From d4e44f1418d71cf53d685e236a95c525089e065a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 18 Mar 2016 12:28:49 +0200 Subject: ASoC: omap-mcbsp: Enable/disable sidetone block auto clock gating for omap3 OMAP3's McBSP2 and McBSP3 module have integrated sidetone block with dedicated SYSCONFIG register. The sidetone is operating from the maain McBSP module's ICLK. For normal operation the sidetone clock auto idle support needs to be disabled when it is activated. Note: This is not enough to avoid choppy sidetone because this AUTOIDLE bit is controlling only the clock auto idle from the McBSP to the sidetone block. If the McBSP_ICLK is idling, the sidetone clock is going to do the same. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/mcbsp.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index c7563e230c7d..4a16e778966b 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -260,6 +260,10 @@ static void omap_st_on(struct omap_mcbsp *mcbsp) if (mcbsp->pdata->enable_st_clock) mcbsp->pdata->enable_st_clock(mcbsp->id, 1); + /* Disable Sidetone clock auto-gating for normal operation */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE)); + /* Enable McBSP Sidetone */ w = MCBSP_READ(mcbsp, SSELCR); MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN); @@ -279,6 +283,10 @@ static void omap_st_off(struct omap_mcbsp *mcbsp) w = MCBSP_READ(mcbsp, SSELCR); MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN)); + /* Enable Sidetone clock auto-gating to reduce power consumption */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE); + if (mcbsp->pdata->enable_st_clock) mcbsp->pdata->enable_st_clock(mcbsp->id, 0); } -- cgit v1.2.3-58-ga151 From 465011fc56717f0227d1aa7a99cce000abc614d8 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sun, 27 Mar 2016 19:06:14 -0300 Subject: ASoC: wm8960: Provide a menu selection text Provide a menu selection text so that users can enable, disable or mark it as module in menuconfig. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..196101b2eab5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -909,7 +909,7 @@ config SND_SOC_WM8955 tristate config SND_SOC_WM8960 - tristate + tristate "Wolfson Microelectronics WM8960 CODEC" config SND_SOC_WM8961 tristate -- cgit v1.2.3-58-ga151 From 65147846796bd443972d9055b3b4c1339e15d53a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 28 Mar 2016 08:31:18 -0300 Subject: ASoC: wm8962: Disable clock if wm8962_runtime_resume() fails When regulator_bulk_enable() fails inside wm8962_runtime_resume(), we should disable the previously enabled clock. Signed-off-by: Fabio Estevam Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 88223608a33f..f3f71ba0ed12 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3800,7 +3800,7 @@ static int wm8962_runtime_resume(struct device *dev) if (ret != 0) { dev_err(dev, "Failed to enable supplies: %d\n", ret); - return ret; + goto disable_clock; } regcache_cache_only(wm8962->regmap, false); @@ -3838,6 +3838,10 @@ static int wm8962_runtime_resume(struct device *dev) msleep(5); return 0; + +disable_clock: + clk_disable_unprepare(wm8962->pdata.mclk); + return ret; } static int wm8962_runtime_suspend(struct device *dev) -- cgit v1.2.3-58-ga151 From 8bfa934e10d99b524bfe80b793e235b9188a7b58 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 28 Mar 2016 08:31:19 -0300 Subject: ASoC: wm8962: Fit error message into a single line The error message fits well into a single line. Signed-off-by: Fabio Estevam Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index f3f71ba0ed12..93f75dcee388 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3798,8 +3798,7 @@ static int wm8962_runtime_resume(struct device *dev) ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); if (ret != 0) { - dev_err(dev, - "Failed to enable supplies: %d\n", ret); + dev_err(dev, "Failed to enable supplies: %d\n", ret); goto disable_clock; } -- cgit v1.2.3-58-ga151 From 937e92dc50231bb41294262abf56d2bdddc4d38c Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sun, 27 Mar 2016 18:36:13 -0300 Subject: ASoC: wm8962: Adjust clk definitions so that simple card can work When trying to use simple card with wm8962 the following probe error happens: wm8962 0-001a: simple-card: set_sysclk error asoc-simple-card sound: ASoC: failed to init 202c000.ssi-wm8962: -22 asoc-simple-card sound: ASoC: failed to instantiate card -22 asoc-simple-card: probe of sound failed with error -22 In simple-card.c the snd_soc_dai_set_sysclk() function is called with clk_id as 0, which is an invalid clock for wm8962. Adjust the clocks source definitions in wm8962.h so that the simple card driver can work successfully. Signed-off-by: Fabio Estevam Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index 910aafd09d21..e63a318a3015 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -16,9 +16,9 @@ #include #include -#define WM8962_SYSCLK_MCLK 1 -#define WM8962_SYSCLK_FLL 2 -#define WM8962_SYSCLK_PLL3 3 +#define WM8962_SYSCLK_MCLK 0 +#define WM8962_SYSCLK_FLL 1 +#define WM8962_SYSCLK_PLL3 2 #define WM8962_FLL 1 -- cgit v1.2.3-58-ga151 From 0400485076e8bb167d5f4b3eb5f6d05e4b4361b7 Mon Sep 17 00:00:00 2001 From: Petr Kulhavy Date: Tue, 29 Mar 2016 09:39:36 +0200 Subject: ASoC: tas571x: implemented digital mute The driver did not have a mute function. The amplifier was brought out of shutdown mode (hard-mute) once for ever in probe(), which was causing clicks and pops when altering the I2C register configuration later. This adds the digital_mute() function. The amplifier unmute in probe() was removed. Signed-off-by: Petr Kulhavy Signed-off-by: Mark Brown --- sound/soc/codecs/tas571x.c | 22 ++++++++++++++++++---- 1 file changed, 18 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad41a34..d003c6ce0794 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -167,6 +167,23 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream, TAS571X_SDI_FMT_MASK, val); } +static int tas571x_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 sysctl2; + int ret; + + sysctl2 = mute ? TAS571X_SYS_CTRL_2_SDN_MASK : 0; + + ret = snd_soc_update_bits(codec, + TAS571X_SYS_CTRL_2_REG, + TAS571X_SYS_CTRL_2_SDN_MASK, + sysctl2); + usleep_range(1000, 2000); + + return ret; +} + static int tas571x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -214,6 +231,7 @@ static int tas571x_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops tas571x_dai_ops = { .set_fmt = tas571x_set_dai_fmt, .hw_params = tas571x_hw_params, + .digital_mute = tas571x_mute, }; static const char *const tas5711_supply_names[] = { @@ -445,10 +463,6 @@ static int tas571x_i2c_probe(struct i2c_client *client, if (ret) return ret; - ret = regmap_update_bits(priv->regmap, TAS571X_SYS_CTRL_2_REG, - TAS571X_SYS_CTRL_2_SDN_MASK, 0); - if (ret) - return ret; memcpy(&priv->codec_driver, &tas571x_codec, sizeof(priv->codec_driver)); priv->codec_driver.controls = priv->chip->controls; -- cgit v1.2.3-58-ga151 From 2dfadff69e8b1da8f8661e9edb131b208cc389b7 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 30 Mar 2016 18:25:07 +0800 Subject: ASoC: rt5677: Avoid duplicate the same test in each switch case Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 24 +++++------------------- 1 file changed, 5 insertions(+), 19 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 33e290b703df..b3f1db5bae4a 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1241,60 +1241,46 @@ static int rt5677_dmic_use_asrc(struct snd_soc_dapm_widget *source, regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO1_CLK_SEL_MASK) >> RT5677_AD_STO1_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 10: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO2_CLK_SEL_MASK) >> RT5677_AD_STO2_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 9: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO3_CLK_SEL_MASK) >> RT5677_AD_STO3_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 8: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO4_CLK_SEL_MASK) >> RT5677_AD_STO4_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 7: regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_MONOL_CLK_SEL_MASK) >> RT5677_AD_MONOL_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 6: regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_MONOR_CLK_SEL_MASK) >> RT5677_AD_MONOR_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; default: - break; + return 0; } + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + return 0; } -- cgit v1.2.3-58-ga151 From 3fcdfc9dad07c2a6ea19dfb98a2151cd86d06c6a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 Mar 2016 11:02:49 -0700 Subject: ASoC: wm8960: Depends on I2C Now that this is directly user selectable it needs to care about its dependencies. Reported-by: kbuild test robot Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 196101b2eab5..3ebf29bc87d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -910,6 +910,7 @@ config SND_SOC_WM8955 config SND_SOC_WM8960 tristate "Wolfson Microelectronics WM8960 CODEC" + depends on I2C config SND_SOC_WM8961 tristate -- cgit v1.2.3-58-ga151 From 630e413dc24da0c2373fd7592aeb0e08cea71cd1 Mon Sep 17 00:00:00 2001 From: Petr Kulhavy Date: Tue, 29 Mar 2016 09:39:35 +0200 Subject: ASoC: tas571x: chip type detection via I2C name The chip selection was relying only on DT. It was not possible to use the driver without DT. This adds the chip type detection from the I2C name, which allows to use the driver from the platform driver without DT. Signed-off-by: Petr Kulhavy Reviewed-by: Kevin Cernekee Signed-off-by: Mark Brown --- sound/soc/codecs/tas571x.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index d003c6ce0794..aafee9bbe01a 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -404,11 +404,10 @@ static int tas571x_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, priv); of_id = of_match_device(tas571x_of_match, dev); - if (!of_id) { - dev_err(dev, "Unknown device type\n"); - return -EINVAL; - } - priv->chip = of_id->data; + if (of_id) + priv->chip = of_id->data; + else + priv->chip = (void *) id->driver_data; priv->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(priv->mclk) && PTR_ERR(priv->mclk) != -ENOENT) { @@ -505,9 +504,9 @@ static const struct of_device_id tas571x_of_match[] = { MODULE_DEVICE_TABLE(of, tas571x_of_match); static const struct i2c_device_id tas571x_i2c_id[] = { - { "tas5711", 0 }, - { "tas5717", 0 }, - { "tas5719", 0 }, + { "tas5711", (kernel_ulong_t) &tas5711_chip }, + { "tas5717", (kernel_ulong_t) &tas5717_chip }, + { "tas5719", (kernel_ulong_t) &tas5717_chip }, { } }; MODULE_DEVICE_TABLE(i2c, tas571x_i2c_id); -- cgit v1.2.3-58-ga151 From a593ed09040fa611f37953afe455e64c7653160d Mon Sep 17 00:00:00 2001 From: Petr Kulhavy Date: Thu, 31 Mar 2016 18:41:25 +0200 Subject: ASoC: tas571x: added missing register literals The list of TAS571x registers was incomplete. Added the missing register definitions up to the register 0x25. Added volatile and read-only register tables into tas5711_regmap_config and tas5717_regmap_config. The chip has 256 registers in total. But from address 0x29 on (0x26 to 0x28 are reserved) the register width varies between 20, 12 and 8 bytes, which the register map cannot represent. Signed-off-by: Petr Kulhavy Signed-off-by: Mark Brown --- sound/soc/codecs/tas571x.c | 28 ++++++++++++++++++++++++++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 50 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index aafee9bbe01a..ef6c8d9b251a 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } @@ -259,6 +263,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = { 1, 1), }; +static const struct regmap_range tas571x_readonly_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG), +}; + +static const struct regmap_range tas571x_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), +}; + +static const struct regmap_access_table tas571x_write_regs = { + .no_ranges = tas571x_readonly_regs_range, + .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range), +}; + +static const struct regmap_access_table tas571x_volatile_regs = { + .yes_ranges = tas571x_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range), + +}; + static const struct reg_default tas5711_reg_defaults[] = { { 0x04, 0x05 }, { 0x05, 0x40 }, @@ -278,6 +302,8 @@ static const struct regmap_config tas5711_regmap_config = { .reg_defaults = tas5711_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; static const struct tas571x_chip tas5711_chip = { @@ -332,6 +358,8 @@ static const struct regmap_config tas5717_regmap_config = { .reg_defaults = tas5717_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; /* This entry is reused for tas5719 as the software interface is identical. */ diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471232cd..cf800c364f0f 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- cgit v1.2.3-58-ga151 From 23a282c4f088efb337957ffa21c677d30eda0784 Mon Sep 17 00:00:00 2001 From: Petr Kulhavy Date: Thu, 31 Mar 2016 18:41:26 +0200 Subject: ASoC: tas571x: added support for TAS5721 This adds support for TAS5721. Signed-off-by: Petr Kulhavy Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/tas571x.c | 76 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 77 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..c011f076d58b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -737,7 +737,7 @@ config SND_SOC_TAS5086 depends on I2C config SND_SOC_TAS571X - tristate "Texas Instruments TAS5711/TAS5717/TAS5719 power amplifiers" + tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers" depends on I2C config SND_SOC_TFA9879 diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index ef6c8d9b251a..b8d19b77bde9 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -4,6 +4,9 @@ * Copyright (C) 2015 Google, Inc. * Copyright (c) 2013 Daniel Mack * + * TAS5721 support: + * Copyright (C) 2016 Petr Kulhavy, Barix AG + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -372,6 +375,77 @@ static const struct tas571x_chip tas5717_chip = { .vol_reg_size = 2, }; +static const char *const tas5721_supply_names[] = { + "AVDD", + "DVDD", + "DRVDD", + "PVDD", +}; + +static const struct snd_kcontrol_new tas5721_controls[] = { + SOC_SINGLE_TLV("Master Volume", + TAS571X_MVOL_REG, + 0, 0xff, 1, tas5711_volume_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", + TAS571X_CH1_VOL_REG, + TAS571X_CH2_VOL_REG, + 0, 0xff, 1, tas5711_volume_tlv), + SOC_DOUBLE("Speaker Switch", + TAS571X_SOFT_MUTE_REG, + TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, + 1, 1), +}; + +static const struct reg_default tas5721_reg_defaults[] = { + {TAS571X_CLK_CTRL_REG, 0x6c}, + {TAS571X_DEV_ID_REG, 0x00}, + {TAS571X_ERR_STATUS_REG, 0x00}, + {TAS571X_SYS_CTRL_1_REG, 0xa0}, + {TAS571X_SDI_REG, 0x05}, + {TAS571X_SYS_CTRL_2_REG, 0x40}, + {TAS571X_SOFT_MUTE_REG, 0x00}, + {TAS571X_MVOL_REG, 0xff}, + {TAS571X_CH1_VOL_REG, 0x30}, + {TAS571X_CH2_VOL_REG, 0x30}, + {TAS571X_CH3_VOL_REG, 0x30}, + {TAS571X_VOL_CFG_REG, 0x91}, + {TAS571X_MODULATION_LIMIT_REG, 0x02}, + {TAS571X_IC_DELAY_CH1_REG, 0xac}, + {TAS571X_IC_DELAY_CH2_REG, 0x54}, + {TAS571X_IC_DELAY_CH3_REG, 0xac}, + {TAS571X_IC_DELAY_CH4_REG, 0x54}, + {TAS571X_PWM_CH_SDN_GROUP_REG, 0x30}, + {TAS571X_START_STOP_PERIOD_REG, 0x0f}, + {TAS571X_OSC_TRIM_REG, 0x82}, + {TAS571X_BKND_ERR_REG, 0x02}, + {TAS571X_INPUT_MUX_REG, 0x17772}, + {TAS571X_CH4_SRC_SELECT_REG, 0x4303}, + {TAS571X_PWM_MUX_REG, 0x1021345}, +}; + +static const struct regmap_config tas5721_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .max_register = 0xff, + .reg_read = tas571x_reg_read, + .reg_write = tas571x_reg_write, + .reg_defaults = tas5721_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5721_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, +}; + + +static const struct tas571x_chip tas5721_chip = { + .supply_names = tas5721_supply_names, + .num_supply_names = ARRAY_SIZE(tas5721_supply_names), + .controls = tas5711_controls, + .num_controls = ARRAY_SIZE(tas5711_controls), + .regmap_config = &tas5721_regmap_config, + .vol_reg_size = 1, +}; + static const struct snd_soc_dapm_widget tas571x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0), @@ -527,6 +601,7 @@ static const struct of_device_id tas571x_of_match[] = { { .compatible = "ti,tas5711", .data = &tas5711_chip, }, { .compatible = "ti,tas5717", .data = &tas5717_chip, }, { .compatible = "ti,tas5719", .data = &tas5717_chip, }, + { .compatible = "ti,tas5721", .data = &tas5721_chip, }, { } }; MODULE_DEVICE_TABLE(of, tas571x_of_match); @@ -535,6 +610,7 @@ static const struct i2c_device_id tas571x_i2c_id[] = { { "tas5711", (kernel_ulong_t) &tas5711_chip }, { "tas5717", (kernel_ulong_t) &tas5717_chip }, { "tas5719", (kernel_ulong_t) &tas5717_chip }, + { "tas5721", (kernel_ulong_t) &tas5721_chip }, { } }; MODULE_DEVICE_TABLE(i2c, tas571x_i2c_id); -- cgit v1.2.3-58-ga151 From ea4d25d5a3709cba0d3df2195edffc400170394f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 31 Mar 2016 19:23:14 +0100 Subject: ASoC: qcom: Fix uninitialized symbol warning. This patch fixes following static checker warning, by initializing the ret to -EINVAL, as one of the code path in lpass_platform_pcm_new() uses this variable uninitialized. sound/soc/qcom/lpass-platform.c:555 lpass_platform_pcm_new() error: uninitialized symbol 'ret'. Reported-by: Dan Carpenter Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 6e8665430bd5..f5cae01d18bd 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -474,7 +474,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) struct lpass_data *drvdata = snd_soc_platform_get_drvdata(soc_runtime->platform); struct lpass_variant *v = drvdata->variant; - int ret; + int ret = -EINVAL; struct lpass_pcm_data *data; size_t size = lpass_platform_pcm_hardware.buffer_bytes_max; -- cgit v1.2.3-58-ga151 From cef794f76485f4a4e6affd851ae9f9d0eb4287a5 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 31 Mar 2016 19:23:15 +0100 Subject: ASoC: qcom: remove IS_ERR_VALUE usage on int. IS_ERR_VALUE should be used only with unsigned long type, signed types should use comparison 'ret < 0' This patch removes such usages. Reported-by: Dan Carpenter Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index f5cae01d18bd..db000c6987a1 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -491,7 +491,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) data->rdma_ch = v->alloc_dma_channel(drvdata, SNDRV_PCM_STREAM_PLAYBACK); - if (IS_ERR_VALUE(data->rdma_ch)) + if (data->rdma_ch < 0) return data->rdma_ch; drvdata->substream[data->rdma_ch] = psubstream; @@ -518,8 +518,10 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) data->wrdma_ch = v->alloc_dma_channel(drvdata, SNDRV_PCM_STREAM_CAPTURE); - if (IS_ERR_VALUE(data->wrdma_ch)) + if (data->wrdma_ch < 0) { + ret = data->wrdma_ch; goto capture_alloc_err; + } drvdata->substream[data->wrdma_ch] = csubstream; -- cgit v1.2.3-58-ga151 From 5ba10dd4a1727714faaac9a48f6aa69d8ee33248 Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 31 Mar 2016 18:00:55 +0200 Subject: ASoC: sti: correct typo errors Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/st,sti-asoc-card.txt | 4 ++-- sound/soc/sti/uniperif.h | 22 +++++++++++----------- 2 files changed, 13 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index 028fa1c82f50..f546dd914812 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -65,7 +65,7 @@ Example: reg = <0x8D82000 0x158>; interrupts = ; dmas = <&fdma0 4 0 1>; - dai-name = "Uni Player #1 (DAC)"; + dai-name = "Uni Player #2 (DAC)"; dma-names = "tx"; uniperiph-id = <2>; version = <5>; @@ -82,7 +82,7 @@ Example: interrupts = ; dmas = <&fdma0 7 0 1>; dma-names = "tx"; - dai-name = "Uni Player #1 (PIO)"; + dai-name = "Uni Player #3 (SPDIF)"; uniperiph-id = <3>; version = <5>; mode = "SPDIF"; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index f0fd5a9944e9..1f82faafb869 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -25,7 +25,7 @@ writel_relaxed((((value) & mask) << shift), ip->base + offset) /* - * AUD_UNIPERIF_SOFT_RST reg + * UNIPERIF_SOFT_RST reg */ #define UNIPERIF_SOFT_RST_OFFSET(ip) 0x0000 @@ -50,7 +50,7 @@ UNIPERIF_SOFT_RST_SOFT_RST_MASK(ip)) /* - * AUD_UNIPERIF_FIFO_DATA reg + * UNIPERIF_FIFO_DATA reg */ #define UNIPERIF_FIFO_DATA_OFFSET(ip) 0x0004 @@ -58,7 +58,7 @@ writel_relaxed(value, ip->base + UNIPERIF_FIFO_DATA_OFFSET(ip)) /* - * AUD_UNIPERIF_CHANNEL_STA_REGN reg + * UNIPERIF_CHANNEL_STA_REGN reg */ #define UNIPERIF_CHANNEL_STA_REGN(ip, n) (0x0060 + (4 * n)) @@ -105,7 +105,7 @@ writel_relaxed(value, ip->base + UNIPERIF_CHANNEL_STA_REG5_OFFSET(ip)) /* - * AUD_UNIPERIF_ITS reg + * UNIPERIF_ITS reg */ #define UNIPERIF_ITS_OFFSET(ip) 0x000C @@ -143,7 +143,7 @@ 0 : (BIT(UNIPERIF_ITS_UNDERFLOW_REC_FAILED_SHIFT(ip)))) /* - * AUD_UNIPERIF_ITS_BCLR reg + * UNIPERIF_ITS_BCLR reg */ /* FIFO_ERROR */ @@ -160,7 +160,7 @@ writel_relaxed(value, ip->base + UNIPERIF_ITS_BCLR_OFFSET(ip)) /* - * AUD_UNIPERIF_ITM reg + * UNIPERIF_ITM reg */ #define UNIPERIF_ITM_OFFSET(ip) 0x0018 @@ -188,7 +188,7 @@ 0 : (BIT(UNIPERIF_ITM_UNDERFLOW_REC_FAILED_SHIFT(ip)))) /* - * AUD_UNIPERIF_ITM_BCLR reg + * UNIPERIF_ITM_BCLR reg */ #define UNIPERIF_ITM_BCLR_OFFSET(ip) 0x001c @@ -213,7 +213,7 @@ UNIPERIF_ITM_BCLR_DMA_ERROR_MASK(ip)) /* - * AUD_UNIPERIF_ITM_BSET reg + * UNIPERIF_ITM_BSET reg */ #define UNIPERIF_ITM_BSET_OFFSET(ip) 0x0020 @@ -767,7 +767,7 @@ SET_UNIPERIF_REG(ip, \ UNIPERIF_CTRL_OFFSET(ip), \ UNIPERIF_CTRL_READER_OUT_SEL_SHIFT(ip), \ - CORAUD_UNIPERIF_CTRL_READER_OUT_SEL_MASK(ip), 1) + UNIPERIF_CTRL_READER_OUT_SEL_MASK(ip), 1) /* UNDERFLOW_REC_WINDOW */ #define UNIPERIF_CTRL_UNDERFLOW_REC_WINDOW_SHIFT(ip) 20 @@ -1046,7 +1046,7 @@ UNIPERIF_STATUS_1_UNDERFLOW_DURATION_MASK(ip), value) /* - * AUD_UNIPERIF_CHANNEL_STA_REGN reg + * UNIPERIF_CHANNEL_STA_REGN reg */ #define UNIPERIF_CHANNEL_STA_REGN(ip, n) (0x0060 + (4 * n)) @@ -1057,7 +1057,7 @@ UNIPERIF_CHANNEL_STA_REGN(ip, n)) /* - * AUD_UNIPERIF_USER_VALIDITY reg + * UNIPERIF_USER_VALIDITY reg */ #define UNIPERIF_USER_VALIDITY_OFFSET(ip) 0x0090 -- cgit v1.2.3-58-ga151 From 38535e8e6962405b7d8a365f2f367f62bca98565 Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 7 Apr 2016 11:25:30 +0200 Subject: ASoC: sti: macro for uniperif tdm regs access Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/uniperif.h | 112 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 112 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 1f82faafb869..400788bc9190 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1100,6 +1100,118 @@ UNIPERIF_DBG_STANDBY_LEFT_SP_SHIFT(ip), \ UNIPERIF_DBG_STANDBY_LEFT_SP_MASK(ip), value) +/* + * UNIPERIF_TDM_ENABLE + */ +#define UNIPERIF_TDM_ENABLE_OFFSET(ip) 0x0118 +#define GET_UNIPERIF_TDM_ENABLE(ip) \ + readl_relaxed(ip->base + UNIPERIF_TDM_ENABLE_OFFSET(ip)) +#define SET_UNIPERIF_TDM_ENABLE(ip, value) \ + writel_relaxed(value, ip->base + UNIPERIF_TDM_ENABLE_OFFSET(ip)) + +/* TDM_ENABLE */ +#define UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip) 0x0 +#define UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip) 0x1 +#define GET_UNIPERIF_TDM_ENABLE_EN_TDM(ip) \ + GET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_ENABLE_OFFSET(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip)) +#define SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_ENABLE_OFFSET(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip), 1) +#define SET_UNIPERIF_TDM_ENABLE_TDM_DISABLE(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_ENABLE_OFFSET(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip), 0) + +/* + * UNIPERIF_TDM_FS_REF_FREQ + */ +#define UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip) 0x011c +#define GET_UNIPERIF_TDM_FS_REF_FREQ(ip) \ + readl_relaxed(ip->base + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip)) +#define SET_UNIPERIF_TDM_FS_REF_FREQ(ip, value) \ + writel_relaxed(value, ip->base + \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip)) + +/* REF_FREQ */ +#define UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip) 0x0 +#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip) 0 +#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip) 1 +#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip) 2 +#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip) 3 +#define UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip) 0x3 +#define GET_UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ(ip) \ + GET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip)) +#define SET_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \ + VALUE_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip)) +#define SET_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \ + VALUE_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip)) +#define SET_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \ + VALUE_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip)) +#define SET_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \ + VALUE_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip)) + +/* + * UNIPERIF_TDM_FS_REF_DIV + */ +#define UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip) 0x0120 +#define GET_UNIPERIF_TDM_FS_REF_DIV(ip) \ + readl_relaxed(ip->base + UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip)) +#define SET_UNIPERIF_TDM_FS_REF_DIV(ip, value) \ + writel_relaxed(value, ip->base + \ + UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip)) + +/* NUM_TIMESLOT */ +#define UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip) 0x0 +#define UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip) 0xff +#define GET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(ip) \ + GET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip)) +#define SET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(ip, value) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip), value) + +/* + * UNIPERIF_TDM_WORD_POS_X_Y + * 32 bits of UNIPERIF_TDM_WORD_POS_X_Y register shall be set in 1 shot + */ +#define UNIPERIF_TDM_WORD_POS_1_2_OFFSET(ip) 0x013c +#define UNIPERIF_TDM_WORD_POS_3_4_OFFSET(ip) 0x0140 +#define UNIPERIF_TDM_WORD_POS_5_6_OFFSET(ip) 0x0144 +#define UNIPERIF_TDM_WORD_POS_7_8_OFFSET(ip) 0x0148 +#define GET_UNIPERIF_TDM_WORD_POS(ip, words) \ + readl_relaxed(ip->base + UNIPERIF_TDM_WORD_POS_##words##_OFFSET(ip)) +#define SET_UNIPERIF_TDM_WORD_POS(ip, words, value) \ + writel_relaxed(value, ip->base + \ + UNIPERIF_TDM_WORD_POS_##words##_OFFSET(ip)) /* * uniperipheral IP capabilities */ -- cgit v1.2.3-58-ga151 From 5295a0dc31d5261ff64406ece25e8d9e91530d2e Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 7 Apr 2016 11:25:31 +0200 Subject: ASoC: sti: rename unip player type into common player & reader type Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/uniperif.h | 19 ++++++++++++++----- sound/soc/sti/uniperif_player.c | 31 +++++++++++-------------------- 2 files changed, 25 insertions(+), 25 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 400788bc9190..75116ab3cbc0 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1219,6 +1219,15 @@ #define UNIPERIF_FIFO_SIZE 70 /* FIFO is 70 cells deep */ #define UNIPERIF_FIFO_FRAMES 4 /* FDMA trigger limit in frames */ +#define UNIPERIF_TYPE_IS_HDMI(p) \ + ((p)->info->type == SND_ST_UNIPERIF_TYPE_HDMI) +#define UNIPERIF_TYPE_IS_PCM(p) \ + ((p)->info->type == SND_ST_UNIPERIF_TYPE_PCM) +#define UNIPERIF_TYPE_IS_SPDIF(p) \ + ((p)->info->type == SND_ST_UNIPERIF_TYPE_SPDIF) +#define UNIPERIF_TYPE_IS_IEC958(p) \ + (UNIPERIF_TYPE_IS_HDMI(p) || \ + UNIPERIF_TYPE_IS_SPDIF(p)) /* * Uniperipheral IP revisions */ @@ -1237,10 +1246,10 @@ enum uniperif_version { }; enum uniperif_type { - SND_ST_UNIPERIF_PLAYER_TYPE_NONE, - SND_ST_UNIPERIF_PLAYER_TYPE_HDMI, - SND_ST_UNIPERIF_PLAYER_TYPE_PCM, - SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF + SND_ST_UNIPERIF_TYPE_NONE, + SND_ST_UNIPERIF_TYPE_HDMI, + SND_ST_UNIPERIF_TYPE_PCM, + SND_ST_UNIPERIF_TYPE_SPDIF }; enum uniperif_state { @@ -1259,7 +1268,7 @@ enum uniperif_iec958_encoding_mode { struct uniperif_info { int id; /* instance value of the uniperipheral IP */ - enum uniperif_type player_type; + enum uniperif_type type; int underflow_enabled; /* Underflow recovery mode */ }; diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 7aca6b92f718..ab4310a1615e 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -26,15 +26,6 @@ /* * Driver specific types. */ -#define UNIPERIF_PLAYER_TYPE_IS_HDMI(p) \ - ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_HDMI) -#define UNIPERIF_PLAYER_TYPE_IS_PCM(p) \ - ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_PCM) -#define UNIPERIF_PLAYER_TYPE_IS_SPDIF(p) \ - ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF) -#define UNIPERIF_PLAYER_TYPE_IS_IEC958(p) \ - (UNIPERIF_PLAYER_TYPE_IS_HDMI(p) || \ - UNIPERIF_PLAYER_TYPE_IS_SPDIF(p)) #define UNIPERIF_PLAYER_CLK_ADJ_MIN -999999 #define UNIPERIF_PLAYER_CLK_ADJ_MAX 1000000 @@ -738,14 +729,14 @@ static int uni_player_prepare(struct snd_pcm_substream *substream, SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(player, trigger_limit); /* Uniperipheral setup depends on player type */ - switch (player->info->player_type) { - case SND_ST_UNIPERIF_PLAYER_TYPE_HDMI: + switch (player->info->type) { + case SND_ST_UNIPERIF_TYPE_HDMI: ret = uni_player_prepare_iec958(player, runtime); break; - case SND_ST_UNIPERIF_PLAYER_TYPE_PCM: + case SND_ST_UNIPERIF_TYPE_PCM: ret = uni_player_prepare_pcm(player, runtime); break; - case SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF: + case SND_ST_UNIPERIF_TYPE_SPDIF: ret = uni_player_prepare_iec958(player, runtime); break; default: @@ -852,8 +843,8 @@ static int uni_player_start(struct uniperif *player) * will not take affect and hang the player. */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) - if (UNIPERIF_PLAYER_TYPE_IS_IEC958(player)) - SET_UNIPERIF_CTRL_SPDIF_FMT_ON(player); + if (UNIPERIF_TYPE_IS_IEC958(player)) + SET_UNIPERIF_CTRL_SPDIF_FMT_ON(player); /* Force channel status update (no update if clk disable) */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) @@ -1012,13 +1003,13 @@ static int uni_player_parse_dt(struct platform_device *pdev, } if (strcasecmp(mode, "hdmi") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI; + info->type = SND_ST_UNIPERIF_TYPE_HDMI; else if (strcasecmp(mode, "pcm") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_PCM; + info->type = SND_ST_UNIPERIF_TYPE_PCM; else if (strcasecmp(mode, "spdif") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF; + info->type = SND_ST_UNIPERIF_TYPE_SPDIF; else - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_NONE; + info->type = SND_ST_UNIPERIF_TYPE_NONE; /* Save the info structure */ player->info = info; @@ -1087,7 +1078,7 @@ int uni_player_init(struct platform_device *pdev, SET_UNIPERIF_CTRL_SPDIF_LAT_OFF(player); SET_UNIPERIF_CONFIG_IDLE_MOD_DISABLE(player); - if (UNIPERIF_PLAYER_TYPE_IS_IEC958(player)) { + if (UNIPERIF_TYPE_IS_IEC958(player)) { /* Set default iec958 status bits */ /* Consumer, PCM, copyright, 2ch, mode 0 */ -- cgit v1.2.3-58-ga151 From 9a00a3e9fea1ea5affa7bea971299d3a5832daad Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 7 Apr 2016 11:25:32 +0200 Subject: ASoC: sti: define tdm type & default tdm hw config Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/uniperif.h | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 75116ab3cbc0..750eb5a07e6c 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1228,6 +1228,9 @@ #define UNIPERIF_TYPE_IS_IEC958(p) \ (UNIPERIF_TYPE_IS_HDMI(p) || \ UNIPERIF_TYPE_IS_SPDIF(p)) +#define UNIPERIF_TYPE_IS_TDM(p) \ + ((p)->info->type == SND_ST_UNIPERIF_TYPE_TDM) + /* * Uniperipheral IP revisions */ @@ -1249,7 +1252,8 @@ enum uniperif_type { SND_ST_UNIPERIF_TYPE_NONE, SND_ST_UNIPERIF_TYPE_HDMI, SND_ST_UNIPERIF_TYPE_PCM, - SND_ST_UNIPERIF_TYPE_SPDIF + SND_ST_UNIPERIF_TYPE_SPDIF, + SND_ST_UNIPERIF_TYPE_TDM }; enum uniperif_state { @@ -1330,6 +1334,28 @@ struct sti_uniperiph_data { struct sti_uniperiph_dai dai_data; }; +static const struct snd_pcm_hardware uni_tdm_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE, + + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + + .channels_min = 1, + .channels_max = 32, + + .periods_min = 2, + .periods_max = 10, + + .period_bytes_min = 128, + .period_bytes_max = 64 * PAGE_SIZE, + .buffer_bytes_max = 256 * PAGE_SIZE +}; + /* uniperiph player*/ int uni_player_init(struct platform_device *pdev, struct uniperif *uni_player); -- cgit v1.2.3-58-ga151 From 44f948bdb175bf326911c9ba0e47389918161ce5 Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 7 Apr 2016 11:25:33 +0200 Subject: ASoC: sti: helper functions for unip tdm slots configuration - sti_uniperiph_set_tdm_slot: store tdm slot config in unip context - sti_uniperiph_get_tdm_word_pos: configure unip tdm slots pos regs Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/sti_uniperif.c | 90 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sti/uniperif.h | 23 +++++++++++ 2 files changed, 113 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 39bcefe5eea0..0c2474c16c11 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -10,6 +10,96 @@ #include "uniperif.h" +/* + * User frame size shall be 2, 4, 6 or 8 32-bits words length + * (i.e. 8, 16, 24 or 32 bytes) + * This constraint comes from allowed values for + * UNIPERIF_I2S_FMT_NUM_CH register + */ +#define UNIPERIF_MAX_FRAME_SZ 0x20 +#define UNIPERIF_ALLOWED_FRAME_SZ (0x08 | 0x10 | 0x18 | UNIPERIF_MAX_FRAME_SZ) + +int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *uni = priv->dai_data.uni; + int i, frame_size, avail_slots; + + if (!UNIPERIF_TYPE_IS_TDM(uni)) { + dev_err(uni->dev, "cpu dai not in tdm mode\n"); + return -EINVAL; + } + + /* store info in unip context */ + uni->tdm_slot.slots = slots; + uni->tdm_slot.slot_width = slot_width; + /* unip is unidirectionnal */ + uni->tdm_slot.mask = (tx_mask != 0) ? tx_mask : rx_mask; + + /* number of available timeslots */ + for (i = 0, avail_slots = 0; i < uni->tdm_slot.slots; i++) { + if ((uni->tdm_slot.mask >> i) & 0x01) + avail_slots++; + } + uni->tdm_slot.avail_slots = avail_slots; + + /* frame size in bytes */ + frame_size = uni->tdm_slot.avail_slots * uni->tdm_slot.slot_width / 8; + + /* check frame size is allowed */ + if ((frame_size > UNIPERIF_MAX_FRAME_SZ) || + (frame_size & ~(int)UNIPERIF_ALLOWED_FRAME_SZ)) { + dev_err(uni->dev, "frame size not allowed: %d bytes\n", + frame_size); + return -EINVAL; + } + + return 0; +} + +int sti_uniperiph_get_tdm_word_pos(struct uniperif *uni, + unsigned int *word_pos) +{ + int slot_width = uni->tdm_slot.slot_width / 8; + int slots_num = uni->tdm_slot.slots; + unsigned int slots_mask = uni->tdm_slot.mask; + int i, j, k; + unsigned int word16_pos[4]; + + /* word16_pos: + * word16_pos[0] = WORDX_LSB + * word16_pos[1] = WORDX_MSB, + * word16_pos[2] = WORDX+1_LSB + * word16_pos[3] = WORDX+1_MSB + */ + + /* set unip word position */ + for (i = 0, j = 0, k = 0; (i < slots_num) && (k < WORD_MAX); i++) { + if ((slots_mask >> i) & 0x01) { + word16_pos[j] = i * slot_width; + + if (slot_width == 4) { + word16_pos[j + 1] = word16_pos[j] + 2; + j++; + } + j++; + + if (j > 3) { + word_pos[k] = word16_pos[1] | + (word16_pos[0] << 8) | + (word16_pos[3] << 16) | + (word16_pos[2] << 24); + j = 0; + k++; + } + } + } + + return 0; +} + /* * sti_uniperiph_dai_create_ctrl * This function is used to create Ctrl associated to DAI but also pcm device. diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 750eb5a07e6c..fb8e427754b6 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1270,6 +1270,14 @@ enum uniperif_iec958_encoding_mode { UNIPERIF_IEC958_ENCODING_MODE_ENCODED }; +enum uniperif_word_pos { + WORD_1_2, + WORD_3_4, + WORD_5_6, + WORD_7_8, + WORD_MAX +}; + struct uniperif_info { int id; /* instance value of the uniperipheral IP */ enum uniperif_type type; @@ -1281,6 +1289,13 @@ struct uniperif_iec958_settings { struct snd_aes_iec958 iec958; }; +struct dai_tdm_slot { + unsigned int mask; + int slots; + int slot_width; + unsigned int avail_slots; +}; + struct uniperif { /* System information */ struct uniperif_info *info; @@ -1317,6 +1332,7 @@ struct uniperif { /* dai properties */ unsigned int daifmt; + struct dai_tdm_slot tdm_slot; /* DAI callbacks */ const struct snd_soc_dai_ops *dai_ops; @@ -1373,4 +1389,11 @@ int sti_uniperiph_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai); +int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width); + +int sti_uniperiph_get_tdm_word_pos(struct uniperif *uni, + unsigned int *word_pos); + #endif -- cgit v1.2.3-58-ga151 From 72199787662612a7747248f8b56bc5d42694538f Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 7 Apr 2016 11:25:34 +0200 Subject: ASoC: sti: helper functions to fix tdm runtime params Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/sti_uniperif.c | 46 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sti/uniperif.h | 6 ++++++ 2 files changed, 52 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 0c2474c16c11..d49badec9e62 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -59,6 +59,52 @@ int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +int sti_uniperiph_fix_tdm_chan(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct uniperif *uni = rule->private; + struct snd_interval t; + + t.min = uni->tdm_slot.avail_slots; + t.max = uni->tdm_slot.avail_slots; + t.openmin = 0; + t.openmax = 0; + t.integer = 0; + + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + +int sti_uniperiph_fix_tdm_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct uniperif *uni = rule->private; + struct snd_mask *maskp = hw_param_mask(params, rule->var); + u64 format; + + switch (uni->tdm_slot.slot_width) { + case 16: + format = SNDRV_PCM_FMTBIT_S16_LE; + break; + case 32: + format = SNDRV_PCM_FMTBIT_S32_LE; + break; + default: + dev_err(uni->dev, "format not supported: %d bits\n", + uni->tdm_slot.slot_width); + return -EINVAL; + } + + maskp->bits[0] &= (u_int32_t)format; + maskp->bits[1] &= (u_int32_t)(format >> 32); + /* clear remaining indexes */ + memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX - 64) / 8); + + if (!maskp->bits[0] && !maskp->bits[1]) + return -EINVAL; + + return 0; +} + int sti_uniperiph_get_tdm_word_pos(struct uniperif *uni, unsigned int *word_pos) { diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index fb8e427754b6..17d5d1030741 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1396,4 +1396,10 @@ int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, int sti_uniperiph_get_tdm_word_pos(struct uniperif *uni, unsigned int *word_pos); +int sti_uniperiph_fix_tdm_chan(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule); + +int sti_uniperiph_fix_tdm_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule); + #endif -- cgit v1.2.3-58-ga151 From 8d8b1e2eddaef25ca0a18500dd9425638f1cfd02 Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 7 Apr 2016 11:25:35 +0200 Subject: ASoC: sti: unip player tdm mode here are the changes to enable player tdm mode: - When TDM_ENABLE is set to 1, the i2s format should be automatically configured. Unfortunately this is not the case (HW bug). Then, we shall force DATA_SIZE setting. - Compute the transfer size for tdm mode: transfer size = user frame size - Manage tdm slots configuration given in DT. - Don't use mclk-fs when unip in tdm mode; use tdm slot config to compute frame size and to set mclk rate. - Refine the hw param (channels & format) according to tdm slot config. Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/sti_uniperif.c | 8 ++- sound/soc/sti/uniperif.h | 11 ++++ sound/soc/sti/uniperif_player.c | 109 +++++++++++++++++++++++++++++++++------- 3 files changed, 110 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index d49badec9e62..488ef4ed8fba 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -181,10 +181,16 @@ int sti_uniperiph_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *uni = priv->dai_data.uni; struct snd_dmaengine_dai_dma_data *dma_data; int transfer_size; - transfer_size = params_channels(params) * UNIPERIF_FIFO_FRAMES; + if (uni->info->type == SND_ST_UNIPERIF_TYPE_TDM) + /* transfer size = user frame size (in 32-bits FIFO cell) */ + transfer_size = snd_soc_params_to_frame_size(params) / 32; + else + transfer_size = params_channels(params) * UNIPERIF_FIFO_FRAMES; dma_data = snd_soc_dai_get_dma_data(dai, substream); dma_data->maxburst = transfer_size; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 17d5d1030741..d0e24468478c 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1389,6 +1389,17 @@ int sti_uniperiph_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai); +static inline int sti_uniperiph_get_user_frame_size( + struct snd_pcm_runtime *runtime) +{ + return (runtime->channels * snd_pcm_format_width(runtime->format) / 8); +} + +static inline int sti_uniperiph_get_unip_tdm_frame_size(struct uniperif *uni) +{ + return (uni->tdm_slot.slots * uni->tdm_slot.slot_width / 8); +} + int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index ab4310a1615e..ff2d73597ce3 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -435,18 +435,11 @@ static int uni_player_prepare_pcm(struct uniperif *player, /* Force slot width to 32 in I2S mode (HW constraint) */ if ((player->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == - SND_SOC_DAIFMT_I2S) { + SND_SOC_DAIFMT_I2S) slot_width = 32; - } else { - switch (runtime->format) { - case SNDRV_PCM_FORMAT_S16_LE: - slot_width = 16; - break; - default: - slot_width = 32; - break; - } - } + else + slot_width = snd_pcm_format_width(runtime->format); + output_frame_size = slot_width * runtime->channels; clk_div = player->mclk / runtime->rate; @@ -521,7 +514,6 @@ static int uni_player_prepare_pcm(struct uniperif *player, SET_UNIPERIF_CONFIG_ONE_BIT_AUD_DISABLE(player); SET_UNIPERIF_I2S_FMT_ORDER_MSB(player); - SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(player); /* No iec958 formatting as outputting to DAC */ SET_UNIPERIF_CTRL_SPDIF_FMT_OFF(player); @@ -529,6 +521,55 @@ static int uni_player_prepare_pcm(struct uniperif *player, return 0; } +static int uni_player_prepare_tdm(struct uniperif *player, + struct snd_pcm_runtime *runtime) +{ + int tdm_frame_size; /* unip tdm frame size in bytes */ + int user_frame_size; /* user tdm frame size in bytes */ + /* default unip TDM_WORD_POS_X_Y */ + unsigned int word_pos[4] = { + 0x04060002, 0x0C0E080A, 0x14161012, 0x1C1E181A}; + int freq, ret; + + tdm_frame_size = + sti_uniperiph_get_unip_tdm_frame_size(player); + user_frame_size = + sti_uniperiph_get_user_frame_size(runtime); + + /* fix 16/0 format */ + SET_UNIPERIF_CONFIG_MEM_FMT_16_0(player); + SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(player); + + /* number of words inserted on the TDM line */ + SET_UNIPERIF_I2S_FMT_NUM_CH(player, user_frame_size / 4 / 2); + + SET_UNIPERIF_I2S_FMT_ORDER_MSB(player); + SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(player); + + /* Enable the tdm functionality */ + SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(player); + + /* number of 8 bits timeslots avail in unip tdm frame */ + SET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(player, tdm_frame_size); + + /* set the timeslot allocation for words in FIFO */ + sti_uniperiph_get_tdm_word_pos(player, word_pos); + SET_UNIPERIF_TDM_WORD_POS(player, 1_2, word_pos[WORD_1_2]); + SET_UNIPERIF_TDM_WORD_POS(player, 3_4, word_pos[WORD_3_4]); + SET_UNIPERIF_TDM_WORD_POS(player, 5_6, word_pos[WORD_5_6]); + SET_UNIPERIF_TDM_WORD_POS(player, 7_8, word_pos[WORD_7_8]); + + /* set unip clk rate (not done vai set_sysclk ops) */ + freq = runtime->rate * tdm_frame_size * 8; + mutex_lock(&player->ctrl_lock); + ret = uni_player_clk_set_rate(player, freq); + if (!ret) + player->mclk = freq; + mutex_unlock(&player->ctrl_lock); + + return 0; +} + /* * ALSA uniperipheral iec958 controls */ @@ -659,11 +700,29 @@ static int uni_player_startup(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; + int ret; + player->substream = substream; player->clk_adj = 0; - return 0; + if (!UNIPERIF_TYPE_IS_TDM(player)) + return 0; + + /* refine hw constraint in tdm mode */ + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + sti_uniperiph_fix_tdm_chan, + player, SNDRV_PCM_HW_PARAM_CHANNELS, + -1); + if (ret < 0) + return ret; + + return snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FORMAT, + sti_uniperiph_fix_tdm_format, + player, SNDRV_PCM_HW_PARAM_FORMAT, + -1); } static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id, @@ -673,7 +732,7 @@ static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id, struct uniperif *player = priv->dai_data.uni; int ret; - if (dir == SND_SOC_CLOCK_IN) + if (UNIPERIF_TYPE_IS_TDM(player) || (dir == SND_SOC_CLOCK_IN)) return 0; if (clk_id != 0) @@ -705,7 +764,13 @@ static int uni_player_prepare(struct snd_pcm_substream *substream, } /* Calculate transfer size (in fifo cells and bytes) for frame count */ - transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES; + if (player->info->type == SND_ST_UNIPERIF_TYPE_TDM) { + /* transfer size = user frame size (in 32 bits FIFO cell) */ + transfer_size = + sti_uniperiph_get_user_frame_size(runtime) / 4; + } else { + transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES; + } /* Calculate number of empty cells available before asserting DREQ */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) { @@ -739,6 +804,9 @@ static int uni_player_prepare(struct snd_pcm_substream *substream, case SND_ST_UNIPERIF_TYPE_SPDIF: ret = uni_player_prepare_iec958(player, runtime); break; + case SND_ST_UNIPERIF_TYPE_TDM: + ret = uni_player_prepare_tdm(player, runtime); + break; default: dev_err(player->dev, "invalid player type"); return -EINVAL; @@ -1008,6 +1076,8 @@ static int uni_player_parse_dt(struct platform_device *pdev, info->type = SND_ST_UNIPERIF_TYPE_PCM; else if (strcasecmp(mode, "spdif") == 0) info->type = SND_ST_UNIPERIF_TYPE_SPDIF; + else if (strcasecmp(mode, "tdm") == 0) + info->type = SND_ST_UNIPERIF_TYPE_TDM; else info->type = SND_ST_UNIPERIF_TYPE_NONE; @@ -1028,7 +1098,8 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = { .trigger = uni_player_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, - .set_sysclk = uni_player_set_sysclk + .set_sysclk = uni_player_set_sysclk, + .set_tdm_slot = sti_uniperiph_set_tdm_slot }; int uni_player_init(struct platform_device *pdev, @@ -1038,7 +1109,6 @@ int uni_player_init(struct platform_device *pdev, player->dev = &pdev->dev; player->state = UNIPERIF_STATE_STOPPED; - player->hw = &uni_player_pcm_hw; player->dai_ops = &uni_player_dai_ops; ret = uni_player_parse_dt(pdev, player); @@ -1048,6 +1118,11 @@ int uni_player_init(struct platform_device *pdev, return ret; } + if (UNIPERIF_TYPE_IS_TDM(player)) + player->hw = &uni_tdm_hw; + else + player->hw = &uni_player_pcm_hw; + /* Get uniperif resource */ player->clk = of_clk_get(pdev->dev.of_node, 0); if (IS_ERR(player->clk)) -- cgit v1.2.3-58-ga151 From 82d4eb91ab1912cb9e8751b9aa0875af2ae36db2 Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 7 Apr 2016 11:25:36 +0200 Subject: ASoC: sti: unip reader tdm mode Here are the changes to enable reader tdm mode: - When TDM_ENABLE is set to 1, the i2s format should be automatically configured. Unfortunately this is not the case (HW bug). Then, we shall force DATA_SIZE setting. - Compute the transfer size for tdm mode: transfer size = user frame size - Manage tdm slots configuration given in DT. - Refine the hw param (channels & format) according to tdm slot config. Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/uniperif_reader.c | 229 +++++++++++++++++++++++++++++----------- 1 file changed, 168 insertions(+), 61 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 8a0eb2050169..eb74a328c928 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -73,55 +73,10 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) return ret; } -static int uni_reader_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime, + struct uniperif *reader) { - struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); - struct uniperif *reader = priv->dai_data.uni; - struct snd_pcm_runtime *runtime = substream->runtime; - int transfer_size, trigger_limit; int slot_width; - int count = 10; - - /* The reader should be stopped */ - if (reader->state != UNIPERIF_STATE_STOPPED) { - dev_err(reader->dev, "%s: invalid reader state %d", __func__, - reader->state); - return -EINVAL; - } - - /* Calculate transfer size (in fifo cells and bytes) for frame count */ - transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES; - - /* Calculate number of empty cells available before asserting DREQ */ - if (reader->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) - trigger_limit = UNIPERIF_FIFO_SIZE - transfer_size; - else - /* - * Since SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0 - * FDMA_TRIGGER_LIMIT also controls when the state switches - * from OFF or STANDBY to AUDIO DATA. - */ - trigger_limit = transfer_size; - - /* Trigger limit must be an even number */ - if ((!trigger_limit % 2) || - (trigger_limit != 1 && transfer_size % 2) || - (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) { - dev_err(reader->dev, "invalid trigger limit %d", trigger_limit); - return -EINVAL; - } - - SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(reader, trigger_limit); - - switch (reader->daifmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_IB_IF: - case SND_SOC_DAIFMT_NB_IF: - SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader); - break; - default: - SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader); - } /* Force slot width to 32 in I2S mode */ if ((reader->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) @@ -173,6 +128,109 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream, return -EINVAL; } + /* Number of channels must be even */ + if ((runtime->channels % 2) || (runtime->channels < 2) || + (runtime->channels > 10)) { + dev_err(reader->dev, "%s: invalid nb of channels", __func__); + return -EINVAL; + } + + SET_UNIPERIF_I2S_FMT_NUM_CH(reader, runtime->channels / 2); + SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader); + + return 0; +} + +static int uni_reader_prepare_tdm(struct snd_pcm_runtime *runtime, + struct uniperif *reader) +{ + int frame_size; /* user tdm frame size in bytes */ + /* default unip TDM_WORD_POS_X_Y */ + unsigned int word_pos[4] = { + 0x04060002, 0x0C0E080A, 0x14161012, 0x1C1E181A}; + + frame_size = sti_uniperiph_get_user_frame_size(runtime); + + /* fix 16/0 format */ + SET_UNIPERIF_CONFIG_MEM_FMT_16_0(reader); + SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(reader); + + /* number of words inserted on the TDM line */ + SET_UNIPERIF_I2S_FMT_NUM_CH(reader, frame_size / 4 / 2); + + SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader); + SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(reader); + SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(reader); + + /* + * set the timeslots allocation for words in FIFO + * + * HW bug: (LSB word < MSB word) => this config is not possible + * So if we want (LSB word < MSB) word, then it shall be + * handled by user + */ + sti_uniperiph_get_tdm_word_pos(reader, word_pos); + SET_UNIPERIF_TDM_WORD_POS(reader, 1_2, word_pos[WORD_1_2]); + SET_UNIPERIF_TDM_WORD_POS(reader, 3_4, word_pos[WORD_3_4]); + SET_UNIPERIF_TDM_WORD_POS(reader, 5_6, word_pos[WORD_5_6]); + SET_UNIPERIF_TDM_WORD_POS(reader, 7_8, word_pos[WORD_7_8]); + + return 0; +} + +static int uni_reader_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *reader = priv->dai_data.uni; + struct snd_pcm_runtime *runtime = substream->runtime; + int transfer_size, trigger_limit, ret; + int count = 10; + + /* The reader should be stopped */ + if (reader->state != UNIPERIF_STATE_STOPPED) { + dev_err(reader->dev, "%s: invalid reader state %d", __func__, + reader->state); + return -EINVAL; + } + + /* Calculate transfer size (in fifo cells and bytes) for frame count */ + if (reader->info->type == SND_ST_UNIPERIF_TYPE_TDM) { + /* transfer size = unip frame size (in 32 bits FIFO cell) */ + transfer_size = + sti_uniperiph_get_user_frame_size(runtime) / 4; + } else { + transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES; + } + + /* Calculate number of empty cells available before asserting DREQ */ + if (reader->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) + trigger_limit = UNIPERIF_FIFO_SIZE - transfer_size; + else + /* + * Since SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0 + * FDMA_TRIGGER_LIMIT also controls when the state switches + * from OFF or STANDBY to AUDIO DATA. + */ + trigger_limit = transfer_size; + + /* Trigger limit must be an even number */ + if ((!trigger_limit % 2) || + (trigger_limit != 1 && transfer_size % 2) || + (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) { + dev_err(reader->dev, "invalid trigger limit %d", trigger_limit); + return -EINVAL; + } + + SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(reader, trigger_limit); + + if (UNIPERIF_TYPE_IS_TDM(reader)) + ret = uni_reader_prepare_tdm(runtime, reader); + else + ret = uni_reader_prepare_pcm(runtime, reader); + if (ret) + return ret; + switch (reader->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(reader); @@ -191,21 +249,26 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream, return -EINVAL; } - SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader); - - /* Data clocking (changing) on the rising edge */ - SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader); - - /* Number of channels must be even */ - - if ((runtime->channels % 2) || (runtime->channels < 2) || - (runtime->channels > 10)) { - dev_err(reader->dev, "%s: invalid nb of channels", __func__); - return -EINVAL; + /* Data clocking (changing) on the rising/falling edge */ + switch (reader->daifmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader); + break; + case SND_SOC_DAIFMT_NB_IF: + SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader); + break; + case SND_SOC_DAIFMT_IB_NF: + SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(reader); + break; + case SND_SOC_DAIFMT_IB_IF: + SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(reader); + break; } - SET_UNIPERIF_I2S_FMT_NUM_CH(reader, runtime->channels / 2); - /* Clear any pending interrupts */ SET_UNIPERIF_ITS_BCLR(reader, GET_UNIPERIF_ITS(reader)); @@ -293,6 +356,32 @@ static int uni_reader_trigger(struct snd_pcm_substream *substream, } } +static int uni_reader_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *reader = priv->dai_data.uni; + int ret; + + if (!UNIPERIF_TYPE_IS_TDM(reader)) + return 0; + + /* refine hw constraint in tdm mode */ + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + sti_uniperiph_fix_tdm_chan, + reader, SNDRV_PCM_HW_PARAM_CHANNELS, + -1); + if (ret < 0) + return ret; + + return snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FORMAT, + sti_uniperiph_fix_tdm_format, + reader, SNDRV_PCM_HW_PARAM_FORMAT, + -1); +} + static void uni_reader_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -310,6 +399,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev, { struct uniperif_info *info; struct device_node *node = pdev->dev.of_node; + const char *mode; /* Allocate memory for the info structure */ info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); @@ -322,6 +412,17 @@ static int uni_reader_parse_dt(struct platform_device *pdev, return -EINVAL; } + /* Read the device mode property */ + if (of_property_read_string(node, "st,mode", &mode)) { + dev_err(&pdev->dev, "uniperipheral mode not defined"); + return -EINVAL; + } + + if (strcasecmp(mode, "tdm") == 0) + info->type = SND_ST_UNIPERIF_TYPE_TDM; + else + info->type = SND_ST_UNIPERIF_TYPE_PCM; + /* Save the info structure */ reader->info = info; @@ -329,11 +430,13 @@ static int uni_reader_parse_dt(struct platform_device *pdev, } static const struct snd_soc_dai_ops uni_reader_dai_ops = { + .startup = uni_reader_startup, .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, .trigger = uni_reader_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, + .set_tdm_slot = sti_uniperiph_set_tdm_slot }; int uni_reader_init(struct platform_device *pdev, @@ -343,7 +446,6 @@ int uni_reader_init(struct platform_device *pdev, reader->dev = &pdev->dev; reader->state = UNIPERIF_STATE_STOPPED; - reader->hw = &uni_reader_pcm_hw; reader->dai_ops = &uni_reader_dai_ops; ret = uni_reader_parse_dt(pdev, reader); @@ -352,6 +454,11 @@ int uni_reader_init(struct platform_device *pdev, return ret; } + if (UNIPERIF_TYPE_IS_TDM(reader)) + reader->hw = &uni_tdm_hw; + else + reader->hw = &uni_reader_pcm_hw; + ret = devm_request_irq(&pdev->dev, reader->irq, uni_reader_irq_handler, IRQF_SHARED, dev_name(&pdev->dev), reader); -- cgit v1.2.3-58-ga151 From 3ee15cac90e168fdea497a168a2e79acb1c4e612 Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 14 Apr 2016 15:29:35 +0200 Subject: ASoC: sti: select player for I2S/TDM TX bus By default, player#0 is connected to I2S/TDM TX bus. This patch connects player#1 to I2S/TDM TX bus. Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 42 +++++++++++++++++++++++++++-------------- 2 files changed, 29 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index d0e24468478c..eb9933c62ad6 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1302,6 +1302,7 @@ struct uniperif { struct device *dev; int ver; /* IP version, used by register access macros */ struct regmap_field *clk_sel; + struct regmap_field *valid_sel; /* capabilities */ const struct snd_pcm_hardware *hw; diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index ff2d73597ce3..ee1c7c245bc7 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -21,7 +21,6 @@ /* sys config registers definitions */ #define SYS_CFG_AUDIO_GLUE 0xA4 -#define SYS_CFG_AUDI0_GLUE_PCM_CLKX 8 /* * Driver specific types. @@ -29,6 +28,7 @@ #define UNIPERIF_PLAYER_CLK_ADJ_MIN -999999 #define UNIPERIF_PLAYER_CLK_ADJ_MAX 1000000 +#define UNIPERIF_PLAYER_I2S_OUT 1 /* player id connected to I2S/TDM TX bus */ /* * Note: snd_pcm_hardware is linked to DMA controller but is declared here to @@ -1013,27 +1013,30 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream, player->substream = NULL; } -static int uni_player_parse_dt_clk_glue(struct platform_device *pdev, - struct uniperif *player) +static int uni_player_parse_dt_audio_glue(struct platform_device *pdev, + struct uniperif *player) { - int bit_offset; struct device_node *node = pdev->dev.of_node; struct regmap *regmap; - - bit_offset = SYS_CFG_AUDI0_GLUE_PCM_CLKX + player->info->id; + struct reg_field regfield[2] = { + /* PCM_CLK_SEL */ + REG_FIELD(SYS_CFG_AUDIO_GLUE, + 8 + player->info->id, + 8 + player->info->id), + /* PCMP_VALID_SEL */ + REG_FIELD(SYS_CFG_AUDIO_GLUE, 0, 1) + }; regmap = syscon_regmap_lookup_by_phandle(node, "st,syscfg"); - if (regmap) { - struct reg_field regfield = - REG_FIELD(SYS_CFG_AUDIO_GLUE, bit_offset, bit_offset); - - player->clk_sel = regmap_field_alloc(regmap, regfield); - } else { + if (!regmap) { dev_err(&pdev->dev, "sti-audio-clk-glue syscf not found\n"); return -EINVAL; } + player->clk_sel = regmap_field_alloc(regmap, regfield[0]); + player->valid_sel = regmap_field_alloc(regmap, regfield[1]); + return 0; } @@ -1084,8 +1087,8 @@ static int uni_player_parse_dt(struct platform_device *pdev, /* Save the info structure */ player->info = info; - /* Get the PCM_CLK_SEL bit from audio-glue-ctrl SoC register */ - if (uni_player_parse_dt_clk_glue(pdev, player)) + /* Get PCM_CLK_SEL & PCMP_VALID_SEL from audio-glue-ctrl SoC reg */ + if (uni_player_parse_dt_audio_glue(pdev, player)) return -EINVAL; return 0; @@ -1139,6 +1142,17 @@ int uni_player_init(struct platform_device *pdev, } } + /* connect to I2S/TDM TX bus */ + if (player->valid_sel && + (player->info->id == UNIPERIF_PLAYER_I2S_OUT)) { + ret = regmap_field_write(player->valid_sel, player->info->id); + if (ret) { + dev_err(player->dev, + "%s: unable to connect to tdm bus", __func__); + return ret; + } + } + ret = devm_request_irq(&pdev->dev, player->irq, uni_player_irq_handler, IRQF_SHARED, dev_name(&pdev->dev), player); -- cgit v1.2.3-58-ga151 From ec513886411e4fc47f98607a1bc79b72899710c4 Mon Sep 17 00:00:00 2001 From: Jeremy McDermond Date: Mon, 18 Apr 2016 17:24:04 -0700 Subject: ASoC: tlv320aic32x4: Change name of probe function The codec's probe function is named aic32x4_probe. This is going to conflict with later work to implement SPI support and separate out I2S into its own file. In line with other drivers in the tree, this function is renamed to aic32x4_codec_probe instead. Signed-off-by: Jeremy McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index f2d3191961e1..9d5b08b80d24 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -596,7 +596,7 @@ static struct snd_soc_dai_driver aic32x4_dai = { .symmetric_rates = 1, }; -static int aic32x4_probe(struct snd_soc_codec *codec) +static int aic32x4_codec_probe(struct snd_soc_codec *codec) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; @@ -655,7 +655,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { - .probe = aic32x4_probe, + .probe = aic32x4_codec_probe, .set_bias_level = aic32x4_set_bias_level, .suspend_bias_off = true, -- cgit v1.2.3-58-ga151 From 3bcfd222f6f0c8758f369ce0db23fa3287db59a6 Mon Sep 17 00:00:00 2001 From: Jeremy McDermond Date: Mon, 18 Apr 2016 17:24:05 -0700 Subject: ASoC: tlv320aic32x4: Break out I2C support into separate module To prepare for abstracting adding SPI support, the I2C pieces needs to be in its own moudle. This patch moves common probe code into aic32x4_probe and common removal code into aic32x4_remove. It also creates a static regmap config structure to be copied in the I2C specific driver. Signed-off-by: Jeremy McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 7 +++- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic32x4-i2c.c | 74 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 76 ++++++++++++------------------------ sound/soc/codecs/tlv320aic32x4.h | 7 ++++ 5 files changed, 113 insertions(+), 53 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic32x4-i2c.c (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..38d07c3f5a38 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -129,7 +129,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4 if I2C + select SND_SOC_TLV320AIC32X4_I2C if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -769,6 +769,11 @@ config SND_SOC_TLV320AIC31XX config SND_SOC_TLV320AIC32X4 tristate +config SND_SOC_TLV320AIC32X4_I2C + tristate + depends on I2C + select SND_SOC_TLV320AIC32X4 + config SND_SOC_TLV320AIC3X tristate "Texas Instruments TLV320AIC3x CODECs" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..7ce294ce3afd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -136,6 +136,7 @@ snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o +snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-ts3a227e-objs := ts3a227e.o @@ -342,6 +343,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o +obj-$(CONFIG_SND_SOC_TLV320AIC32X4_I2C) += snd-soc-tlv320aic32x4-i2c.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c new file mode 100644 index 000000000000..59606cf3008f --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4-i2c.c @@ -0,0 +1,74 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4-i2c.c + * + * Copyright 2011 NW Digital Radio + * + * Author: Jeremy McDermond + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include + +#include "tlv320aic32x4.h" + +static int aic32x4_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + struct regmap_config config; + + config = aic32x4_regmap_config; + config.reg_bits = 8; + config.val_bits = 8; + + regmap = devm_regmap_init_i2c(i2c, &config); + return aic32x4_probe(&i2c->dev, regmap); +} + +static int aic32x4_i2c_remove(struct i2c_client *i2c) +{ + return aic32x4_remove(&i2c->dev); +} + +static const struct i2c_device_id aic32x4_i2c_id[] = { + { "tlv320aic32x4", 0 }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); + +static const struct of_device_id aic32x4_of_id[] = { + { .compatible = "ti,tlv320aic32x4", }, + { /* senitel */ } +}; +MODULE_DEVICE_TABLE(of, aic32x4_of_id); + +static struct i2c_driver aic32x4_i2c_driver = { + .driver = { + .name = "tlv320aic32x4", + .of_match_table = aic32x4_of_id, + }, + .probe = aic32x4_i2c_probe, + .remove = aic32x4_i2c_remove, + .id_table = aic32x4_i2c_id, +}; + +module_i2c_driver(aic32x4_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver I2C"); +MODULE_AUTHOR("Jeremy McDermond "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 9d5b08b80d24..c6b6d551f4fe 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -30,7 +30,6 @@ #include #include #include -#include #include #include #include @@ -287,14 +286,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = { }, }; -static const struct regmap_config aic32x4_regmap = { - .reg_bits = 8, - .val_bits = 8, - +const struct regmap_config aic32x4_regmap_config = { .max_register = AIC32X4_RMICPGAVOL, .ranges = aic32x4_regmap_pages, .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), }; +EXPORT_SYMBOL(aic32x4_regmap_config); static inline int aic32x4_get_divs(int mclk, int rate) { @@ -777,24 +774,22 @@ error_ldo: return ret; } -static int aic32x4_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +int aic32x4_probe(struct device *dev, struct regmap *regmap) { - struct aic32x4_pdata *pdata = i2c->dev.platform_data; struct aic32x4_priv *aic32x4; - struct device_node *np = i2c->dev.of_node; + struct aic32x4_pdata *pdata = dev->platform_data; + struct device_node *np = dev->of_node; int ret; - aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv), + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + aic32x4 = devm_kzalloc(dev, sizeof(struct aic32x4_priv), GFP_KERNEL); if (aic32x4 == NULL) return -ENOMEM; - aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap); - if (IS_ERR(aic32x4->regmap)) - return PTR_ERR(aic32x4->regmap); - - i2c_set_clientdata(i2c, aic32x4); + dev_set_drvdata(dev, aic32x4); if (pdata) { aic32x4->power_cfg = pdata->power_cfg; @@ -804,7 +799,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, } else if (np) { ret = aic32x4_parse_dt(aic32x4, np); if (ret) { - dev_err(&i2c->dev, "Failed to parse DT node\n"); + dev_err(dev, "Failed to parse DT node\n"); return ret; } } else { @@ -814,71 +809,48 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } - aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk"); + aic32x4->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(aic32x4->mclk)) { - dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); + dev_err(dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); return PTR_ERR(aic32x4->mclk); } if (gpio_is_valid(aic32x4->rstn_gpio)) { - ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, + ret = devm_gpio_request_one(dev, aic32x4->rstn_gpio, GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); if (ret != 0) return ret; } - ret = aic32x4_setup_regulators(&i2c->dev, aic32x4); + ret = aic32x4_setup_regulators(dev, aic32x4); if (ret) { - dev_err(&i2c->dev, "Failed to setup regulators\n"); + dev_err(dev, "Failed to setup regulators\n"); return ret; } - ret = snd_soc_register_codec(&i2c->dev, + ret = snd_soc_register_codec(dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); if (ret) { - dev_err(&i2c->dev, "Failed to register codec\n"); + dev_err(dev, "Failed to register codec\n"); aic32x4_disable_regulators(aic32x4); return ret; } - i2c_set_clientdata(i2c, aic32x4); - return 0; } +EXPORT_SYMBOL(aic32x4_probe); -static int aic32x4_i2c_remove(struct i2c_client *client) +int aic32x4_remove(struct device *dev) { - struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client); + struct aic32x4_priv *aic32x4 = dev_get_drvdata(dev); aic32x4_disable_regulators(aic32x4); - snd_soc_unregister_codec(&client->dev); + snd_soc_unregister_codec(dev); + return 0; } - -static const struct i2c_device_id aic32x4_i2c_id[] = { - { "tlv320aic32x4", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); - -static const struct of_device_id aic32x4_of_id[] = { - { .compatible = "ti,tlv320aic32x4", }, - { /* senitel */ } -}; -MODULE_DEVICE_TABLE(of, aic32x4_of_id); - -static struct i2c_driver aic32x4_i2c_driver = { - .driver = { - .name = "tlv320aic32x4", - .of_match_table = aic32x4_of_id, - }, - .probe = aic32x4_i2c_probe, - .remove = aic32x4_i2c_remove, - .id_table = aic32x4_i2c_id, -}; - -module_i2c_driver(aic32x4_i2c_driver); +EXPORT_SYMBOL(aic32x4_remove); MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); MODULE_AUTHOR("Javier Martin "); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index 995f033a855d..a197dd51addc 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -10,6 +10,13 @@ #ifndef _TLV320AIC32X4_H #define _TLV320AIC32X4_H +struct device; +struct regmap_config; + +extern const struct regmap_config aic32x4_regmap_config; +int aic32x4_probe(struct device *dev, struct regmap *regmap); +int aic32x4_remove(struct device *dev); + /* tlv320aic32x4 register space (in decimal to match datasheet) */ #define AIC32X4_PAGE1 128 -- cgit v1.2.3-58-ga151 From 125bc681bc5d8183594e30492f0345a61ab3cc94 Mon Sep 17 00:00:00 2001 From: Jeremy McDermond Date: Mon, 18 Apr 2016 17:24:06 -0700 Subject: ASoC: tlv320aic32x4: Add SPI support Add support for running the tlv32x4 control channel over SPI rather than I2C. Signed-off-by: Jeremy McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 +++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic32x4-spi.c | 76 ++++++++++++++++++++++++++++++++++++ 3 files changed, 84 insertions(+) create mode 100644 sound/soc/codecs/tlv320aic32x4-spi.c (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 38d07c3f5a38..3cc56e7d6e78 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -130,6 +130,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C select SND_SOC_TLV320AIC32X4_I2C if I2C + select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -774,6 +775,11 @@ config SND_SOC_TLV320AIC32X4_I2C depends on I2C select SND_SOC_TLV320AIC32X4 +config SND_SOC_TLV320AIC32X4_SPI + tristate + depends on SPI_MASTER + select SND_SOC_TLV320AIC32X4 + config SND_SOC_TLV320AIC3X tristate "Texas Instruments TLV320AIC3x CODECs" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ce294ce3afd..6b59bdb20aa1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -137,6 +137,7 @@ snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o +snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-ts3a227e-objs := ts3a227e.o @@ -344,6 +345,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4_I2C) += snd-soc-tlv320aic32x4-i2c.o +obj-$(CONFIG_SND_SOC_TLV320AIC32X4_SPI) += snd-soc-tlv320aic32x4-spi.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c new file mode 100644 index 000000000000..724fcdd491b2 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4-spi.c @@ -0,0 +1,76 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4-spi.c + * + * Copyright 2011 NW Digital Radio + * + * Author: Jeremy McDermond + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include + +#include "tlv320aic32x4.h" + +static int aic32x4_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + struct regmap_config config; + + config = aic32x4_regmap_config; + config.reg_bits = 7; + config.pad_bits = 1; + config.val_bits = 8; + config.read_flag_mask = 0x01; + + regmap = devm_regmap_init_spi(spi, &config); + return aic32x4_probe(&spi->dev, regmap); +} + +static int aic32x4_spi_remove(struct spi_device *spi) +{ + return aic32x4_remove(&spi->dev); +} + +static const struct spi_device_id aic32x4_spi_id[] = { + { "tlv320aic32x4", 0 }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(spi, aic32x4_spi_id); + +static const struct of_device_id aic32x4_of_id[] = { + { .compatible = "ti,tlv320aic32x4", }, + { /* senitel */ } +}; +MODULE_DEVICE_TABLE(of, aic32x4_of_id); + +static struct spi_driver aic32x4_spi_driver = { + .driver = { + .name = "tlv320aic32x4", + .owner = THIS_MODULE, + .of_match_table = aic32x4_of_id, + }, + .probe = aic32x4_spi_probe, + .remove = aic32x4_spi_remove, + .id_table = aic32x4_spi_id, +}; + +module_spi_driver(aic32x4_spi_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver SPI"); +MODULE_AUTHOR("Jeremy McDermond "); +MODULE_LICENSE("GPL"); -- cgit v1.2.3-58-ga151 From f5cc17720b61af382a053d49a4a14c3d14857a5b Mon Sep 17 00:00:00 2001 From: Bastien Nocera Date: Tue, 19 Apr 2016 18:00:20 +0200 Subject: ASoC: tlv320aix31xx: Add ACPI match for Lenovo 100S The Lenovo 100S netbook has a codec controller for which there is a driver, but doesn't know how to access the device. This adds the necessary ACPI table for the driver to find the device. Device (TTLV) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "10TI3100") // _HID: Hardware ID Name (_CID, "10TI3100") // _CID: Compatible ID Name (_DDN, "TI TLV320AIC3100 Codec Controller ") // _DDN: DOS Device Name Name (_UID, One) // _UID: Unique ID Signed-off-by: Bastien Nocera Tested-by: Jan Schmidt Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index ee4def4f819f..3c5e1df01c19 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -1280,10 +1281,19 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); +#ifdef CONFIG_ACPI +static const struct acpi_device_id aic31xx_acpi_match[] = { + { "10TI3100", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match); +#endif + static struct i2c_driver aic31xx_i2c_driver = { .driver = { .name = "tlv320aic31xx-codec", .of_match_table = of_match_ptr(tlv320aic31xx_of_match), + .acpi_match_table = ACPI_PTR(aic31xx_acpi_match), }, .probe = aic31xx_i2c_probe, .remove = aic31xx_i2c_remove, -- cgit v1.2.3-58-ga151 From 041f9d336f28d3a45b31799bb8b5b2e1fa322321 Mon Sep 17 00:00:00 2001 From: Jeremy McDermond Date: Tue, 19 Apr 2016 09:59:04 -0700 Subject: ASoC: tlv320aic32x4: Add 96k sample rate The TLV320AIC32x4 series supports 96ksps rates in hardware. This patch adds the necessary PLL divider values and clock settings to the table to make 96ksps work. Signed-off-by: Jeremy McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c6b6d551f4fe..2f8480c93b3c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -159,7 +159,10 @@ static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 48k rate */ {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, - {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}, + + /* 96k rate */ + {AIC32X4_FREQ_25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1}, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -564,7 +567,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000 #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -- cgit v1.2.3-58-ga151 From 2ebdf684082fa9ad924df1b2f80653920c7ca097 Mon Sep 17 00:00:00 2001 From: Stephen Boyd Date: Tue, 19 Apr 2016 18:08:00 -0700 Subject: ASoC: rsnd: Remove CLK_IS_ROOT This flag is a no-op now (see commit 47b0eeb3dc8a "clk: Deprecate CLK_IS_ROOT", 2016-02-02) so remove it. Signed-off-by: Stephen Boyd Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 606399de684d..49354d17ea55 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -492,9 +492,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ if (!count) { clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, req_rate); + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); @@ -506,9 +504,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, else { for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->onecell.clks = adg->clkout; -- cgit v1.2.3-58-ga151 From b84fff5afb16627a8973256992f3951ac3e90d84 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Tue, 19 Apr 2016 13:12:43 +0800 Subject: ASoC: topology: Set the link ID when creating a FE DAI link Topology will set the link's generic id when creating a FE link. Device drivers can check the id for link specific initialization. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 2 +- sound/soc/soc-topology.c | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index db86b447b515..e4701a3c6331 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -423,7 +423,7 @@ struct snd_soc_tplg_pcm { __le32 size; /* in bytes of this structure */ char pcm_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - __le32 pcm_id; /* unique ID - used to match */ + __le32 pcm_id; /* unique ID - used to match with DAI link */ __le32 dai_id; /* unique ID - used to match */ __le32 playback; /* supports playback mode */ __le32 capture; /* supports capture mode */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 1cf94d7fb9f4..bdbfcef4c319 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1598,6 +1598,7 @@ static int soc_tplg_link_create(struct soc_tplg *tplg, link->name = pcm->pcm_name; link->stream_name = pcm->pcm_name; + link->id = pcm->pcm_id; /* pass control to component driver for optional further init */ ret = soc_tplg_dai_link_load(tplg, link); -- cgit v1.2.3-58-ga151 From 67d1c21e37301ca3cea3705951950ce21f2723e1 Mon Sep 17 00:00:00 2001 From: Guneshwor Singh Date: Tue, 19 Apr 2016 13:12:50 +0800 Subject: ASoC: topology: Set CPU DAI name and enable DPCM by default for FE link When creating a FE link, the cpu_dai_name will come from topology and dpcm will be enabled by default. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index bdbfcef4c319..ca5f82885031 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1586,6 +1586,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, return snd_soc_register_dai(tplg->comp, dai_drv); } +/* create the FE DAI link */ static int soc_tplg_link_create(struct soc_tplg *tplg, struct snd_soc_tplg_pcm *pcm) { @@ -1600,6 +1601,15 @@ static int soc_tplg_link_create(struct soc_tplg *tplg, link->stream_name = pcm->pcm_name; link->id = pcm->pcm_id; + link->cpu_dai_name = pcm->dai_name; + link->codec_name = "snd-soc-dummy"; + link->codec_dai_name = "snd-soc-dummy-dai"; + + /* enable DPCM */ + link->dynamic = 1; + link->dpcm_playback = pcm->playback; + link->dpcm_capture = pcm->capture; + /* pass control to component driver for optional further init */ ret = soc_tplg_dai_link_load(tplg, link); if (ret < 0) { -- cgit v1.2.3-58-ga151 From 20d2cecbb7b9c35235c97f0dfa520525c28f8841 Mon Sep 17 00:00:00 2001 From: Jeremy McDermond Date: Wed, 20 Apr 2016 11:39:11 -0700 Subject: ASoC: tlv320aic32x4: Implement resistors on input pins The input pins of the aic3204 have resistors inline with them. The current code assumes that you want a 10k resistor inline with your inputs and implements it as a simple switch. This patch creates an enum for each pin and allows you to switch between not connected, 10k, 20k and 40k ohm values. This more closely models the acutal aic3204 part. These pin settings are documented in TI's SLAA557 pages 135 and 136 (http://www.ti.com/lit/ml/slaa557/slaa557.pdf). Signed-off-by: Jeremy McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 2f8480c93b3c..621f4210cd27 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -183,16 +183,34 @@ static const struct snd_kcontrol_new lor_output_mixer_controls[] = { SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0), }; +static const char * const resistor_text[] = { + "Off", "10 kOhm", "20 kOhm", "40 kOhm", +}; + +static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, + resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, + resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, + resistor_text); + static const struct snd_kcontrol_new left_input_mixer_controls[] = { - SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0), - SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0), - SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0), + SOC_DAPM_ENUM("IN1_L P Switch", in1l_lpga_p_enum), + SOC_DAPM_ENUM("IN2_L P Switch", in2l_lpga_p_enum), + SOC_DAPM_ENUM("IN3_L P Switch", in3l_lpga_p_enum), }; +static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, + resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, + resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, + resistor_text); + static const struct snd_kcontrol_new right_input_mixer_controls[] = { - SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0), - SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0), - SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0), + SOC_DAPM_ENUM("IN1_R P Switch", in1r_rpga_p_enum), + SOC_DAPM_ENUM("IN2_R P Switch", in2r_rpga_p_enum), + SOC_DAPM_ENUM("IN3_R P Switch", in3r_rpga_p_enum), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { -- cgit v1.2.3-58-ga151 From 13a06ed55dba0ae3f983ef3c4ea70fc42066e1b5 Mon Sep 17 00:00:00 2001 From: Jeremy McDermond Date: Wed, 20 Apr 2016 11:39:12 -0700 Subject: ASoC: tlv320aic32x4: Add additional input pins The input mixers support routing the IN1_R pin to the Left PGA and the IN2_L pin to the Right PGA. This patch allows for those routings. Signed-off-by: Jeremy McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 621f4210cd27..0eb8acc8cd66 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -193,11 +193,14 @@ static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, resistor_text); static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, + resistor_text); static const struct snd_kcontrol_new left_input_mixer_controls[] = { SOC_DAPM_ENUM("IN1_L P Switch", in1l_lpga_p_enum), SOC_DAPM_ENUM("IN2_L P Switch", in2l_lpga_p_enum), SOC_DAPM_ENUM("IN3_L P Switch", in3l_lpga_p_enum), + SOC_DAPM_ENUM("IN1_R P Switch", in1r_lpga_p_enum), }; static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, @@ -206,11 +209,14 @@ static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, resistor_text); static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, + resistor_text); static const struct snd_kcontrol_new right_input_mixer_controls[] = { SOC_DAPM_ENUM("IN1_R P Switch", in1r_rpga_p_enum), SOC_DAPM_ENUM("IN2_R P Switch", in2r_rpga_p_enum), SOC_DAPM_ENUM("IN3_R P Switch", in3r_rpga_p_enum), + SOC_DAPM_ENUM("IN2_L P Switch", in2l_rpga_p_enum), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { @@ -285,6 +291,7 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { {"Left Input Mixer", "IN1_L P Switch", "IN1_L"}, {"Left Input Mixer", "IN2_L P Switch", "IN2_L"}, {"Left Input Mixer", "IN3_L P Switch", "IN3_L"}, + {"Left Input Mixer", "IN1_R P Switch", "IN1_R"}, {"Left ADC", NULL, "Left Input Mixer"}, @@ -292,6 +299,7 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { {"Right Input Mixer", "IN1_R P Switch", "IN1_R"}, {"Right Input Mixer", "IN2_R P Switch", "IN2_R"}, {"Right Input Mixer", "IN3_R P Switch", "IN3_R"}, + {"Right Input Mixer", "IN2_L P Switch", "IN2_L"}, {"Right ADC", NULL, "Right Input Mixer"}, }; -- cgit v1.2.3-58-ga151 From d349caeb05104ef01392abc6c7cfc8ab516c7be4 Mon Sep 17 00:00:00 2001 From: PC Liao Date: Thu, 21 Apr 2016 19:38:14 +0800 Subject: ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. Signed-off-by: PC Liao Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 10 +++++ sound/soc/mediatek/mt8173-rt5650.c | 50 ++++++++++++++++++++-- 2 files changed, 57 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef1714d..5bfa6b60530b 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,21 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional subnodes: +- codec-capture : the subnode of rt5650 codec capture +Required codec-capture subnode properties: +- sound-dai: audio codec dai name on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + codec-capture { + sound-dai = <&rt5650 1>; + }; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1b7f1c..a27a6673dbe3 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,29 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_capture_dai = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_capture_dai, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_capture_dai, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_warn(card->dev, + "Only one dai codec found in DTS, enabled rt5645 AD filter\n"); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +127,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +171,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +199,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + struct device_node *np; + const char *codec_capture_dai; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +223,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (of_find_node_by_name(platform_node, "codec-capture")) { + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (!np) { + dev_err(&pdev->dev, + "%s: Can't find codec-capture DT node\n", + __func__); + return -EINVAL; + } + ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + if (ret < 0) { + dev_err(&pdev->dev, + "%s codec_capture_dai name fail %d\n", + __func__, ret); + return ret; + } + mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- cgit v1.2.3-58-ga151 From c0133e3b0265341e7d62e150df18709af33c3a30 Mon Sep 17 00:00:00 2001 From: Koro Chen Date: Wed, 20 Apr 2016 10:59:56 +0200 Subject: ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver This creates pcmC0D2p for the HDMI playback in the same card. Signed-off-by: Koro Chen Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 ++-- sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 27 ++++++++++++++++++++++ 3 files changed, 31 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt index f205ce9e31dd..ac28cdb4910e 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt @@ -1,15 +1,16 @@ -MT8173 with RT5650 RT5676 CODECS +MT8173 with RT5650 RT5676 CODECS and HDMI via I2S Required properties: - compatible : "mediatek,mt8173-rt5650-rt5676" - mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs + and of the hdmi encoder node - mediatek,platform: the phandle of MT8173 ASoC platform Example: sound { compatible = "mediatek,mt8173-rt5650-rt5676"; - mediatek,audio-codec = <&rt5650 &rt5676>; + mediatek,audio-codec = <&rt5650 &rt5676 &hdmi0>; mediatek,platform = <&afe>; }; diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index f7e789e97fbc..3abf51c07851 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_MT8173_RT5650_RT5676 depends on SND_SOC_MEDIATEK && I2C select SND_SOC_RT5645 select SND_SOC_RT5677 + select SND_SOC_HDMI_CODEC help This adds ASoC driver for Mediatek MT8173 boards with the RT5650 and RT5676 codecs. diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index 5c4c58c69c51..bb593926c62d 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -134,7 +134,9 @@ static struct snd_soc_dai_link_component mt8173_rt5650_rt5676_codecs[] = { enum { DAI_LINK_PLAYBACK, DAI_LINK_CAPTURE, + DAI_LINK_HDMI, DAI_LINK_CODEC_I2S, + DAI_LINK_HDMI_I2S, DAI_LINK_INTERCODEC }; @@ -161,6 +163,16 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .dynamic = 1, .dpcm_capture = 1, }, + [DAI_LINK_HDMI] = { + .name = "HDMI", + .stream_name = "HDMI PCM", + .cpu_dai_name = "HDMI", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, /* Back End DAI links */ [DAI_LINK_CODEC_I2S] = { @@ -177,6 +189,13 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .dpcm_playback = 1, .dpcm_capture = 1, }, + [DAI_LINK_HDMI_I2S] = { + .name = "HDMI BE", + .cpu_dai_name = "HDMIO", + .no_pcm = 1, + .codec_dai_name = "i2s-hifi", + .dpcm_playback = 1, + }, /* rt5676 <-> rt5650 intercodec link: Sets rt5676 I2S2 as master */ [DAI_LINK_INTERCODEC] = { .name = "rt5650_rt5676 intercodec", @@ -251,6 +270,14 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) mt8173_rt5650_rt5676_dais[DAI_LINK_INTERCODEC].codec_of_node = mt8173_rt5650_rt5676_codecs[1].of_node; + mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 2); + if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- cgit v1.2.3-58-ga151 From 54aba08f13fab00f31938993c9f7a9b0cd54f666 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 22 Apr 2016 09:09:14 +0000 Subject: ASoC: tidyup alphabetical order for SND_SOC_Bxx Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 8 ++++---- sound/soc/codecs/Makefile | 4 ++-- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..f7f64bd019a5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK5386 select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C + select SND_SOC_BT_SCO select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C select SND_SOC_CS42L51_I2C if I2C @@ -64,7 +65,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DMIC - select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_GTM601 @@ -365,6 +365,9 @@ config SND_SOC_ALC5623 config SND_SOC_ALC5632 tristate +config SND_SOC_BT_SCO + tristate + config SND_SOC_CQ0093VC tristate @@ -471,9 +474,6 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_BT_SCO - tristate - config SND_SOC_DMIC tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..12a1fe22eaa1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -32,6 +32,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs42l51-objs := cs42l51.o @@ -55,7 +56,6 @@ snd-soc-da7218-objs := da7218.o snd-soc-da7219-objs := da7219.o da7219-aad.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o @@ -241,6 +241,7 @@ obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o @@ -264,7 +265,6 @@ obj-$(CONFIG_SND_SOC_DA7218) += snd-soc-da7218.o obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o -- cgit v1.2.3-58-ga151 From 3c9e014c442caefa14c71494ca4473121007f60f Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 26 Apr 2016 18:07:10 +0800 Subject: ASoC: rt298: reset AD dilter is there is no MCLK rt298 need to reset AD filter and the ADC settings will take effort. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt298.c | 20 ++++++++++++++++++++ sound/soc/codecs/rt298.h | 2 ++ 2 files changed, 22 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index f0e6c06e89ac..178b1cc22e05 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -481,6 +481,26 @@ static int rt298_adc_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), 0x7080, 0x7000); + /* If MCLK doesn't exist, reset AD filter */ + if (!(snd_soc_read(codec, RT298_VAD_CTRL) & 0x200)) { + pr_info("NO MCLK\n"); + switch (nid) { + case RT298_ADC_IN1: + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x2, 0x2); + mdelay(10); + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x2, 0x0); + break; + case RT298_ADC_IN2: + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x4, 0x4); + mdelay(10); + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x4, 0x0); + break; + } + } break; case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(codec, diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h index d66f8847b676..3638f3d61209 100644 --- a/sound/soc/codecs/rt298.h +++ b/sound/soc/codecs/rt298.h @@ -137,6 +137,7 @@ #define RT298_A_BIAS_CTRL2 0x02 #define RT298_POWER_CTRL1 0x03 #define RT298_A_BIAS_CTRL3 0x04 +#define RT298_D_FILTER_CTRL 0x05 #define RT298_POWER_CTRL2 0x08 #define RT298_I2S_CTRL1 0x09 #define RT298_I2S_CTRL2 0x0a @@ -148,6 +149,7 @@ #define RT298_IRQ_CTRL 0x33 #define RT298_WIND_FILTER_CTRL 0x46 #define RT298_PLL_CTRL1 0x49 +#define RT298_VAD_CTRL 0x4e #define RT298_CBJ_CTRL1 0x4f #define RT298_CBJ_CTRL2 0x50 #define RT298_PLL_CTRL 0x63 -- cgit v1.2.3-58-ga151 From 9ff49ce475cfeaf486321a2db8132a9500740faa Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 26 Apr 2016 18:07:11 +0800 Subject: ASoC: rt298: fix capture doesn't work at some cases RT298_CBJ_CTRL1(0x4f) bit 10 is needed for headset capture. It will be turned off when "VREF" widget is on and be turned on when bias level is ON. It is odd. And if "VREF" is turned on in bias level is ON, RT298_CBJ_CTRL1(0x4f) bit 10 will be turned off. This patch move the bit control from rt298_set_bias_level and rt298_vref_event to rt298_jack_detect. So it will be turned on once a jack is plugged in. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt298.c | 31 +++---------------------------- 1 file changed, 3 insertions(+), 28 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 178b1cc22e05..68cf8d5a174f 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -275,6 +275,8 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) } else { *mic = false; regmap_write(rt298->regmap, RT298_SET_MIC1, 0x20); + regmap_update_bits(rt298->regmap, + RT298_CBJ_CTRL1, 0x0400, 0x0000); } } else { regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf); @@ -539,30 +541,12 @@ static int rt298_mic1_event(struct snd_soc_dapm_widget *w, return 0; } -static int rt298_vref_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, - RT298_CBJ_CTRL1, 0x0400, 0x0000); - mdelay(50); - break; - default: - return 0; - } - - return 0; -} - static const struct snd_soc_dapm_widget rt298_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY_S("HV", 1, RT298_POWER_CTRL1, 12, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("VREF", RT298_POWER_CTRL1, - 0, 1, rt298_vref_event, SND_SOC_DAPM_PRE_PMU), + 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY_S("BG_MBIAS", 1, RT298_POWER_CTRL2, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT298_POWER_CTRL2, @@ -953,18 +937,9 @@ static int rt298_set_bias_level(struct snd_soc_codec *codec, } break; - case SND_SOC_BIAS_ON: - mdelay(30); - snd_soc_update_bits(codec, - RT298_CBJ_CTRL1, 0x0400, 0x0400); - - break; - case SND_SOC_BIAS_STANDBY: snd_soc_write(codec, RT298_SET_AUDIO_POWER, AC_PWRST_D3); - snd_soc_update_bits(codec, - RT298_CBJ_CTRL1, 0x0400, 0x0000); break; default: -- cgit v1.2.3-58-ga151 From 27becea06e73c96b825ecddd8b3a59642364514a Mon Sep 17 00:00:00 2001 From: PC Liao Date: Tue, 26 Apr 2016 14:30:18 +0800 Subject: ASoC: mediatek: HDMI audio LR channel swapped Because LRCK of TDM use High to Low as default setting, this patch changes the TDM setting to inverse LRCK. Signed-off-by: PC Liao Signed-off-by: Mark Brown --- sound/soc/mediatek/mtk-afe-pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index f1c58a2c12fb..2b5df2ef51a3 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -123,6 +123,7 @@ #define AFE_TDM_CON1_WLEN_32BIT (0x2 << 8) #define AFE_TDM_CON1_MSB_ALIGNED (0x1 << 4) #define AFE_TDM_CON1_1_BCK_DELAY (0x1 << 3) +#define AFE_TDM_CON1_LRCK_INV (0x1 << 2) #define AFE_TDM_CON1_BCK_INV (0x1 << 1) #define AFE_TDM_CON1_EN (0x1 << 0) @@ -449,6 +450,7 @@ static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream, runtime->rate * runtime->channels * 32); val = AFE_TDM_CON1_BCK_INV | + AFE_TDM_CON1_LRCK_INV | AFE_TDM_CON1_1_BCK_DELAY | AFE_TDM_CON1_MSB_ALIGNED | /* I2S mode */ AFE_TDM_CON1_WLEN_32BIT | -- cgit v1.2.3-58-ga151 From 989ff7754a27a1fcc0c3b2794c985b31811871a0 Mon Sep 17 00:00:00 2001 From: Jim Lodes Date: Mon, 25 Apr 2016 11:10:07 -0500 Subject: ASoC: omap-pcm: Initialize DMA configuration Initialize the dma_slave_config for PCM DMA transfers, instead of leaving it uninitialized. Keeps previous data on the stack from giving us invalid values in uninitialized members of the config structure. Signed-off-by: Jim Lodes Signed-off-by: J.D. Schroeder Acked-by: Jarkko Nikula Reviewed-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 6bb623a2a4df..eb81b9a2293a 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -82,6 +82,8 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct dma_chan *chan; int err = 0; + memset(&config, 0x00, sizeof(config)); + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); /* return if this is a bufferless transfer e.g. -- cgit v1.2.3-58-ga151 From bd023ada36a66fe3d2fde619926b49b9a0428133 Mon Sep 17 00:00:00 2001 From: Andreas Dannenberg Date: Tue, 26 Apr 2016 17:15:57 -0500 Subject: ASoC: add support for TAS5720 digital amplifier The Texas Instruments TAS5720L/M device is a high-efficiency mono Class-D audio power amplifier optimized for high transient power capability to use the dynamic power headroom of small loudspeakers. Its digital time division multiplexed (TDM) interface enables up to 16 devices to share the same bus. Signed-off-by: Andreas Dannenberg Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 620 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++++++ 4 files changed, 720 insertions(+) create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h (limited to 'sound/soc') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c011f076d58b..06d298b79b9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -124,6 +124,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS2552 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C + select SND_SOC_TAS5720 if I2C select SND_SOC_TFA9879 if I2C select SND_SOC_TLV320AIC23_I2C if I2C select SND_SOC_TLV320AIC23_SPI if SPI_MASTER @@ -740,6 +741,13 @@ config SND_SOC_TAS571X tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers" depends on I2C +config SND_SOC_TAS5720 + tristate "Texas Instruments TAS5720 Mono Audio amplifier" + depends on I2C + help + Enable support for Texas Instruments TAS5720L/M high-efficiency mono + Class-D audio power amplifiers. + config SND_SOC_TFA9879 tristate "NXP Semiconductors TFA9879 amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..83d352ede6fd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -129,6 +129,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o +snd-soc-tas5720-objs := tas5720.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o @@ -335,6 +336,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o +obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o obj-$(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/tas5720.c b/sound/soc/codecs/tas5720.c new file mode 100644 index 000000000000..f54fb46b77c2 --- /dev/null +++ b/sound/soc/codecs/tas5720.c @@ -0,0 +1,620 @@ +/* + * tas5720.c - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier + * + * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Andreas Dannenberg + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "tas5720.h" + +/* Define how often to check (and clear) the fault status register (in ms) */ +#define TAS5720_FAULT_CHECK_INTERVAL 200 + +static const char * const tas5720_supply_names[] = { + "dvdd", /* Digital power supply. Connect to 3.3-V supply. */ + "pvdd", /* Class-D amp and analog power supply (connected). */ +}; + +#define TAS5720_NUM_SUPPLIES ARRAY_SIZE(tas5720_supply_names) + +struct tas5720_data { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas5720_client; + struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES]; + struct delayed_work fault_check_work; + unsigned int last_fault; +}; + +static int tas5720_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int rate = params_rate(params); + bool ssz_ds; + int ret; + + switch (rate) { + case 44100: + case 48000: + ssz_ds = false; + break; + case 88200: + case 96000: + ssz_ds = true; + break; + default: + dev_err(codec->dev, "unsupported sample rate: %u\n", rate); + return -EINVAL; + } + + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG, + TAS5720_SSZ_DS, ssz_ds); + if (ret < 0) { + dev_err(codec->dev, "error setting sample rate: %d\n", ret); + return ret; + } + + return 0; +} + +static int tas5720_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 serial_format; + int ret; + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_vdbg(codec->dev, "DAI Format master is not found\n"); + return -EINVAL; + } + + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): + /* 1st data bit occur one BCLK cycle after the frame sync */ + serial_format = TAS5720_SAIF_I2S; + break; + case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF): + /* + * Note that although the TAS5720 does not have a dedicated DSP + * mode it doesn't care about the LRCLK duty cycle during TDM + * operation. Therefore we can use the device's I2S mode with + * its delaying of the 1st data bit to receive DSP_A formatted + * data. See device datasheet for additional details. + */ + serial_format = TAS5720_SAIF_I2S; + break; + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF): + /* + * Similar to DSP_A, we can use the fact that the TAS5720 does + * not care about the LRCLK duty cycle during TDM to receive + * DSP_B formatted data in LEFTJ mode (no delaying of the 1st + * data bit). + */ + serial_format = TAS5720_SAIF_LEFTJ; + break; + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): + /* No delay after the frame sync */ + serial_format = TAS5720_SAIF_LEFTJ; + break; + default: + dev_vdbg(codec->dev, "DAI Format is not found\n"); + return -EINVAL; + } + + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG, + TAS5720_SAIF_FORMAT_MASK, + serial_format); + if (ret < 0) { + dev_err(codec->dev, "error setting SAIF format: %d\n", ret); + return ret; + } + + return 0; +} + +static int tas5720_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_codec *codec = dai->codec; + unsigned int first_slot; + int ret; + + if (!tx_mask) { + dev_err(codec->dev, "tx masks must not be 0\n"); + return -EINVAL; + } + + /* + * Determine the first slot that is being requested. We will only + * use the first slot that is found since the TAS5720 is a mono + * amplifier. + */ + first_slot = __ffs(tx_mask); + + if (first_slot > 7) { + dev_err(codec->dev, "slot selection out of bounds (%u)\n", + first_slot); + return -EINVAL; + } + + /* Enable manual TDM slot selection (instead of I2C ID based) */ + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG, + TAS5720_TDM_CFG_SRC, TAS5720_TDM_CFG_SRC); + if (ret < 0) + goto error_snd_soc_update_bits; + + /* Configure the TDM slot to process audio from */ + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, + TAS5720_TDM_SLOT_SEL_MASK, first_slot); + if (ret < 0) + goto error_snd_soc_update_bits; + + return 0; + +error_snd_soc_update_bits: + dev_err(codec->dev, "error configuring TDM mode: %d\n", ret); + return ret; +} + +static int tas5720_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int ret; + + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, + TAS5720_MUTE, mute ? TAS5720_MUTE : 0); + if (ret < 0) { + dev_err(codec->dev, "error (un-)muting device: %d\n", ret); + return ret; + } + + return 0; +} + +static void tas5720_fault_check_work(struct work_struct *work) +{ + struct tas5720_data *tas5720 = container_of(work, struct tas5720_data, + fault_check_work.work); + struct device *dev = tas5720->codec->dev; + unsigned int curr_fault; + int ret; + + ret = regmap_read(tas5720->regmap, TAS5720_FAULT_REG, &curr_fault); + if (ret < 0) { + dev_err(dev, "failed to read FAULT register: %d\n", ret); + goto out; + } + + /* Check/handle all errors except SAIF clock errors */ + curr_fault &= TAS5720_OCE | TAS5720_DCE | TAS5720_OTE; + + /* + * Only flag errors once for a given occurrence. This is needed as + * the TAS5720 will take time clearing the fault condition internally + * during which we don't want to bombard the system with the same + * error message over and over. + */ + if ((curr_fault & TAS5720_OCE) && !(tas5720->last_fault & TAS5720_OCE)) + dev_crit(dev, "experienced an over current hardware fault\n"); + + if ((curr_fault & TAS5720_DCE) && !(tas5720->last_fault & TAS5720_DCE)) + dev_crit(dev, "experienced a DC detection fault\n"); + + if ((curr_fault & TAS5720_OTE) && !(tas5720->last_fault & TAS5720_OTE)) + dev_crit(dev, "experienced an over temperature fault\n"); + + /* Store current fault value so we can detect any changes next time */ + tas5720->last_fault = curr_fault; + + if (!curr_fault) + goto out; + + /* + * Periodically toggle SDZ (shutdown bit) H->L->H to clear any latching + * faults as long as a fault condition persists. Always going through + * the full sequence no matter the first return value to minimizes + * chances for the device to end up in shutdown mode. + */ + ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, 0); + if (ret < 0) + dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret); + + ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, TAS5720_SDZ); + if (ret < 0) + dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret); + +out: + /* Schedule the next fault check at the specified interval */ + schedule_delayed_work(&tas5720->fault_check_work, + msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL)); +} + +static int tas5720_codec_probe(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + unsigned int device_id; + int ret; + + tas5720->codec = codec; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret != 0) { + dev_err(codec->dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + ret = regmap_read(tas5720->regmap, TAS5720_DEVICE_ID_REG, &device_id); + if (ret < 0) { + dev_err(codec->dev, "failed to read device ID register: %d\n", + ret); + goto probe_fail; + } + + if (device_id != TAS5720_DEVICE_ID) { + dev_err(codec->dev, "wrong device ID. expected: %u read: %u\n", + TAS5720_DEVICE_ID, device_id); + ret = -ENODEV; + goto probe_fail; + } + + /* Set device to mute */ + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, + TAS5720_MUTE, TAS5720_MUTE); + if (ret < 0) + goto error_snd_soc_update_bits; + + /* + * Enter shutdown mode - our default when not playing audio - to + * minimize current consumption. On the TAS5720 there is no real down + * side doing so as all device registers are preserved and the wakeup + * of the codec is rather quick which we do using a dapm widget. + */ + ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, 0); + if (ret < 0) + goto error_snd_soc_update_bits; + + INIT_DELAYED_WORK(&tas5720->fault_check_work, tas5720_fault_check_work); + + return 0; + +error_snd_soc_update_bits: + dev_err(codec->dev, "error configuring device registers: %d\n", ret); + +probe_fail: + regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + return ret; +} + +static int tas5720_codec_remove(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + cancel_delayed_work_sync(&tas5720->fault_check_work); + + ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret < 0) + dev_err(codec->dev, "failed to disable supplies: %d\n", ret); + + return ret; +}; + +static int tas5720_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + if (event & SND_SOC_DAPM_POST_PMU) { + /* Take TAS5720 out of shutdown mode */ + ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, TAS5720_SDZ); + if (ret < 0) { + dev_err(codec->dev, "error waking codec: %d\n", ret); + return ret; + } + + /* + * Observe codec shutdown-to-active time. The datasheet only + * lists a nominal value however just use-it as-is without + * additional padding to minimize the delay introduced in + * starting to play audio (actually there is other setup done + * by the ASoC framework that will provide additional delays, + * so we should always be safe). + */ + msleep(25); + + /* Turn on TAS5720 periodic fault checking/handling */ + tas5720->last_fault = 0; + schedule_delayed_work(&tas5720->fault_check_work, + msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL)); + } else if (event & SND_SOC_DAPM_PRE_PMD) { + /* Disable TAS5720 periodic fault checking/handling */ + cancel_delayed_work_sync(&tas5720->fault_check_work); + + /* Place TAS5720 in shutdown mode to minimize current draw */ + ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, 0); + if (ret < 0) { + dev_err(codec->dev, "error shutting down codec: %d\n", + ret); + return ret; + } + } + + return 0; +} + +#ifdef CONFIG_PM +static int tas5720_suspend(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + regcache_cache_only(tas5720->regmap, true); + regcache_mark_dirty(tas5720->regmap); + + ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret < 0) + dev_err(codec->dev, "failed to disable supplies: %d\n", ret); + + return ret; +} + +static int tas5720_resume(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret < 0) { + dev_err(codec->dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(tas5720->regmap, false); + + ret = regcache_sync(tas5720->regmap); + if (ret < 0) { + dev_err(codec->dev, "failed to sync regcache: %d\n", ret); + return ret; + } + + return 0; +} +#else +#define tas5720_suspend NULL +#define tas5720_resume NULL +#endif + +static bool tas5720_is_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5720_DEVICE_ID_REG: + case TAS5720_FAULT_REG: + return true; + default: + return false; + } +} + +static const struct regmap_config tas5720_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS5720_MAX_REG, + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5720_is_volatile_reg, +}; + +/* + * DAC analog gain. There are four discrete values to select from, ranging + * from 19.2 dB to 26.3dB. + */ +static const DECLARE_TLV_DB_RANGE(dac_analog_tlv, + 0x0, 0x0, TLV_DB_SCALE_ITEM(1920, 0, 0), + 0x1, 0x1, TLV_DB_SCALE_ITEM(2070, 0, 0), + 0x2, 0x2, TLV_DB_SCALE_ITEM(2350, 0, 0), + 0x3, 0x3, TLV_DB_SCALE_ITEM(2630, 0, 0), +); + +/* + * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that + * setting the gain below -100 dB (register value <0x7) is effectively a MUTE + * as per device datasheet. + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0); + +static const struct snd_kcontrol_new tas5720_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv), + SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, + TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), +}; + +static const struct snd_soc_dapm_widget tas5720_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas5720_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("OUT") +}; + +static const struct snd_soc_dapm_route tas5720_audio_map[] = { + { "DAC", NULL, "DAC IN" }, + { "OUT", NULL, "DAC" }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas5720 = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + + .controls = tas5720_snd_controls, + .num_controls = ARRAY_SIZE(tas5720_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), +}; + +/* PCM rates supported by the TAS5720 driver */ +#define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +/* Formats supported by TAS5720 driver */ +#define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops tas5720_speaker_dai_ops = { + .hw_params = tas5720_hw_params, + .set_fmt = tas5720_set_dai_fmt, + .set_tdm_slot = tas5720_set_dai_tdm_slot, + .digital_mute = tas5720_mute, +}; + +/* + * TAS5720 DAI structure + * + * Note that were are advertising .playback.channels_max = 2 despite this being + * a mono amplifier. The reason for that is that some serial ports such as TI's + * McASP module have a minimum number of channels (2) that they can output. + * Advertising more channels than we have will allow us to interface with such + * a serial port without really any negative side effects as the TAS5720 will + * simply ignore any extra channel(s) asides from the one channel that is + * configured to be played back. + */ +static struct snd_soc_dai_driver tas5720_dai[] = { + { + .name = "tas5720-amplifier", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = TAS5720_RATES, + .formats = TAS5720_FORMATS, + }, + .ops = &tas5720_speaker_dai_ops, + }, +}; + +static int tas5720_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev = &client->dev; + struct tas5720_data *data; + int ret; + int i; + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->tas5720_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas5720_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(dev, "failed to allocate register map: %d\n", ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tas5720_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + dev_set_drvdata(dev, data); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas5720, + tas5720_dai, ARRAY_SIZE(tas5720_dai)); + if (ret < 0) { + dev_err(dev, "failed to register codec: %d\n", ret); + return ret; + } + + return 0; +} + +static int tas5720_remove(struct i2c_client *client) +{ + struct device *dev = &client->dev; + + snd_soc_unregister_codec(dev); + + return 0; +} + +static const struct i2c_device_id tas5720_id[] = { + { "tas5720", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5720_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas5720_of_match[] = { + { .compatible = "ti,tas5720", }, + { }, +}; +MODULE_DEVICE_TABLE(of, tas5720_of_match); +#endif + +static struct i2c_driver tas5720_i2c_driver = { + .driver = { + .name = "tas5720", + .of_match_table = of_match_ptr(tas5720_of_match), + }, + .probe = tas5720_probe, + .remove = tas5720_remove, + .id_table = tas5720_id, +}; + +module_i2c_driver(tas5720_i2c_driver); + +MODULE_AUTHOR("Andreas Dannenberg "); +MODULE_DESCRIPTION("TAS5720 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h new file mode 100644 index 000000000000..3d077c779b12 --- /dev/null +++ b/sound/soc/codecs/tas5720.h @@ -0,0 +1,90 @@ +/* + * tas5720.h - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier + * + * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Andreas Dannenberg + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __TAS5720_H__ +#define __TAS5720_H__ + +/* Register Address Map */ +#define TAS5720_DEVICE_ID_REG 0x00 +#define TAS5720_POWER_CTRL_REG 0x01 +#define TAS5720_DIGITAL_CTRL1_REG 0x02 +#define TAS5720_DIGITAL_CTRL2_REG 0x03 +#define TAS5720_VOLUME_CTRL_REG 0x04 +#define TAS5720_ANALOG_CTRL_REG 0x06 +#define TAS5720_FAULT_REG 0x08 +#define TAS5720_DIGITAL_CLIP2_REG 0x10 +#define TAS5720_DIGITAL_CLIP1_REG 0x11 +#define TAS5720_MAX_REG TAS5720_DIGITAL_CLIP1_REG + +/* TAS5720_DEVICE_ID_REG */ +#define TAS5720_DEVICE_ID 0x01 + +/* TAS5720_POWER_CTRL_REG */ +#define TAS5720_DIG_CLIP_MASK GENMASK(7, 2) +#define TAS5720_SLEEP BIT(1) +#define TAS5720_SDZ BIT(0) + +/* TAS5720_DIGITAL_CTRL1_REG */ +#define TAS5720_HPF_BYPASS BIT(7) +#define TAS5720_TDM_CFG_SRC BIT(6) +#define TAS5720_SSZ_DS BIT(3) +#define TAS5720_SAIF_RIGHTJ_24BIT (0x0) +#define TAS5720_SAIF_RIGHTJ_20BIT (0x1) +#define TAS5720_SAIF_RIGHTJ_18BIT (0x2) +#define TAS5720_SAIF_RIGHTJ_16BIT (0x3) +#define TAS5720_SAIF_I2S (0x4) +#define TAS5720_SAIF_LEFTJ (0x5) +#define TAS5720_SAIF_FORMAT_MASK GENMASK(2, 0) + +/* TAS5720_DIGITAL_CTRL2_REG */ +#define TAS5720_MUTE BIT(4) +#define TAS5720_TDM_SLOT_SEL_MASK GENMASK(2, 0) + +/* TAS5720_ANALOG_CTRL_REG */ +#define TAS5720_PWM_RATE_6_3_FSYNC (0x0 << 4) +#define TAS5720_PWM_RATE_8_4_FSYNC (0x1 << 4) +#define TAS5720_PWM_RATE_10_5_FSYNC (0x2 << 4) +#define TAS5720_PWM_RATE_12_6_FSYNC (0x3 << 4) +#define TAS5720_PWM_RATE_14_7_FSYNC (0x4 << 4) +#define TAS5720_PWM_RATE_16_8_FSYNC (0x5 << 4) +#define TAS5720_PWM_RATE_20_10_FSYNC (0x6 << 4) +#define TAS5720_PWM_RATE_24_12_FSYNC (0x7 << 4) +#define TAS5720_PWM_RATE_MASK GENMASK(6, 4) +#define TAS5720_ANALOG_GAIN_19_2DBV (0x0 << 2) +#define TAS5720_ANALOG_GAIN_20_7DBV (0x1 << 2) +#define TAS5720_ANALOG_GAIN_23_5DBV (0x2 << 2) +#define TAS5720_ANALOG_GAIN_26_3DBV (0x3 << 2) +#define TAS5720_ANALOG_GAIN_MASK GENMASK(3, 2) +#define TAS5720_ANALOG_GAIN_SHIFT (0x2) + +/* TAS5720_FAULT_REG */ +#define TAS5720_OC_THRESH_100PCT (0x0 << 4) +#define TAS5720_OC_THRESH_75PCT (0x1 << 4) +#define TAS5720_OC_THRESH_50PCT (0x2 << 4) +#define TAS5720_OC_THRESH_25PCT (0x3 << 4) +#define TAS5720_OC_THRESH_MASK GENMASK(5, 4) +#define TAS5720_CLKE BIT(3) +#define TAS5720_OCE BIT(2) +#define TAS5720_DCE BIT(1) +#define TAS5720_OTE BIT(0) +#define TAS5720_FAULT_MASK GENMASK(3, 0) + +/* TAS5720_DIGITAL_CLIP1_REG */ +#define TAS5720_CLIP1_MASK GENMASK(7, 2) +#define TAS5720_CLIP1_SHIFT (0x2) + +#endif /* __TAS5720_H__ */ -- cgit v1.2.3-58-ga151 From 7de76b621f77aba456f594e4621eca2e94a146f3 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 27 Apr 2016 14:52:38 +0800 Subject: ASoC: topology: Check failure to create a widget Stop loading topology info if error happens when creating a widget. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index ca5f82885031..0224a6458f3b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1500,9 +1500,11 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; ret = soc_tplg_dapm_widget_create(tplg, widget); - if (ret < 0) + if (ret < 0) { dev_err(tplg->dev, "ASoC: failed to load widget %s\n", widget->name); + return ret; + } } return 0; -- cgit v1.2.3-58-ga151 From 06eb49f72fa57f5a49acdf9f4af84d2d326513b3 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 27 Apr 2016 14:52:56 +0800 Subject: ASoC: topology: Check size mismatch of ABI objects before parsing If size mismatch of manifest, ABI headers or elements is found, stop parsing topology info and return the error. New fields may be append to the tail of ABI objects which will cause object size to increase. If user space and kernel use different versions of ABI, size mismatch will be detected here. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 31 +++++++++++++++++++++++++++++-- 1 file changed, 29 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0224a6458f3b..29ae3d3a0f8a 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1023,6 +1023,11 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; + if (control_hdr->size != sizeof(*control_hdr)) { + dev_err(tplg->dev, "ASoC: invalid control size\n"); + return -EINVAL; + } + switch (control_hdr->ops.info) { case SND_SOC_TPLG_CTL_VOLSW: case SND_SOC_TPLG_CTL_STROBE: @@ -1499,6 +1504,11 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; + if (widget->size != sizeof(*widget)) { + dev_err(tplg->dev, "ASoC: invalid widget size\n"); + return -EINVAL; + } + ret = soc_tplg_dapm_widget_create(tplg, widget); if (ret < 0) { dev_err(tplg->dev, "ASoC: failed to load widget %s\n", @@ -1652,8 +1662,6 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) return 0; - pcm = (struct snd_soc_tplg_pcm *)tplg->pos; - if (soc_tplg_check_elem_count(tplg, sizeof(struct snd_soc_tplg_pcm), count, hdr->payload_size, "PCM DAI")) { @@ -1663,7 +1671,13 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ + pcm = (struct snd_soc_tplg_pcm *)tplg->pos; for (i = 0; i < count; i++) { + if (pcm->size != sizeof(*pcm)) { + dev_err(tplg->dev, "ASoC: invalid pcm size\n"); + return -EINVAL; + } + soc_tplg_pcm_create(tplg, pcm); pcm++; } @@ -1683,6 +1697,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, return 0; manifest = (struct snd_soc_tplg_manifest *)tplg->pos; + if (manifest->size != sizeof(*manifest)) { + dev_err(tplg->dev, "ASoC: invalid manifest size\n"); + return -EINVAL; + } + tplg->pos += sizeof(struct snd_soc_tplg_manifest); if (tplg->comp && tplg->ops && tplg->ops->manifest) @@ -1699,6 +1718,14 @@ static int soc_valid_header(struct soc_tplg *tplg, if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size) return 0; + if (hdr->size != sizeof(*hdr)) { + dev_err(tplg->dev, + "ASoC: invalid header size for type %d at offset 0x%lx size 0x%zx.\n", + hdr->type, soc_tplg_get_hdr_offset(tplg), + tplg->fw->size); + return -EINVAL; + } + /* big endian firmware objects not supported atm */ if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) { dev_err(tplg->dev, -- cgit v1.2.3-58-ga151 From 1a5658c213116d56a1a38e83588f6636a57d6374 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 28 Apr 2016 01:49:48 +0000 Subject: ASoC: rsnd: count .probe/.remove for rsnd_mod_call() Current rsnd_mod_call is counting its calling count to avoid unbalanced function pair calling for error cases (ex init <-> quit). SSI parent is now controlled as "mod" on current rsnd driver. Because of this reason, SSI .remove function will be called twice if it was used as SSI parent when user tried unbind. But probe/remove pair were not counted. This patch counts probe/remove functions to avoid it. Special thans Hiep Reported-by: Hiep Cao Minh Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index fc89a67258ca..a8f61d79333b 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -276,8 +276,9 @@ struct rsnd_mod { /* * status * - * 0xH0000CB0 + * 0xH0000CBA * + * A 0: probe 1: remove * B 0: init 1: quit * C 0: start 1: stop * @@ -287,19 +288,19 @@ struct rsnd_mod { * H 0: fallback * H 0: hw_params */ +#define __rsnd_mod_shift_probe 0 +#define __rsnd_mod_shift_remove 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 #define __rsnd_mod_shift_stop 8 -#define __rsnd_mod_shift_probe 28 /* always called */ -#define __rsnd_mod_shift_remove 28 /* always called */ #define __rsnd_mod_shift_irq 28 /* always called */ #define __rsnd_mod_shift_pcm_new 28 /* always called */ #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ -#define __rsnd_mod_add_probe 0 -#define __rsnd_mod_add_remove 0 +#define __rsnd_mod_add_probe 1 +#define __rsnd_mod_add_remove -1 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -310,7 +311,7 @@ struct rsnd_mod { #define __rsnd_mod_add_hw_params 0 #define __rsnd_mod_call_probe 0 -#define __rsnd_mod_call_remove 0 +#define __rsnd_mod_call_remove 1 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 -- cgit v1.2.3-58-ga151 From ca2cd6bc6663314ba7fbf66ba7b14e099671420e Mon Sep 17 00:00:00 2001 From: anish kumar Date: Wed, 27 Apr 2016 15:39:05 -0700 Subject: ASoC: Add max98371 codec driver Signed-off-by: anish kumar Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/max98371.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/max98371.c | 442 +++++++++++++++++++++ sound/soc/codecs/max98371.h | 67 ++++ 5 files changed, 531 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h (limited to 'sound/soc') diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 000000000000..6c285235e64b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@0x31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..d764f1329042 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX98371 if I2C select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C @@ -522,6 +523,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX98371 + tristate + config SND_SOC_MAX9867 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..92ef8719c311 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -74,6 +74,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 000000000000..21fae0932137 --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,442 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_min_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 101, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_max_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 208, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_rot_gain, + 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0), + 2, 6, TLV_DB_SCALE_ITEM(-100, -100, 0), + 7, 8, TLV_DB_SCALE_ITEM(-800, -200, 0), + 9, 11, TLV_DB_SCALE_ITEM(-1200, -300, 0), + 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0), + 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0), +); + +static const struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv, + 0, 7, TLV_DB_SCALE_ITEM(0, 50, 0), + 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0) +); + +static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv, + 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0), +); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC", NULL, "HiFi Playback"}, + {"SPK_OUT", NULL, "DAC"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static const struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static const struct snd_soc_codec_driver max98371_codec = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, ®); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar "); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 000000000000..9f6330964d98 --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,67 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_DHT_MAX_WIDTH 4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_DHT_GAIN 0x31 +#define MAX98371_DHT_GAIN_WIDTH 0x4 +#define MAX98371_DHT_ROT_WIDTH 0x4 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif -- cgit v1.2.3-58-ga151 From 8ea416748bb04b7a778cb8d2fd5ec7fa51b9d521 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Thu, 5 May 2016 11:19:18 +0530 Subject: ASoC: topology: Fix memory leak in widget creation name and sname allocated in widget create are not freed when creation is successful, so free them. Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 29ae3d3a0f8a..ee7f15aa46fc 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1481,6 +1481,8 @@ widget: widget->dobj.type = SND_SOC_DOBJ_WIDGET; widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; + kfree(template.sname); + kfree(template.name); list_add(&widget->dobj.list, &tplg->comp->dobj_list); return 0; -- cgit v1.2.3-58-ga151 From e5b7d71aa5b32180adec49a17c752e577c68f740 Mon Sep 17 00:00:00 2001 From: Andrea Adami Date: Fri, 6 May 2016 17:27:34 +0200 Subject: ASoC: pxa: Fix module autoload for platform drivers These platform drivers are lacking MODULE_ALIAS so module autoloading doesn't work. Tested on corgi and poodle with kernel 4.4. Signed-off-by: Andrea Adami Signed-off-by: Mark Brown --- sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/mmp-pcm.c | 1 + sound/soc/pxa/mmp-sspa.c | 1 + sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-ac97.c | 1 + sound/soc/pxa/pxa2xx-pcm.c | 1 + 8 files changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index ec522e94b0e2..b6cb9950f05d 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -133,3 +133,4 @@ module_platform_driver(mmp_driver); MODULE_AUTHOR("Leo Yan "); MODULE_DESCRIPTION("ALSA SoC Brownstone"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:brownstone-audio"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 5c8f9db50a47..d1661fa6ee08 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -207,3 +207,4 @@ module_platform_driver(mioa701_wm9713_driver); MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)"); MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mioa701-wm9713"); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 51e790d006f5..96df9b2d8fc4 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -248,3 +248,4 @@ module_platform_driver(mmp_pcm_driver); MODULE_AUTHOR("Leo Yan "); MODULE_DESCRIPTION("MMP Soc Audio DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-pcm-audio"); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index eca60c29791a..ca8b23f8c525 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -482,3 +482,4 @@ module_platform_driver(asoc_mmp_sspa_driver); MODULE_AUTHOR("Leo Yan "); MODULE_DESCRIPTION("MMP SSPA SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-sspa-dai"); diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 4e74d9573f03..bcc81e920a67 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -161,3 +161,4 @@ module_platform_driver(palm27x_wm9712_driver); MODULE_AUTHOR("Marek Vasut "); MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:palm27x-asoc"); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index da03fad1b9cd..3cad990dad2c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -833,3 +833,4 @@ module_platform_driver(asoc_ssp_driver); MODULE_AUTHOR("Mark Brown "); MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-ssp-dai"); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index f3de615aacd7..9615e6de1306 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -287,3 +287,4 @@ module_platform_driver(pxa2xx_ac97_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa2xx-ac97"); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 9f390398d518..410d48b93031 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -117,3 +117,4 @@ module_platform_driver(pxa_pcm_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-pcm-audio"); -- cgit v1.2.3-58-ga151 From 7c3767115a04bc7aa87bdbd3352d1801d4bbeea4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 9 May 2016 13:38:10 +0300 Subject: ASoC: simple-card: Add pm callbacks to platform driver Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using simple-card. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2389ab47e25f..466492b7d4f5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -643,6 +643,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match); static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, -- cgit v1.2.3-58-ga151 From ee057d2ee73259f455cfbb7a4db808fc6b6405dd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 May 2016 02:22:37 +0000 Subject: ASoC: rsnd: don't use prohibited number to PDMACHCRn.SRS Current rsnd_dmapp_get_id() returns 0xFF as error code if system used strange connection. It will be used as PDMACHCRn.SRS, but 0xFF is prohibited number. In order not to use prohibited number, this patch indicates error message and returns 0x00 (same as SSI00) in error case. Special thanks to Dung-san. Reported-by: Nguyen Viet Dung Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7658e8fd7bdc..6bc93cbb3049 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, size = ARRAY_SIZE(gen2_id_table_cmd); } - if (!entry) - return 0xFF; + if ((!entry) || (size <= id)) { + struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); - if (size <= id) - return 0xFF; + dev_err(dev, "unknown connection (%s[%d])\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + /* use non-prohibited SRS number as error */ + return 0x00; /* SSI00 */ + } return entry[id]; } -- cgit v1.2.3-58-ga151 From 1135ef1139b3ebd5dc762b6b02384f8a7a84f8d4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 11 May 2016 14:14:05 +0300 Subject: ASoC: twl6040: Select LPPLL during standby When the codec is in standby we do not need to keep the HPPLL active. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bc3de2e844e6..7a744b581cc1 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -824,7 +824,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, { struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int ret = 0; switch (level) { case SND_SOC_BIAS_ON: @@ -832,12 +832,16 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (priv->codec_powered) + if (priv->codec_powered) { + /* Select low power PLL in standby */ + ret = twl6040_set_pll(twl6040, TWL6040_SYSCLK_SEL_LPPLL, + 32768, 19200000); break; + } ret = twl6040_power(twl6040, 1); if (ret) - return ret; + break; priv->codec_powered = 1; @@ -853,7 +857,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, break; } - return 0; + return ret; } static int twl6040_startup(struct snd_pcm_substream *substream, -- cgit v1.2.3-58-ga151 From af37d21a3245c4a54d608c1d85588203ebfe1ef9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 11 May 2016 11:48:11 +0800 Subject: ASoC: max98371 Remove duplicate entry in max98371_reg Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/max98371.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index 21fae0932137..cf0a39bb631a 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -107,7 +107,6 @@ static const struct reg_default max98371_reg[] = { { 0x39, 0x00 }, { 0x3A, 0x00 }, { 0x3B, 0x00 }, - { 0x3B, 0x00 }, { 0x3C, 0x00 }, { 0x3D, 0x00 }, { 0x3E, 0x00 }, -- cgit v1.2.3-58-ga151 From de1965159a34951a86267d13db4f2a67234139d3 Mon Sep 17 00:00:00 2001 From: Sergei Shtylyov Date: Thu, 12 May 2016 01:36:40 +0300 Subject: rcar: src: skip disabled-SRC nodes The current device tree representation of the R-Car Sample Rate Converters (SRC) assumes that they are numbered consecutively, starting from 0. Alas, this is not the case with the R8A7794 SoC where SRC0 isn't present. In order to keep the existing device trees working, I'm suggesting to use a disabled node for SRC0. Teach the SRC probe to just skip disabled nodes. Signed-off-by: Sergei Shtylyov Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 15d6ffe8be74..e39f916d0f2f 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -572,6 +572,9 @@ int rsnd_src_probe(struct rsnd_priv *priv) i = 0; for_each_child_of_node(node, np) { + if (!of_device_is_available(np)) + goto skip; + src = rsnd_src_get(priv, i); snprintf(name, RSND_SRC_NAME_SIZE, "%s.%d", @@ -595,6 +598,7 @@ int rsnd_src_probe(struct rsnd_priv *priv) if (ret) goto rsnd_src_probe_done; +skip: i++; } -- cgit v1.2.3-58-ga151 From 2213fc350841e99598e52232b56add7b873529a0 Mon Sep 17 00:00:00 2001 From: Jeremy McDermond Date: Wed, 11 May 2016 12:09:53 -0700 Subject: ASoC: tlv320aic32x4: Properly implement the positive and negative pins into the mixers The TLV320AIC32x4 has a very flexible mixer on the inputs to the ADCs. Each mixer has an available set of available pins that can be connected to the ADC positive and negative pins via three different resistor values. This allows for configuration of differential inputs as well as doing level manipulation between sources going into the mixers. The current code only provides positive pins and I implemented the resistors in an earlier patch. It turns out that it appears to more accurately model what's happening to implement each of the pins as a MUX rather than on/off switches and a mixer. This way each pin can be set to its desired resistor value. Since there are no switches, the mixer is no longer necessary in the DAPM path. I set the DAPM paths such that the "off" position of any of the MUXes turns the path off. This should allow for any input confiuration available on the codec. Signed-off-by: Jeremy McDermond Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 210 ++++++++++++++++++++++++++++++--------- 1 file changed, 161 insertions(+), 49 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 0eb8acc8cd66..85d4978d0384 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -187,36 +187,67 @@ static const char * const resistor_text[] = { "Off", "10 kOhm", "20 kOhm", "40 kOhm", }; -static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, - resistor_text); - -static const struct snd_kcontrol_new left_input_mixer_controls[] = { - SOC_DAPM_ENUM("IN1_L P Switch", in1l_lpga_p_enum), - SOC_DAPM_ENUM("IN2_L P Switch", in2l_lpga_p_enum), - SOC_DAPM_ENUM("IN3_L P Switch", in3l_lpga_p_enum), - SOC_DAPM_ENUM("IN1_R P Switch", in1r_lpga_p_enum), +/* Left mixer pins */ +static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, resistor_text); + +static SOC_ENUM_SINGLE_DECL(cml_lpga_n_enum, AIC32X4_LMICPGANIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_lpga_n_enum, AIC32X4_LMICPGANIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_lpga_n_enum, AIC32X4_LMICPGANIN, 2, resistor_text); + +static const struct snd_kcontrol_new in1l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN1_L L+ Switch", in1l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in2l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN2_L L+ Switch", in2l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in3l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN3_L L+ Switch", in3l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in1r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN1_R L+ Switch", in1r_lpga_p_enum), +}; +static const struct snd_kcontrol_new cml_to_lmixer_controls[] = { + SOC_DAPM_ENUM("CM_L L- Switch", cml_lpga_n_enum), +}; +static const struct snd_kcontrol_new in2r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN2_R L- Switch", in2r_lpga_n_enum), +}; +static const struct snd_kcontrol_new in3r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN3_R L- Switch", in3r_lpga_n_enum), }; -static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, - resistor_text); - -static const struct snd_kcontrol_new right_input_mixer_controls[] = { - SOC_DAPM_ENUM("IN1_R P Switch", in1r_rpga_p_enum), - SOC_DAPM_ENUM("IN2_R P Switch", in2r_rpga_p_enum), - SOC_DAPM_ENUM("IN3_R P Switch", in3r_rpga_p_enum), - SOC_DAPM_ENUM("IN2_L P Switch", in2l_rpga_p_enum), +/* Right mixer pins */ +static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, resistor_text); +static SOC_ENUM_SINGLE_DECL(cmr_rpga_n_enum, AIC32X4_RMICPGANIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1l_rpga_n_enum, AIC32X4_RMICPGANIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_rpga_n_enum, AIC32X4_RMICPGANIN, 2, resistor_text); + +static const struct snd_kcontrol_new in1r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN1_R R+ Switch", in1r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in2r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN2_R R+ Switch", in2r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in3r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN3_R R+ Switch", in3r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in2l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN2_L R+ Switch", in2l_rpga_p_enum), +}; +static const struct snd_kcontrol_new cmr_to_rmixer_controls[] = { + SOC_DAPM_ENUM("CM_R R- Switch", cmr_rpga_n_enum), +}; +static const struct snd_kcontrol_new in1l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN1_L R- Switch", in1l_rpga_n_enum), +}; +static const struct snd_kcontrol_new in3l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN3_L R- Switch", in3l_rpga_n_enum), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { @@ -240,14 +271,39 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { &lor_output_mixer_controls[0], ARRAY_SIZE(lor_output_mixer_controls)), SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, - &left_input_mixer_controls[0], - ARRAY_SIZE(left_input_mixer_controls)), - SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, - &right_input_mixer_controls[0], - ARRAY_SIZE(right_input_mixer_controls)), - SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), + SND_SOC_DAPM_MUX("IN1_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN2_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN3_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in3r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN2_L to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2l_to_rmixer_controls), + SND_SOC_DAPM_MUX("CM_R to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + cmr_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN1_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in1l_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN3_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in3l_to_rmixer_controls), + + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_MUX("IN1_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN2_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN3_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in3l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN1_R to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1r_to_lmixer_controls), + SND_SOC_DAPM_MUX("CM_L to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + cml_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN2_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in2r_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN3_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in3r_to_lmixer_controls), + SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), SND_SOC_DAPM_OUTPUT("HPL"), @@ -287,21 +343,77 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { {"LOR Power", NULL, "LOR Output Mixer"}, {"LOR", NULL, "LOR Power"}, - /* Left input */ - {"Left Input Mixer", "IN1_L P Switch", "IN1_L"}, - {"Left Input Mixer", "IN2_L P Switch", "IN2_L"}, - {"Left Input Mixer", "IN3_L P Switch", "IN3_L"}, - {"Left Input Mixer", "IN1_R P Switch", "IN1_R"}, - - {"Left ADC", NULL, "Left Input Mixer"}, - /* Right Input */ - {"Right Input Mixer", "IN1_R P Switch", "IN1_R"}, - {"Right Input Mixer", "IN2_R P Switch", "IN2_R"}, - {"Right Input Mixer", "IN3_R P Switch", "IN3_R"}, - {"Right Input Mixer", "IN2_L P Switch", "IN2_L"}, - - {"Right ADC", NULL, "Right Input Mixer"}, + {"Right ADC", NULL, "IN1_R to Right Mixer Positive Resistor"}, + {"IN1_R to Right Mixer Positive Resistor", "10 kOhm", "IN1_R"}, + {"IN1_R to Right Mixer Positive Resistor", "20 kOhm", "IN1_R"}, + {"IN1_R to Right Mixer Positive Resistor", "40 kOhm", "IN1_R"}, + + {"Right ADC", NULL, "IN2_R to Right Mixer Positive Resistor"}, + {"IN2_R to Right Mixer Positive Resistor", "10 kOhm", "IN2_R"}, + {"IN2_R to Right Mixer Positive Resistor", "20 kOhm", "IN2_R"}, + {"IN2_R to Right Mixer Positive Resistor", "40 kOhm", "IN2_R"}, + + {"Right ADC", NULL, "IN3_R to Right Mixer Positive Resistor"}, + {"IN3_R to Right Mixer Positive Resistor", "10 kOhm", "IN3_R"}, + {"IN3_R to Right Mixer Positive Resistor", "20 kOhm", "IN3_R"}, + {"IN3_R to Right Mixer Positive Resistor", "40 kOhm", "IN3_R"}, + + {"Right ADC", NULL, "IN2_L to Right Mixer Positive Resistor"}, + {"IN2_L to Right Mixer Positive Resistor", "10 kOhm", "IN2_L"}, + {"IN2_L to Right Mixer Positive Resistor", "20 kOhm", "IN2_L"}, + {"IN2_L to Right Mixer Positive Resistor", "40 kOhm", "IN2_L"}, + + {"Right ADC", NULL, "CM_R to Right Mixer Negative Resistor"}, + {"CM_R to Right Mixer Negative Resistor", "10 kOhm", "CM_R"}, + {"CM_R to Right Mixer Negative Resistor", "20 kOhm", "CM_R"}, + {"CM_R to Right Mixer Negative Resistor", "40 kOhm", "CM_R"}, + + {"Right ADC", NULL, "IN1_L to Right Mixer Negative Resistor"}, + {"IN1_L to Right Mixer Negative Resistor", "10 kOhm", "IN1_L"}, + {"IN1_L to Right Mixer Negative Resistor", "20 kOhm", "IN1_L"}, + {"IN1_L to Right Mixer Negative Resistor", "40 kOhm", "IN1_L"}, + + {"Right ADC", NULL, "IN3_L to Right Mixer Negative Resistor"}, + {"IN3_L to Right Mixer Negative Resistor", "10 kOhm", "IN3_L"}, + {"IN3_L to Right Mixer Negative Resistor", "20 kOhm", "IN3_L"}, + {"IN3_L to Right Mixer Negative Resistor", "40 kOhm", "IN3_L"}, + + /* Left Input */ + {"Left ADC", NULL, "IN1_L to Left Mixer Positive Resistor"}, + {"IN1_L to Left Mixer Positive Resistor", "10 kOhm", "IN1_L"}, + {"IN1_L to Left Mixer Positive Resistor", "20 kOhm", "IN1_L"}, + {"IN1_L to Left Mixer Positive Resistor", "40 kOhm", "IN1_L"}, + + {"Left ADC", NULL, "IN2_L to Left Mixer Positive Resistor"}, + {"IN2_L to Left Mixer Positive Resistor", "10 kOhm", "IN2_L"}, + {"IN2_L to Left Mixer Positive Resistor", "20 kOhm", "IN2_L"}, + {"IN2_L to Left Mixer Positive Resistor", "40 kOhm", "IN2_L"}, + + {"Left ADC", NULL, "IN3_L to Left Mixer Positive Resistor"}, + {"IN3_L to Left Mixer Positive Resistor", "10 kOhm", "IN3_L"}, + {"IN3_L to Left Mixer Positive Resistor", "20 kOhm", "IN3_L"}, + {"IN3_L to Left Mixer Positive Resistor", "40 kOhm", "IN3_L"}, + + {"Left ADC", NULL, "IN1_R to Left Mixer Positive Resistor"}, + {"IN1_R to Left Mixer Positive Resistor", "10 kOhm", "IN1_R"}, + {"IN1_R to Left Mixer Positive Resistor", "20 kOhm", "IN1_R"}, + {"IN1_R to Left Mixer Positive Resistor", "40 kOhm", "IN1_R"}, + + {"Left ADC", NULL, "CM_L to Left Mixer Negative Resistor"}, + {"CM_L to Left Mixer Negative Resistor", "10 kOhm", "CM_L"}, + {"CM_L to Left Mixer Negative Resistor", "20 kOhm", "CM_L"}, + {"CM_L to Left Mixer Negative Resistor", "40 kOhm", "CM_L"}, + + {"Left ADC", NULL, "IN2_R to Left Mixer Negative Resistor"}, + {"IN2_R to Left Mixer Negative Resistor", "10 kOhm", "IN2_R"}, + {"IN2_R to Left Mixer Negative Resistor", "20 kOhm", "IN2_R"}, + {"IN2_R to Left Mixer Negative Resistor", "40 kOhm", "IN2_R"}, + + {"Left ADC", NULL, "IN3_R to Left Mixer Negative Resistor"}, + {"IN3_R to Left Mixer Negative Resistor", "10 kOhm", "IN3_R"}, + {"IN3_R to Left Mixer Negative Resistor", "20 kOhm", "IN3_R"}, + {"IN3_R to Left Mixer Negative Resistor", "40 kOhm", "IN3_R"}, }; static const struct regmap_range_cfg aic32x4_regmap_pages[] = { -- cgit v1.2.3-58-ga151 From 45c04704e467fffe3525205454d9627325dae308 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 18 May 2016 16:19:01 +0300 Subject: ASoC: twl6040: Disconnect AUX output pads on digital mute Disconnect also the path to AUXL from the HF path during digital_mute to avoid pop noise leakage to Line-out pads. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- include/linux/mfd/twl6040.h | 1 + sound/soc/codecs/twl6040.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/include/linux/mfd/twl6040.h b/include/linux/mfd/twl6040.h index 8f9fc3d26e6d..8e95cd87cd74 100644 --- a/include/linux/mfd/twl6040.h +++ b/include/linux/mfd/twl6040.h @@ -134,6 +134,7 @@ #define TWL6040_HFDACENA (1 << 0) #define TWL6040_HFPGAENA (1 << 1) #define TWL6040_HFDRVENA (1 << 4) +#define TWL6040_HFSWENA (1 << 6) /* VIBCTLL/R (0x18/0x1A) fields */ diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bc3de2e844e6..d1e3a932cbf3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -983,9 +983,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i if (mute) { /* Power down drivers and DACs */ hflctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | - TWL6040_HFDRVENA); + TWL6040_HFDRVENA | TWL6040_HFSWENA); hfrctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | - TWL6040_HFDRVENA); + TWL6040_HFDRVENA | TWL6040_HFSWENA); } twl6040_reg_write(twl6040, TWL6040_REG_HFLCTL, hflctl); -- cgit v1.2.3-58-ga151 From d3030d11961a8c103cf07aed59905276ddfc06c2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 May 2016 18:30:39 +0100 Subject: ASoC: ak4642: Enable cache usage to fix crashes on resume The ak4642 driver is using a regmap cache sync to restore the configuration of the chip on resume but (as Peter observed) does not actually define a register cache which means that the resume is never going to work and we trigger asserts in regmap. Fix this by enabling caching. Reported-by: Geert Uytterhoeven Reported-by: Peter Ujfalusi Tested-by: Geert Uytterhoeven Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/ak4642.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1ee8506c06c7..4d8b9e49e8d6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -560,6 +560,7 @@ static const struct regmap_config ak4642_regmap = { .max_register = FIL1_3, .reg_defaults = ak4642_reg, .num_reg_defaults = NUM_AK4642_REG_DEFAULTS, + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4643_regmap = { @@ -568,6 +569,7 @@ static const struct regmap_config ak4643_regmap = { .max_register = SPK_MS, .reg_defaults = ak4643_reg, .num_reg_defaults = ARRAY_SIZE(ak4643_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4648_regmap = { @@ -576,6 +578,7 @@ static const struct regmap_config ak4648_regmap = { .max_register = EQ_FBEQE, .reg_defaults = ak4648_reg, .num_reg_defaults = ARRAY_SIZE(ak4648_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct ak4642_drvdata ak4642_drvdata = { -- cgit v1.2.3-58-ga151 From b01518ca88410195ace38ce755c1206588e5c167 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Mon, 23 May 2016 16:00:54 +0530 Subject: ASoC: kirkwood: fix build failure While building m32r allmodconfig the build failed with: ERROR: "bad_dma_ops" [sound/soc/kirkwood/snd-soc-kirkwood.ko] undefined! To satisfy the dependency CONFIG_SND_KIRKWOOD_SOC should depend on HAS_DMA. Signed-off-by: Sudip Mukherjee Signed-off-by: Mark Brown --- sound/soc/kirkwood/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 132bb83f8e99..bc3c7b5ac752 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,7 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST + depends on HAS_DMA help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- cgit v1.2.3-58-ga151