == wm8994->mclk_rate[id]) + return 0; + + ret = clk_set_rate(wm8994->mclk[id].clk, *freq); + if (ret < 0) + return ret; + + *freq = clk_get_rate(wm8994->mclk[id].clk); + + return 0; +} + static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_component *component = dai->component; struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); - int i; + int ret, i; switch (dai->id) { case 1: @@ -2392,7 +2509,12 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, switch (clk_id) { case WM8994_SYSCLK_MCLK1: wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_MCLK1; - wm8994->mclk[0] = freq; + + ret = wm8994_set_mclk_rate(wm8994, dai->id - 1, &freq); + if (ret < 0) + return ret; + + wm8994->mclk_rate[0] = freq; dev_dbg(dai->dev, "AIF%d using MCLK1 at %uHz\n", dai->id, freq); break; @@ -2400,7 +2522,12 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, case WM8994_SYSCLK_MCLK2: /* TODO: Set GPIO AF */ wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_MCLK2; - wm8994->mclk[1] = freq; + + ret = wm8994_set_mclk_rate(wm8994, dai->id - 1, &freq); + if (ret < 0) + return ret; + + wm8994->mclk_rate[1] = freq; dev_dbg(dai->dev, "AIF%d using MCLK2 at %uHz\n", dai->id, freq); break; @@ -4456,6 +4583,7 @@ static const struct snd_soc_component_driver soc_component_dev_wm8994 = { static int wm8994_probe(struct platform_device *pdev) { struct wm8994_priv *wm8994; + int ret; wm8994 = devm_kzalloc(&pdev->dev, sizeof(struct wm8994_priv), GFP_KERNEL); @@ -4467,6 +4595,16 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + wm8994->mclk[WM8994_MCLK1].id = "MCLK1"; + wm8994->mclk[WM8994_MCLK2].id = "MCLK2"; + + ret = devm_clk_bulk_get_optional(pdev->dev.parent, ARRAY_SIZE(wm8994->mclk), + wm8994->mclk); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to get clocks: %d\n", ret); + return ret; + } + pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 1d6f2abe1c11..41c4b126114d 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -6,6 +6,7 @@ #ifndef _WM8994_H #define _WM8994_H +#include <linux/clk.h> #include <sound/soc.h> #include <linux/firmware.h> #include <linux/completion.h> @@ -14,6 +15,12 @@ #include "wm_hubs.h" +enum { + WM8994_MCLK1, + WM8994_MCLK2, + WM8994_NUM_MCLK +}; + /* Sources for AIF1/2 SYSCLK - use with set_dai_sysclk() */ #define WM8994_SYSCLK_MCLK1 1 #define WM8994_SYSCLK_MCLK2 2 @@ -73,9 +80,10 @@ struct wm8994; struct wm8994_priv { struct wm_hubs_data hubs; struct wm8994 *wm8994; + struct clk_bulk_data mclk[WM8994_NUM_MCLK]; int sysclk[2]; int sysclk_rate[2]; - int mclk[2]; + int mclk_rate[2]; int aifclk[2]; int aifdiv[2]; int channels[2]; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa99c008a925..65e8cd4be930 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -25,6 +25,16 @@ 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_MQS + tristate "Medium Quality Sound (MQS) module support" + depends on SND_SOC_FSL_SAI + select REGMAP_MMIO + help + Say Y if you want to add Medium Quality Sound (MQS) + support for the Freescale CPUs. + 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 diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index c0dd04422fe9..8cde88c72d93 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -23,6 +23,7 @@ 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 +snd-soc-fsl-mqs-objs := fsl_mqs.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 @@ -33,6 +34,7 @@ obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o obj-$(CONFIG_SND_SOC_FSL_ESAI) += snd-soc-fsl-esai.o obj-$(CONFIG_SND_SOC_FSL_MICFIL) += snd-soc-fsl-micfil.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o +obj-$(CONFIG_SND_SOC_FSL_MQS) += snd-soc-fsl-mqs.o obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o # MPC5200 Platform Support diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index cfa40ef6b1ca..0bf91a6f54b9 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -115,7 +115,7 @@ static void fsl_asrc_sel_proc(int inrate, int outrate, * within range [ANCA, ANCA+ANCB-1], depends on the channels of pair A * while pair A and pair C are comparatively independent. */ -static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) +int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) { enum asrc_pair_index index = ASRC_INVALID_PAIR; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -158,7 +158,7 @@ static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) * * It clears the resource from asrc_priv and releases the occupied channels. */ -static void fsl_asrc_release_pair(struct fsl_asrc_pair *pair) +void fsl_asrc_release_pair(struct fsl_asrc_pair *pair) { struct fsl_asrc *asrc_priv = pair->asrc_priv; enum asrc_pair_index index = pair->index; @@ -265,6 +265,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; enum asrc_pair_index index = pair->index; + enum asrc_word_width input_word_width; + enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; u32 clk_index[2], div[2]; int in, out, channels; @@ -283,9 +285,32 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; } - /* Validate output width */ - if (config->output_word_width == ASRC_WIDTH_8_BIT) { - pair_err("does not support 8bit width output\n"); + switch (snd_pcm_format_width(config->input_format)) { + case 8: + input_word_width = ASRC_WIDTH_8_BIT; + break; + case 16: + input_word_width = ASRC_WIDTH_16_BIT; + break; + case 24: + input_word_width = ASRC_WIDTH_24_BIT; + break; + default: + pair_err("does not support this input format, %d\n", + config->input_format); + return -EINVAL; + } + + switch (snd_pcm_format_width(config->output_format)) { + case 16: + output_word_width = ASRC_WIDTH_16_BIT; + break; + case 24: + output_word_width = ASRC_WIDTH_24_BIT; + break; + default: + pair_err("does not support this output format, %d\n", + config->output_format); return -EINVAL; } @@ -383,8 +408,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* Implement word_width configurations */ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR1(index), ASRMCR1i_OW16_MASK | ASRMCR1i_IWD_MASK, - ASRMCR1i_OW16(config->output_word_width) | - ASRMCR1i_IWD(config->input_word_width)); + ASRMCR1i_OW16(output_word_width) | + ASRMCR1i_IWD(input_word_width)); /* Enable BUFFER STALL */ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index), @@ -497,13 +522,13 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); - int width = params_width(params); struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; unsigned int channels = params_channels(params); unsigned int rate = params_rate(params); struct asrc_config config; - int word_width, ret; + snd_pcm_format_t format; + int ret; ret = fsl_asrc_request_pair(channels, pair); if (ret) { @@ -513,15 +538,10 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, pair->config = &config; - if (width == 16) - width = ASRC_WIDTH_16_BIT; - else - width = ASRC_WIDTH_24_BIT; - if (asrc_priv->asrc_width == 16) - word_width = ASRC_WIDTH_16_BIT; + format = SNDRV_PCM_FORMAT_S16_LE; else - word_width = ASRC_WIDTH_24_BIT; + format = SNDRV_PCM_FORMAT_S24_LE; config.pair = pair->index; config.channel_num = channels; @@ -529,13 +549,13 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.outclk = OUTCLK_ASRCK1_CLK; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - config.input_word_width = width; - config.output_word_width = word_width; + config.input_format = params_format(params); + config.output_format = format; config.input_sample_rate = rate; config.output_sample_rate = asrc_priv->asrc_rate; } else { - config.input_word_width = word_width; - config.output_word_width = width; + config.input_format = format; + config.output_format = params_format(params); config.input_sample_rate = asrc_priv->asrc_rate; config.output_sample_rate = rate; } @@ -604,7 +624,7 @@ static int fsl_asrc_dai_probe(struct snd_soc_dai *dai) #define FSL_ASRC_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S20_3LE) + SNDRV_PCM_FMTBIT_S24_3LE) static struct snd_soc_dai_driver fsl_asrc_dai = { .probe = fsl_asrc_dai_probe, @@ -615,7 +635,8 @@ static struct snd_soc_dai_driver fsl_asrc_dai = { .rate_min = 5512, .rate_max = 192000, .rates = SNDRV_PCM_RATE_KNOT, - .formats = FSL_ASRC_FORMATS, + .formats = FSL_ASRC_FORMATS | + SNDRV_PCM_FMTBIT_S8, }, .capture = { .stream_name = "ASRC-Capture", diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index c60075112570..2b57e8c53728 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -342,8 +342,8 @@ struct asrc_config { unsigned int dma_buffer_size; unsigned int input_sample_rate; unsigned int output_sample_rate; - enum asrc_word_width input_word_width; - enum asrc_word_width output_word_width; + snd_pcm_format_t input_format; + snd_pcm_format_t output_format; enum asrc_inclk inclk; enum asrc_outclk outclk; }; @@ -462,4 +462,7 @@ struct fsl_asrc { #define DRV_NAME "fsl-asrc-dai" extern struct snd_soc_component_driver fsl_asrc_component; struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir); +int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair); +void fsl_asrc_release_pair(struct fsl_asrc_pair *pair); + #endif /* _FSL_ASRC_H */ diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 01052a0808b0..2a60fc6142b1 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -16,13 +16,11 @@ #define FSL_ASRC_DMABUF_SIZE (256 * 1024) -static const struct snd_pcm_hardware snd_imx_hardware = { +static struct snd_pcm_hardware snd_imx_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP_VALID, .buffer_bytes_max = FSL_ASRC_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 65535, /* Limited by SDMA engine */ @@ -270,12 +268,25 @@ static int fsl_asrc_dma_hw_free(struct snd_pcm_substream *substream) static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) { + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; 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_dmaengine_dai_dma_data *dma_data; struct device *dev = component->dev; struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); struct fsl_asrc_pair *pair; + struct dma_chan *tmp_chan = NULL; + u8 dir = tx ? OUT : IN; + bool release_pair = true; + int ret = 0; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(dev, "failed to set pcm hw params periods\n"); + return ret; + } pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL); if (!pair) @@ -285,11 +296,50 @@ static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) runtime->private_data = pair; - snd_pcm_hw_constraint_integer(substream->runtime, - SNDRV_PCM_HW_PARAM_PERIODS); + /* Request a dummy pair, which will be released later. + * Request pair function needs channel num as input, for this + * dummy pair, we just request "1" channel temporarily. + */ + ret = fsl_asrc_request_pair(1, pair); + if (ret < 0) { + dev_err(dev, "failed to request asrc pair\n"); + goto req_pair_err; + } + + /* Request a dummy dma channel, which will be released later. */ + tmp_chan = fsl_asrc_get_dma_channel(pair, dir); + if (!tmp_chan) { + dev_err(dev, "failed to get dma channel\n"); + ret = -EINVAL; + goto dma_chan_err; + } + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + /* Refine the snd_imx_hardware according to caps of DMA. */ + ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream, + dma_data, + &snd_imx_hardware, + tmp_chan); + if (ret < 0) { + dev_err(dev, "failed to refine runtime hwparams\n"); + goto out; + } + + release_pair = false; snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); - return 0; +out: + dma_release_channel(tmp_chan); + +dma_chan_err: + fsl_asrc_release_pair(pair); + +req_pair_err: + if (release_pair) + kfree(pair); + + return ret; } static int fsl_asrc_dma_shutdown(struct snd_pcm_substream *substream) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c new file mode 100644 index 000000000000..c1619a553514 --- /dev/null +++ b/sound/soc/fsl/fsl_mqs.c @@ -0,0 +1,333 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC IMX MQS driver +// +// Copyright (C) 2014-2015 Freescale Semiconductor, Inc. +// Copyright 2019 NXP + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/mfd/syscon.h> +#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h> +#include <linux/pm_runtime.h> +#include <linux/of.h> +#include <linux/pm.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/initval.h> + +#define REG_MQS_CTRL 0x00 + +#define MQS_EN_MASK (0x1 << 28) +#define MQS_EN_SHIFT (28) +#define MQS_SW_RST_MASK (0x1 << 24) +#define MQS_SW_RST_SHIFT (24) +#define MQS_OVERSAMPLE_MASK (0x1 << 20) +#define MQS_OVERSAMPLE_SHIFT (20) +#define MQS_CLK_DIV_MASK (0xFF << 0) +#define MQS_CLK_DIV_SHIFT (0) + +/* codec private data */ +struct fsl_mqs { + struct regmap *regmap; + struct clk *mclk; + struct clk *ipg; + + unsigned int reg_iomuxc_gpr2; + unsigned int reg_mqs_ctrl; + bool use_gpr; +}; + +#define FSL_MQS_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define FSL_MQS_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static int fsl_mqs_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 fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); + unsigned long mclk_rate; + int div, res; + int bclk, lrclk; + + mclk_rate = clk_get_rate(mqs_priv->mclk); + bclk = snd_soc_params_to_bclk(params); + lrclk = params_rate(params); + + /* + * mclk_rate / (oversample(32,64) * FS * 2 * divider ) = repeat_rate; + * if repeat_rate is 8, mqs can achieve better quality. + * oversample rate is fix to 32 currently. + */ + div = mclk_rate / (32 * lrclk * 2 * 8); + res = mclk_rate % (32 * lrclk * 2 * 8); + + if (res == 0 && div > 0 && div <= 256) { + if (mqs_priv->use_gpr) { + regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, + IMX6SX_GPR2_MQS_CLK_DIV_MASK, + (div - 1) << IMX6SX_GPR2_MQS_CLK_DIV_SHIFT); + regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, + IMX6SX_GPR2_MQS_OVERSAMPLE_MASK, 0); + } else { + regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, + MQS_CLK_DIV_MASK, + (div - 1) << MQS_CLK_DIV_SHIFT); + regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, + MQS_OVERSAMPLE_MASK, 0); + } + } else { + dev_err(component->dev, "can't get proper divider\n"); + } + + return 0; +} + +static int fsl_mqs_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + /* Only LEFT_J & SLAVE mode is supported. */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int fsl_mqs_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); + + if (mqs_priv->use_gpr) + regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, + IMX6SX_GPR2_MQS_EN_MASK, + 1 << IMX6SX_GPR2_MQS_EN_SHIFT); + else + regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, + MQS_EN_MASK, + 1 << MQS_EN_SHIFT); + return 0; +} + +static void fsl_mqs_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); + + if (mqs_priv->use_gpr) + regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, + IMX6SX_GPR2_MQS_EN_MASK, 0); + else + regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, + MQS_EN_MASK, 0); +} + +const static struct snd_soc_component_driver soc_codec_fsl_mqs = { + .idle_bias_on = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct snd_soc_dai_ops fsl_mqs_dai_ops = { + .startup = fsl_mqs_startup, + .shutdown = fsl_mqs_shutdown, + .hw_params = fsl_mqs_hw_params, + .set_fmt = fsl_mqs_set_dai_fmt, +}; + +static struct snd_soc_dai_driver fsl_mqs_dai = { + .name = "fsl-mqs-dai", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = FSL_MQS_RATES, + .formats = FSL_MQS_FORMATS, + }, + .ops = &fsl_mqs_dai_ops, +}; + +static const struct regmap_config fsl_mqs_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = REG_MQS_CTRL, + .cache_type = REGCACHE_NONE, +}; + +static int fsl_mqs_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *gpr_np = 0; + struct fsl_mqs *mqs_priv; + void __iomem *regs; + int ret = 0; + + mqs_priv = devm_kzalloc(&pdev->dev, sizeof(*mqs_priv), GFP_KERNEL); + if (!mqs_priv) + return -ENOMEM; + + /* On i.MX6sx the MQS control register is in GPR domain + * But in i.MX8QM/i.MX8QXP the control register is moved + * to its own domain. + */ + if (of_device_is_compatible(np, "fsl,imx8qm-mqs")) + mqs_priv->use_gpr = false; + else + mqs_priv->use_gpr = true; + + if (mqs_priv->use_gpr) { + gpr_np = of_parse_phandle(np, "gpr", 0); + if (IS_ERR(gpr_np)) { + dev_err(&pdev->dev, "failed to get gpr node by phandle\n"); + ret = PTR_ERR(gpr_np); + goto out; + } + + mqs_priv->regmap = syscon_node_to_regmap(gpr_np); + if (IS_ERR(mqs_priv->regmap)) { + dev_err(&pdev->dev, "failed to get gpr regmap\n"); + ret = PTR_ERR(mqs_priv->regmap); + goto out; + } + } else { + regs = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + mqs_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "core", + regs, + &fsl_mqs_regmap_config); + if (IS_ERR(mqs_priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap: %ld\n", + PTR_ERR(mqs_priv->regmap)); + return PTR_ERR(mqs_priv->regmap); + } + + mqs_priv->ipg = devm_clk_get(&pdev->dev, "core"); + if (IS_ERR(mqs_priv->ipg)) { + dev_err(&pdev->dev, "failed to get the clock: %ld\n", + PTR_ERR(mqs_priv->ipg)); + goto out; + } + } + + mqs_priv->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(mqs_priv->mclk)) { + dev_err(&pdev->dev, "failed to get the clock: %ld\n", + PTR_ERR(mqs_priv->mclk)); + goto out; + } + + dev_set_drvdata(&pdev->dev, mqs_priv); + pm_runtime_enable(&pdev->dev); + + return devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, + &fsl_mqs_dai, 1); +out: + if (!IS_ERR(gpr_np)) + of_node_put(gpr_np); + + return ret; +} + +static int fsl_mqs_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + +#ifdef CONFIG_PM +static int fsl_mqs_runtime_resume(struct device *dev) +{ + struct fsl_mqs *mqs_priv = dev_get_drvdata(dev); + + if (mqs_priv->ipg) + clk_prepare_enable(mqs_priv->ipg); + + if (mqs_priv->mclk) + clk_prepare_enable(mqs_priv->mclk); + + if (mqs_priv->use_gpr) + regmap_write(mqs_priv->regmap, IOMUXC_GPR2, + mqs_priv->reg_iomuxc_gpr2); + else + regmap_write(mqs_priv->regmap, REG_MQS_CTRL, + mqs_priv->reg_mqs_ctrl); + return 0; +} + +static int fsl_mqs_runtime_suspend(struct device *dev) +{ + struct fsl_mqs *mqs_priv = dev_get_drvdata(dev); + + if (mqs_priv->use_gpr) + regmap_read(mqs_priv->regmap, IOMUXC_GPR2, + &mqs_priv->reg_iomuxc_gpr2); + else + regmap_read(mqs_priv->regmap, REG_MQS_CTRL, + &mqs_priv->reg_mqs_ctrl); + + if (mqs_priv->mclk) + clk_disable_unprepare(mqs_priv->mclk); + + if (mqs_priv->ipg) + clk_disable_unprepare(mqs_priv->ipg); + + return 0; +} +#endif + +static const struct dev_pm_ops fsl_mqs_pm_ops = { + SET_RUNTIME_PM_OPS(fsl_mqs_runtime_suspend, + fsl_mqs_runtime_resume, + NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +static const struct of_device_id fsl_mqs_dt_ids[] = { + { .compatible = "fsl,imx8qm-mqs", }, + { .compatible = "fsl,imx6sx-mqs", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_mqs_dt_ids); + +static struct platform_driver fsl_mqs_driver = { + .probe = fsl_mqs_probe, + .remove = fsl_mqs_remove, + .driver = { + .name = "fsl-mqs", + .of_match_table = fsl_mqs_dt_ids, + .pm = &fsl_mqs_pm_ops, + }, +}; + +module_platform_driver(fsl_mqs_driver); + +MODULE_AUTHOR("Shengjiu Wang <Shengjiu.Wang@nxp.com>"); +MODULE_DESCRIPTION("MQS codec driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform: fsl-mqs"); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b0a6fead1a6a..537dc69256f0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -799,15 +799,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, u32 wl = SSI_SxCCR_WL(sample_size); int ret; - /* - * SSI is properly configured if it is enabled and running in - * the synchronous mode; Note that AC97 mode is an exception - * that should set separate configurations for STCCR and SRCCR - * despite running in the synchronous mode. - */ - if (ssi->streams && ssi->synchronous) - return 0; - if (fsl_ssi_is_i2s_master(ssi)) { ret = fsl_ssi_set_bclk(substream, dai, hw_params); if (ret) @@ -823,6 +814,15 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + /* + * SSI is properly configured if it is enabled and running in + * the synchronous mode; Note that AC97 mode is an exception + * that should set separate configurations for STCCR and SRCCR + * despite running in the synchronous mode. + */ + if (ssi->streams && ssi->synchronous) + return 0; + if (!fsl_ssi_is_ac97(ssi)) { /* * Keep the ssi->i2s_net intact while having a local variable diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 9cbc982d46a9..54f2ee3010ee 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -193,6 +193,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: pdata->restore_stream = false; + /* fallthrough */ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index 54ac2fd41925..67f06c95eec5 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -6,6 +6,7 @@ #include <linux/acpi.h> #include <linux/device.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index eaf3e2208a06..70bb86f3342f 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -12,6 +12,7 @@ */ #include <linux/dmi.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 8879c3be29d5..c68a5b85a4a0 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -48,6 +48,7 @@ struct cht_mc_private { #define CHT_RT5645_SSP2_AIF2 BIT(16) /* default is using AIF1 */ #define CHT_RT5645_SSP0_AIF1 BIT(17) #define CHT_RT5645_SSP0_AIF2 BIT(18) +#define CHT_RT5645_PMC_PLT_CLK_0 BIT(19) static unsigned long cht_rt5645_quirk = 0; @@ -59,6 +60,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF1 enabled"); if (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2) dev_info(dev, "quirk SSP0_AIF2 enabled"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + dev_info(dev, "quirk PMC_PLT_CLK_0 enabled"); } static int platform_clock_control(struct snd_soc_dapm_widget *w, @@ -226,16 +229,22 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, return 0; } -/* uncomment when we have a real quirk static int cht_rt5645_quirk_cb(const struct dmi_system_id *id) { cht_rt5645_quirk = (unsigned long)id->driver_data; return 1; } -*/ static const struct dmi_system_id cht_rt5645_quirk_table[] = { { + /* Strago family Chromebooks */ + .callback = cht_rt5645_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_Strago"), + }, + .driver_data = (void *)CHT_RT5645_PMC_PLT_CLK_0, + }, + { }, }; @@ -526,6 +535,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) int dai_index = 0; int ret_val = 0; int i; + const char *mclk_name; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -662,11 +672,15 @@ static int snd_cht_mc_probe(struct platform_device *pdev) if (ret_val) return ret_val; - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { - dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + dev_err(&pdev->dev, "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 4977b5a65eb8..9d657421730a 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -8,6 +8,7 @@ * Mengdong Lin <mengdong.lin@intel.com> */ +#include <linux/gpio/consumer.h> #include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index e6357d306cb8..863a477d3405 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -36,7 +36,7 @@ SOC_INTEL_IS_CPU(byt, INTEL_FAM6_ATOM_SILVERMONT); SOC_INTEL_IS_CPU(cht, INTEL_FAM6_ATOM_AIRMONT); SOC_INTEL_IS_CPU(apl, INTEL_FAM6_ATOM_GOLDMONT); SOC_INTEL_IS_CPU(glk, INTEL_FAM6_ATOM_GOLDMONT_PLUS); -SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); +SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_L); static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) { diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index 1186a03a88d6..6068bb697e22 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -223,6 +223,8 @@ struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc, if (ipc->ops.reply_msg_match != NULL) header = ipc->ops.reply_msg_match(header, &mask); + else + mask = (u64)-1; if (list_empty(&ipc->rx_list)) { dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n", diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c index 212370bf704c..3466675f2678 100644 --- a/sound/soc/intel/skylake/skl-debug.c +++ b/sound/soc/intel/skylake/skl-debug.c @@ -188,7 +188,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf, memset(d->fw_read_buff, 0, FW_REG_BUF); if (w0_stat_sz > 0) - __iowrite32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2); + __ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2); for (offset = 0; offset < FW_REG_SIZE; offset += 16) { ret += snprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset); diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index ab3d23c7bd65..19f328d71f24 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -136,7 +136,7 @@ int skl_nhlt_update_topology_bin(struct skl_dev *skl) struct hdac_bus *bus = skl_to_bus(skl); struct device *dev = bus->dev; - dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n", + dev_dbg(dev, "oem_id %.6s, oem_table_id %.8s oem_revision %d\n", nhlt->header.oem_id, nhlt->header.oem_table_id, nhlt->header.oem_revision); diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig index 6b757168693e..e72f826062e9 100644 --- a/sound/soc/jz4740/Kconfig +++ b/sound/soc/jz4740/Kconfig @@ -1,30 +1,9 @@ # SPDX-License-Identifier: GPL-2.0-only -config SND_JZ4740_SOC - tristate "SoC Audio for Ingenic JZ4740 SoC" - 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 - the JZ4740 I2S interface. You will also need to select the audio - interfaces to support below. - -if SND_JZ4740_SOC - config SND_JZ4740_SOC_I2S tristate "SoC Audio (I2S protocol) for Ingenic JZ4740 SoC" + depends on MIPS || COMPILE_TEST depends on HAS_IOMEM + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y if you want to use I2S protocol and I2S codec on Ingenic JZ4740 based boards. - -config SND_JZ4740_SOC_QI_LB60 - tristate "SoC Audio support for Qi LB60" - depends on HAS_IOMEM - depends on JZ4740_QI_LB60 || COMPILE_TEST - select SND_JZ4740_SOC_I2S - select SND_SOC_JZ4740_CODEC - help - Say Y if you want to add support for ASoC audio on the Qi LB60 board - a.k.a Qi Ben NanoNote. - -endif diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile index fb10e9ad9ff7..f8701c9b09fe 100644 --- a/sound/soc/jz4740/Makefile +++ b/sound/soc/jz4740/Makefile @@ -5,8 +5,3 @@ snd-soc-jz4740-i2s-objs := jz4740-i2s.o obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o - -# Jz4740 Machine Support -snd-soc-qi-lb60-objs := qi_lb60.o - -obj-$(CONFIG_SND_JZ4740_SOC_QI_LB60) += snd-soc-qi-lb60.o diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c deleted file mode 100644 index 8ef6f41dcfbe..000000000000 --- a/sound/soc/jz4740/qi_lb60.c +++ /dev/null @@ -1,106 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de> - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <linux/gpio/consumer.h> - -struct qi_lb60 { - struct gpio_desc *snd_gpio; - struct gpio_desc *amp_gpio; -}; - -static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *ctrl, int event) -{ - struct qi_lb60 *qi_lb60 = snd_soc_card_get_drvdata(widget->dapm->card); - int on = !SND_SOC_DAPM_EVENT_OFF(event); - - gpiod_set_value_cansleep(qi_lb60->snd_gpio, on); - gpiod_set_value_cansleep(qi_lb60->amp_gpio, on); - - return 0; -} - -static const struct snd_soc_dapm_widget qi_lb60_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", qi_lb60_spk_event), - SND_SOC_DAPM_MIC("Mic", NULL), -}; - -static const struct snd_soc_dapm_route qi_lb60_routes[] = { - {"Mic", NULL, "MIC"}, - {"Speaker", NULL, "LOUT"}, - {"Speaker", NULL, "ROUT"}, -}; - -SND_SOC_DAILINK_DEFS(hifi, - DAILINK_COMP_ARRAY(COMP_CPU("jz4740-i2s")), - DAILINK_COMP_ARRAY(COMP_CODEC("jz4740-codec", "jz4740-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("jz4740-i2s"))); - -static struct snd_soc_dai_link qi_lb60_dai = { - .name = "jz4740", - .stream_name = "jz4740", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - SND_SOC_DAILINK_REG(hifi), -}; - -static struct snd_soc_card qi_lb60_card = { - .name = "QI LB60", - .owner = THIS_MODULE, - .dai_link = &qi_lb60_dai, - .num_links = 1, - - .dapm_widgets = qi_lb60_widgets, - .num_dapm_widgets = ARRAY_SIZE(qi_lb60_widgets), - .dapm_routes = qi_lb60_routes, - .num_dapm_routes = ARRAY_SIZE(qi_lb60_routes), - .fully_routed = true, -}; - -static int qi_lb60_probe(struct platform_device *pdev) -{ - struct qi_lb60 *qi_lb60; - struct snd_soc_card *card = &qi_lb60_card; - - qi_lb60 = devm_kzalloc(&pdev->dev, sizeof(*qi_lb60), GFP_KERNEL); - if (!qi_lb60) - return -ENOMEM; - - qi_lb60->snd_gpio = devm_gpiod_get(&pdev->dev, "snd", GPIOD_OUT_LOW); - if (IS_ERR(qi_lb60->snd_gpio)) - return PTR_ERR(qi_lb60->snd_gpio); - - qi_lb60->amp_gpio = devm_gpiod_get(&pdev->dev, "amp", GPIOD_OUT_LOW); - if (IS_ERR(qi_lb60->amp_gpio)) - return PTR_ERR(qi_lb60->amp_gpio); - - card->dev = &pdev->dev; - - snd_soc_card_set_drvdata(card, qi_lb60); - - return devm_snd_soc_register_card(&pdev->dev, card); -} - -static struct platform_driver qi_lb60_driver = { - .driver = { - .name = "qi-lb60-audio", - }, - .probe = qi_lb60_probe, -}; - -module_platform_driver(qi_lb60_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:qi-lb60-audio"); diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index d16563408465..10ea4fdbeb1e 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -241,7 +241,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; int hd_audio = 0; - int hd_align = 1; + int hd_align = 0; /* set hd mode */ switch (substream->runtime->format) { @@ -254,7 +254,6 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, break; case SNDRV_PCM_FORMAT_S24_LE: hd_audio = 1; - hd_align = 0; break; default: dev_err(afe->dev, "%s() error: unsupported format %d\n", diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 8e3e86619b35..60086858e920 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -99,7 +99,7 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR && MFD_CROS_EC && I2C + depends on QCOM_APR && CROS_EC && I2C select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index c16b0ffe8cfc..d951100bf770 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -422,11 +422,6 @@ static const struct dailink_match_data dailink_match[] = { }, }; -static int of_dev_node_match(struct device *dev, const void *data) -{ - return dev->of_node == data; -} - static int rockchip_sound_codec_node_match(struct device_node *np_codec) { struct device *dev; @@ -438,8 +433,8 @@ static int rockchip_sound_codec_node_match(struct device_node *np_codec) continue; if (dailink_match[i].bus_type) { - dev = bus_find_device(dailink_match[i].bus_type, NULL, - np_codec, of_dev_node_match); + dev = bus_find_device_by_of_node(dailink_match[i].bus_type, + np_codec); if (!dev) continue; put_device(dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 88978a3036c4..4a47ba94559f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -125,6 +125,9 @@ static umode_t soc_dev_attr_is_visible(struct kobject *kobj, struct device *dev = kobj_to_dev(kobj); struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + if (!rtd) + return 0; + if (attr == &dev_attr_pmdown_time.attr) return attr->mode; /* always visible */ return rtd->num_codecs ? attr->mode : 0; /* enabled only with codec */ @@ -285,28 +288,29 @@ static int snd_soc_rtdcom_add(struct snd_soc_pcm_runtime *rtd, return 0; } - rtdcom = kmalloc(sizeof(*rtdcom), GFP_KERNEL); + /* + * created rtdcom here will be freed when rtd->dev was freed. + * see + * soc_free_pcm_runtime() :: device_unregister(rtd->dev) + */ + rtdcom = devm_kzalloc(rtd->dev, sizeof(*rtdcom), GFP_KERNEL); if (!rtdcom) return -ENOMEM; rtdcom->component = component; INIT_LIST_HEAD(&rtdcom->list); + /* + * When rtd was freed, created rtdcom here will be + * also freed. + * And we don't need to call list_del(&rtdcom->list) + * when freed, because rtd is also freed. + */ list_add_tail(&rtdcom->list, &rtd->component_list); return 0; } -static void snd_soc_rtdcom_del_all(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_rtdcom_list *rtdcom1, *rtdcom2; - - for_each_rtdcom_safe(rtd, rtdcom1, rtdcom2) - kfree(rtdcom1); - - INIT_LIST_HEAD(&rtd->component_list); -} - struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd, const char *driver_name) { @@ -355,56 +359,111 @@ EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream); static const struct snd_soc_ops null_snd_soc_ops; +static void soc_release_rtd_dev(struct device *dev) +{ + /* "dev" means "rtd->dev" */ + kfree(dev); +} + +static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd) +{ + if (!rtd) + return; + + kfree(rtd->codec_dais); + list_del(&rtd->list); + + /* + * we don't need to call kfree() for rtd->dev + * see + * soc_release_rtd_dev() + * + * We don't need rtd->dev NULL check, because + * it is alloced *before* rtd. + * see + * soc_new_pcm_runtime() + */ + device_unregister(rtd->dev); + kfree(rtd); +} + static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_pcm_runtime *rtd; + struct device *dev; + int ret; + + /* + * for rtd->dev + */ + dev = kzalloc(sizeof(struct device), GFP_KERNEL); + if (!dev) + return NULL; + + dev->parent = card->dev; + dev->release = soc_release_rtd_dev; + dev->groups = soc_dev_attr_groups; + + dev_set_name(dev, "%s", dai_link->name); + + ret = device_register(dev); + if (ret < 0) { + put_device(dev); /* soc_release_rtd_dev */ + return NULL; + } + /* + * for rtd + */ rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); if (!rtd) - return NULL; + goto free_rtd; - INIT_LIST_HEAD(&rtd->component_list); - rtd->card = card; - rtd->dai_link = dai_link; - if (!rtd->dai_link->ops) - rtd->dai_link->ops = &null_snd_soc_ops; + rtd->dev = dev; + dev_set_drvdata(dev, rtd); + /* + * for rtd->codec_dais + */ rtd->codec_dais = kcalloc(dai_link->num_codecs, sizeof(struct snd_soc_dai *), GFP_KERNEL); - if (!rtd->codec_dais) { - kfree(rtd); - return NULL; - } + if (!rtd->codec_dais) + goto free_rtd; - return rtd; -} + /* + * rtd remaining settings + */ + INIT_LIST_HEAD(&rtd->component_list); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); + INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); -static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd) -{ - kfree(rtd->codec_dais); - snd_soc_rtdcom_del_all(rtd); - kfree(rtd); -} + rtd->card = card; + rtd->dai_link = dai_link; + if (!rtd->dai_link->ops) + rtd->dai_link->ops = &null_snd_soc_ops; -static void soc_add_pcm_runtime(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd) -{ /* see for_each_card_rtds */ list_add_tail(&rtd->list, &card->rtd_list); rtd->num = card->num_rtd; card->num_rtd++; + + return rtd; + +free_rtd: + soc_free_pcm_runtime(rtd); + return NULL; } static void soc_remove_pcm_runtimes(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd, *_rtd; - for_each_card_rtds_safe(card, rtd, _rtd) { - list_del(&rtd->list); + for_each_card_rtds_safe(card, rtd, _rtd) soc_free_pcm_runtime(rtd); - } card->num_rtd = 0; } @@ -930,7 +989,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, } } - soc_add_pcm_runtime(card, rtd); return 0; _err_defer: @@ -1126,7 +1184,6 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order) return 0; } -static void soc_rtd_free(struct snd_soc_pcm_runtime *rtd); /* remove me */ static void soc_remove_link_dais(struct snd_soc_card *card) { int i; @@ -1136,10 +1193,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card) for_each_comp_order(order) { for_each_card_rtds(card, rtd) { - - /* finalize rtd device */ - soc_rtd_free(rtd); - /* remove the CODEC DAI */ for_each_rtd_codec_dai(rtd, i, codec_dai) soc_remove_dai(codec_dai, order); @@ -1417,49 +1470,6 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link); -static void soc_rtd_free(struct snd_soc_pcm_runtime *rtd) -{ - if (rtd->dev_registered) { - /* we don't need to call kfree() for rtd->dev */ - device_unregister(rtd->dev); - rtd->dev_registered = 0; - } -} - -static void soc_rtd_release(struct device *dev) -{ - kfree(dev); -} - -static int soc_rtd_init(struct snd_soc_pcm_runtime *rtd, const char *name) -{ - int ret = 0; - - /* register the rtd device */ - rtd->dev = kzalloc(sizeof(struct device), GFP_KERNEL); - if (!rtd->dev) - return -ENOMEM; - rtd->dev->parent = rtd->card->dev; - rtd->dev->release = soc_rtd_release; - rtd->dev->groups = soc_dev_attr_groups; - dev_set_name(rtd->dev, "%s", name); - dev_set_drvdata(rtd->dev, rtd); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); - ret = device_register(rtd->dev); - if (ret < 0) { - /* calling put_device() here to free the rtd->dev */ - put_device(rtd->dev); - dev_err(rtd->card->dev, - "ASoC: failed to register runtime device: %d\n", ret); - return ret; - } - rtd->dev_registered = 1; - return 0; -} - static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais, struct snd_soc_pcm_runtime *rtd) { @@ -1509,10 +1519,6 @@ static int soc_link_init(struct snd_soc_card *card, return ret; } - ret = soc_rtd_init(rtd, dai_link->name); - if (ret) - return ret; - /* add DPCM sysfs entries */ soc_dpcm_debugfs_add(rtd); @@ -1853,7 +1859,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) for_each_component(component) { - /* does this component override FEs ? */ + /* does this component override BEs ? */ if (!component->driver->ignore_machine) continue; @@ -1874,7 +1880,7 @@ match: continue; } - dev_info(card->dev, "info: override FE DAI link %s\n", + dev_info(card->dev, "info: override BE DAI link %s\n", card->dai_link[i].name); /* override platform component */ diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 748f5f641002..f2c98a9cbf75 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -118,12 +118,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea struct device *dma_dev = dmaengine_dma_dev(pcm, substream); struct dma_chan *chan = pcm->chan[substream->stream]; struct snd_dmaengine_dai_dma_data *dma_data; - struct dma_slave_caps dma_caps; struct snd_pcm_hardware hw; - u32 addr_widths = BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | - BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | - BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); - snd_pcm_format_t i; int ret; if (pcm->config && pcm->config->pcm_hardware) @@ -145,56 +140,12 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) hw.info |= SNDRV_PCM_INFO_BATCH; - ret = dma_get_slave_caps(chan, &dma_caps); - if (ret == 0) { - if (dma_caps.cmd_pause && dma_caps.cmd_resume) - hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; - if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) - hw.info |= SNDRV_PCM_INFO_BATCH; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - addr_widths = dma_caps.dst_addr_widths; - else - addr_widths = dma_caps.src_addr_widths; - } - - /* - * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep - * hw.formats set to 0, meaning no restrictions are in place. - * In this case it's the responsibility of the DAI driver to - * provide the supported format information. - */ - if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) - /* - * Prepare formats mask for valid/allowed sample types. If the - * dma does not have support for the given physical word size, - * it needs to be masked out so user space can not use the - * format which produces corrupted audio. - * In case the dma driver does not implement the slave_caps the - * default assumption is that it supports 1, 2 and 4 bytes - * widths. - */ - for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) { - int bits = snd_pcm_format_physical_width(i); - - /* - * Enable only samples with DMA supported physical - * widths - */ - switch (bits) { - case 8: - case 16: - case 24: - case 32: - case 64: - if (addr_widths & (1 << (bits / 8))) - hw.formats |= pcm_format_to_bits(i); - break; - default: - /* Unsupported types */ - break; - } - } + ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream, + dma_data, + &hw, + chan); + if (ret) + return ret; return snd_soc_set_runtime_hwparams(substream, &hw); } @@ -306,6 +257,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i])) pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; + + if (rtd->pcm->streams[i].pcm->name[0] == '\0') { + strscpy_pad(rtd->pcm->streams[i].pcm->name, + rtd->pcm->streams[i].pcm->id, + sizeof(rtd->pcm->streams[i].pcm->name)); + } } return 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a1b99ac57d9e..66910500e3b6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } @@ -1074,6 +1074,42 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index c25939c5611b..0fd032914a31 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -80,12 +80,6 @@ struct soc_tplg { static int soc_tplg_process_headers(struct soc_tplg *tplg); static void soc_tplg_complete(struct soc_tplg *tplg); -struct snd_soc_dapm_widget * -snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget); -struct snd_soc_dapm_widget * -snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget); static void soc_tplg_denum_remove_texts(struct soc_enum *se); static void soc_tplg_denum_remove_values(struct soc_enum *se); diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index bb8036ae567e..56a3ab66b46b 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -142,6 +142,14 @@ config SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE Say Y if you want to enable caching the memory windows. If unsure, select "N". +config SND_SOC_SOF_DEBUG_ENABLE_FIRMWARE_TRACE + bool "SOF enable firmware trace" + help + The firmware trace can be enabled either at build-time with + this option, or dynamically by setting flags in the SOF core + module parameter (similar to dynamic debug) + If unsure, select "N". + config SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST bool "SOF enable IPC flood test" help @@ -150,6 +158,14 @@ config SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST Say Y if you want to enable IPC flood test. If unsure, select "N". +config SND_SOC_SOF_DEBUG_RETAIN_DSP_CONTEXT + bool "SOF retain DSP context on any FW exceptions" + help + This option keeps the DSP in D0 state so that firmware debug + information can be retained and dumped to userspace. + Say Y if you want to retain DSP context for FW exceptions. + If unsure, select "N". + endif ## SND_SOC_SOF_DEBUG endif ## SND_SOC_SOF_OPTIONS diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 81f28f7ff1a0..5998861a9002 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -16,6 +16,11 @@ #include "sof-priv.h" #include "ops.h" +/* see SOF_DBG_ flags */ +int sof_core_debug; +module_param_named(sof_debug, sof_core_debug, int, 0444); +MODULE_PARM_DESC(sof_debug, "SOF core debug options (0x0 all off)"); + /* SOF defaults if not provided by the platform in ms */ #define TIMEOUT_DEFAULT_IPC_MS 500 #define TIMEOUT_DEFAULT_BOOT_MS 2000 @@ -350,12 +355,20 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) 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); + if (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_FIRMWARE_TRACE) || + (sof_core_debug & SOF_DBG_ENABLE_TRACE)) { + sdev->dtrace_is_supported = true; + + /* 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); + } + } else { + dev_dbg(sdev->dev, "SOF firmware trace disabled\n"); } /* hereafter all FW boot flows are for PM reasons */ @@ -453,7 +466,8 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) 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) + !sof_ops(sdev)->ipc_msg_data || !sof_ops(sdev)->ipc_pcm_params || + !sof_ops(sdev)->fw_ready) return -EINVAL; INIT_LIST_HEAD(&sdev->pcm_list); diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 54cd431faab7..b8a4e899154c 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -461,3 +461,19 @@ void snd_sof_free_debug(struct snd_sof_dev *sdev) debugfs_remove_recursive(sdev->debugfs_root); } EXPORT_SYMBOL_GPL(snd_sof_free_debug); + +void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev) +{ + if (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_RETAIN_DSP_CONTEXT) || + (sof_core_debug & SOF_DBG_RETAIN_CTX)) { + /* should we prevent DSP entering D3 ? */ + dev_info(sdev->dev, "info: preventing DSP entering D3 state to preserve context\n"); + pm_runtime_get_noresume(sdev->dev); + } + + /* dump vital information to the logs */ + snd_sof_dsp_dbg_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX); + snd_sof_ipc_dump(sdev); + snd_sof_trace_notify_for_error(sdev); +} +EXPORT_SYMBOL(snd_sof_handle_fw_exception); diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 80e2826fb447..f395d0638876 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -247,7 +247,7 @@ 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; + u32 status, panic, imrx, imrd; /* now try generic SOF status messages */ status = snd_sof_dsp_read(sdev, BDW_DSP_BAR, SHIM_IPCD); @@ -256,6 +256,26 @@ static void bdw_dump(struct snd_sof_dev *sdev, u32 flags) BDW_STACK_DUMP_SIZE); snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack, BDW_STACK_DUMP_SIZE); + + /* provide some context for firmware debug */ + imrx = snd_sof_dsp_read(sdev, BDW_DSP_BAR, SHIM_IMRX); + imrd = snd_sof_dsp_read(sdev, BDW_DSP_BAR, SHIM_IMRD); + dev_err(sdev->dev, + "error: ipc host -> DSP: pending %s complete %s raw 0x%8.8x\n", + panic & SHIM_IPCX_BUSY ? "yes" : "no", + panic & SHIM_IPCX_DONE ? "yes" : "no", panic); + dev_err(sdev->dev, + "error: mask host: pending %s complete %s raw 0x%8.8x\n", + imrx & SHIM_IMRX_BUSY ? "yes" : "no", + imrx & SHIM_IMRX_DONE ? "yes" : "no", imrx); + dev_err(sdev->dev, + "error: ipc DSP -> host: pending %s complete %s raw 0x%8.8x\n", + status & SHIM_IPCD_BUSY ? "yes" : "no", + status & SHIM_IPCD_DONE ? "yes" : "no", status); + dev_err(sdev->dev, + "error: mask DSP: pending %s complete %s raw 0x%8.8x\n", + imrd & SHIM_IMRD_BUSY ? "yes" : "no", + imrd & SHIM_IMRD_DONE ? "yes" : "no", imrd); } /* diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index a1e514f71739..b2597ecfdc1c 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -145,7 +145,7 @@ 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; + u32 status, panic, imrd, imrx; /* now try generic SOF status messages */ status = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCD); @@ -154,6 +154,27 @@ static void byt_dump(struct snd_sof_dev *sdev, u32 flags) BYT_STACK_DUMP_SIZE); snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack, BYT_STACK_DUMP_SIZE); + + /* provide some context for firmware debug */ + imrx = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IMRX); + imrd = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IMRD); + dev_err(sdev->dev, + "error: ipc host -> DSP: pending %s complete %s raw 0x%8.8x\n", + panic & SHIM_IPCX_BUSY ? "yes" : "no", + panic & SHIM_IPCX_DONE ? "yes" : "no", panic); + dev_err(sdev->dev, + "error: mask host: pending %s complete %s raw 0x%8.8x\n", + imrx & SHIM_IMRX_BUSY ? "yes" : "no", + imrx & SHIM_IMRX_DONE ? "yes" : "no", imrx); + dev_err(sdev->dev, + "error: ipc DSP -> host: pending %s complete %s raw 0x%8.8x\n", + status & SHIM_IPCD_BUSY ? "yes" : "no", + status & SHIM_IPCD_DONE ? "yes" : "no", status); + dev_err(sdev->dev, + "error: mask DSP: pending %s complete %s raw 0x%8.8x\n", + imrd & SHIM_IMRD_BUSY ? "yes" : "no", + imrd & SHIM_IMRD_DONE ? "yes" : "no", imrd); + } /* diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 9b730f183529..575f5f5877d8 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -89,6 +89,7 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev, 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; + struct sof_ipc_fw_version *v = &sdev->fw_ready.version; int ret; u32 size, rate, bits; @@ -116,9 +117,17 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev, /* 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; + /* update no_stream_position flag for ipc params */ + if (hda && hda->no_ipc_position) { + /* For older ABIs set host_period_bytes to zero to inform + * FW we don't want position updates. Newer versions use + * no_stream_position for this purpose. + */ + if (v->abi_version < SOF_ABI_VER(3, 10, 0)) + ipc_params->host_period_bytes = 0; + else + ipc_params->no_stream_position = 1; + } ipc_params->stream_tag = hstream->stream_tag; diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index b2f359d2f7e5..26675dfe0240 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -210,9 +210,7 @@ static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg, 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_ipc_dump(ipc->sdev); - snd_sof_trace_notify_for_error(ipc->sdev); + snd_sof_handle_fw_exception(ipc->sdev); ret = -ETIMEDOUT; } else { /* copy the data returned from DSP */ @@ -794,12 +792,6 @@ 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; diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 730f3259dd02..44f789bf7fb0 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -28,10 +28,15 @@ #include <uapi/sound/sof/fw.h> /* 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) +#define SOF_DBG_ENABLE_TRACE BIT(0) +#define SOF_DBG_REGS BIT(1) +#define SOF_DBG_MBOX BIT(2) +#define SOF_DBG_TEXT BIT(3) +#define SOF_DBG_PCI BIT(4) +#define SOF_DBG_RETAIN_CTX BIT(5) /* prevent DSP D3 on FW exception */ + +/* global debug state set by SOF_DBG_ flags */ +extern int sof_core_debug; /* max BARs mmaped devices can use */ #define SND_SOF_BARS 8 @@ -128,7 +133,7 @@ struct snd_sof_dsp_ops { * 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 */ + int (*fw_ready)(struct snd_sof_dev *sdev, u32 msg_id); /* mandatory */ /* connect pcm substream to a host stream */ int (*pcm_open)(struct snd_sof_dev *sdev, @@ -434,6 +439,7 @@ struct snd_sof_dev { int dma_trace_pages; wait_queue_head_t trace_sleep; u32 host_offset; + u32 dtrace_is_supported; /* set with Kconfig or module parameter */ u32 dtrace_is_enabled; u32 dtrace_error; u32 dtrace_draining; @@ -575,6 +581,7 @@ void snd_sof_get_status(struct snd_sof_dev *sdev, u32 panic_code, struct sof_ipc_panic_info *panic_info, void *stack, size_t stack_words); int snd_sof_init_trace_ipc(struct snd_sof_dev *sdev); +void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev); /* * Platform specific ops. diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index 4c3cff031fd6..b0e4556c8536 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -162,6 +162,9 @@ int snd_sof_init_trace_ipc(struct snd_sof_dev *sdev) struct sof_ipc_reply ipc_reply; int ret; + if (!sdev->dtrace_is_supported) + return 0; + if (sdev->dtrace_is_enabled || !sdev->dma_trace_pages) return -EINVAL; @@ -222,6 +225,9 @@ int snd_sof_init_trace(struct snd_sof_dev *sdev) { int ret; + if (!sdev->dtrace_is_supported) + return 0; + /* set false before start initialization */ sdev->dtrace_is_enabled = false; @@ -277,6 +283,9 @@ 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_supported) + return 0; + if (sdev->dtrace_is_enabled && sdev->host_offset != posn->host_offset) { sdev->host_offset = posn->host_offset; wake_up(&sdev->trace_sleep); @@ -293,6 +302,9 @@ int snd_sof_trace_update_pos(struct snd_sof_dev *sdev, /* 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_supported) + return; + if (sdev->dtrace_is_enabled) { dev_err(sdev->dev, "error: waking up any trace sleepers\n"); sdev->dtrace_error = true; @@ -305,7 +317,7 @@ void snd_sof_release_trace(struct snd_sof_dev *sdev) { int ret; - if (!sdev->dtrace_is_enabled) + if (!sdev->dtrace_is_supported || !sdev->dtrace_is_enabled) return; ret = snd_sof_dma_trace_trigger(sdev, SNDRV_PCM_TRIGGER_STOP); @@ -326,6 +338,9 @@ EXPORT_SYMBOL(snd_sof_release_trace); void snd_sof_free_trace(struct snd_sof_dev *sdev) { + if (!sdev->dtrace_is_supported) + return; + snd_sof_release_trace(sdev); snd_dma_free_pages(&sdev->dmatb); diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index ef4273361d0d..e20267504b16 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -100,7 +100,7 @@ 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"); - stm32_sai_pclk_disable(&sai->pdev->dev); + stm32_sai_pclk_disable(&sai->pdev->dev); return -EINVAL; } diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index dee8fc70a64f..8e2fb81ad05c 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -23,14 +23,31 @@ #include "omap-mcbsp.h" #include "../codecs/cx20442.h" +static struct gpio_desc *handset_mute; +static struct gpio_desc *handsfree_mute; + +static int ams_delta_event_handset(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handset_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int ams_delta_event_handsfree(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handsfree_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + /* Board specific DAPM widgets */ static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { /* Handset */ SND_SOC_DAPM_MIC("Mouthpiece", NULL), - SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_HP("Earpiece", ams_delta_event_handset), /* Handsfree/Speakerphone */ SND_SOC_DAPM_MIC("Microphone", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Speaker", ams_delta_event_handsfree), }; /* How they are connected to codec pins */ @@ -542,6 +559,16 @@ static int ams_delta_probe(struct platform_device *pdev) card->dev = &pdev->dev; + handset_mute = devm_gpiod_get(card->dev, "handset_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handset_mute)) + return PTR_ERR(handset_mute); + + handsfree_mute = devm_gpiod_get(card->dev, "handsfree_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handsfree_mute)) + return PTR_ERR(handsfree_mute); + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index f04d9fb5130f..d89b5c928c4d 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -187,57 +187,9 @@ static void toggle_clock(struct davinci_mcbsp_dev *dev, int playback) static void davinci_mcbsp_start(struct davinci_mcbsp_dev *dev, 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); int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); u32 spcr; u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST; - spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - if (spcr & mask) { - /* start off disabled */ - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, - spcr & ~mask); - toggle_clock(dev, playback); - } - if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM | - DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) { - /* Start the sample generator */ - spcr |= DAVINCI_MCBSP_SPCR_GRST; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); - } - - if (playback) { - /* Stop the DMA to avoid data loss */ - /* while the transmitter is out of reset to handle XSYNCERR */ - if (component->driver->ops->trigger) { - int ret = component->driver->ops->trigger(substream, - SNDRV_PCM_TRIGGER_STOP); - if (ret < 0) - printk(KERN_DEBUG "Playback DMA stop failed\n"); - } - - /* Enable the transmitter */ - spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - spcr |= DAVINCI_MCBSP_SPCR_XRST; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); - - /* wait for any unexpected frame sync error to occur */ - udelay(100); - - /* Disable the transmitter to clear any outstanding XSYNCERR */ - spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - spcr &= ~DAVINCI_MCBSP_SPCR_XRST; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); - toggle_clock(dev, playback); - - /* Restart the DMA */ - if (component->driver->ops->trigger) { - int ret = component->driver->ops->trigger(substream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - printk(KERN_DEBUG "Playback DMA start failed\n"); - } - } /* Enable transmitter or receiver */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -575,7 +527,41 @@ static int davinci_i2s_prepare(struct snd_pcm_substream *substream, { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + u32 spcr; + u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST; + davinci_mcbsp_stop(dev, playback); + + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + if (spcr & mask) { + /* start off disabled */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, + spcr & ~mask); + toggle_clock(dev, playback); + } + if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) { + /* Start the sample generator */ + spcr |= DAVINCI_MCBSP_SPCR_GRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + } + + if (playback) { + /* Enable the transmitter */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr |= DAVINCI_MCBSP_SPCR_XRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + + /* wait for any unexpected frame sync error to occur */ + udelay(100); + + /* Disable the transmitter to clear any outstanding XSYNCERR */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr &= ~DAVINCI_MCBSP_SPCR_XRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + toggle_clock(dev, playback); + } + return 0; } diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 48970efe7838..fb652e73abeb 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -564,7 +564,6 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) int ret; u32 val; struct xlnx_pcm_drv_data *aud_drv_data; - struct resource *res; struct device *dev = &pdev->dev; aud_drv_data = devm_kzalloc(dev, sizeof(*aud_drv_data), GFP_KERNEL); @@ -584,13 +583,7 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "audio formatter node:addr to resource failed\n"); - ret = -ENXIO; - goto clk_err; - } - aud_drv_data->mmio = devm_ioremap_resource(dev, res); + aud_drv_data->mmio = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(aud_drv_data->mmio)) { dev_err(dev, "audio formatter ioremap failed\n"); ret = PTR_ERR(aud_drv_data->mmio); |