From 815b776cf5983ab69d548146fb979adac5dec4de Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:36 +0200 Subject: ASoC: sta32x: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 48740855566d..7e18200dd6a9 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -833,23 +833,6 @@ static struct snd_soc_dai_driver sta32x_dai = { .ops = &sta32x_dai_ops, }; -#ifdef CONFIG_PM -static int sta32x_suspend(struct snd_soc_codec *codec) -{ - sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int sta32x_resume(struct snd_soc_codec *codec) -{ - sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define sta32x_suspend NULL -#define sta32x_resume NULL -#endif - static int sta32x_probe(struct snd_soc_codec *codec) { struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); @@ -936,7 +919,6 @@ static int sta32x_remove(struct snd_soc_codec *codec) struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); sta32x_watchdog_stop(sta32x); - sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return 0; @@ -955,9 +937,8 @@ static bool sta32x_reg_is_volatile(struct device *dev, unsigned int reg) static const struct snd_soc_codec_driver sta32x_codec = { .probe = sta32x_probe, .remove = sta32x_remove, - .suspend = sta32x_suspend, - .resume = sta32x_resume, .set_bias_level = sta32x_set_bias_level, + .suspend_bias_off = true, .controls = sta32x_snd_controls, .num_controls = ARRAY_SIZE(sta32x_snd_controls), .dapm_widgets = sta32x_dapm_widgets, -- cgit v1.2.3 From 2062c1ff3596e1ae8aafe8082460d03d9a420282 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:37 +0200 Subject: ASoC: sta350: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sta350.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index cc97dd52aa9c..bda2ee18769e 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -912,23 +912,6 @@ static struct snd_soc_dai_driver sta350_dai = { .ops = &sta350_dai_ops, }; -#ifdef CONFIG_PM -static int sta350_suspend(struct snd_soc_codec *codec) -{ - sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int sta350_resume(struct snd_soc_codec *codec) -{ - sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define sta350_suspend NULL -#define sta350_resume NULL -#endif - static int sta350_probe(struct snd_soc_codec *codec) { struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); @@ -1065,7 +1048,6 @@ static int sta350_remove(struct snd_soc_codec *codec) { struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); - sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); return 0; @@ -1074,9 +1056,8 @@ static int sta350_remove(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver sta350_codec = { .probe = sta350_probe, .remove = sta350_remove, - .suspend = sta350_suspend, - .resume = sta350_resume, .set_bias_level = sta350_set_bias_level, + .suspend_bias_off = true, .controls = sta350_snd_controls, .num_controls = ARRAY_SIZE(sta350_snd_controls), .dapm_widgets = sta350_dapm_widgets, -- cgit v1.2.3 From cfbb77ce368b8d4181e06f8982a440702567eb98 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:38 +0200 Subject: ASoC: sta529: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sta529.c | 35 ++--------------------------------- 1 file changed, 2 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 89c748dd3d6e..b0f436d10125 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -319,41 +319,10 @@ static struct snd_soc_dai_driver sta529_dai = { .ops = &sta529_dai_ops, }; -static int sta529_probe(struct snd_soc_codec *codec) -{ - sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -/* power down chip */ -static int sta529_remove(struct snd_soc_codec *codec) -{ - sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int sta529_suspend(struct snd_soc_codec *codec) -{ - sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int sta529_resume(struct snd_soc_codec *codec) -{ - sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static const struct snd_soc_codec_driver sta529_codec_driver = { - .probe = sta529_probe, - .remove = sta529_remove, .set_bias_level = sta529_set_bias_level, - .suspend = sta529_suspend, - .resume = sta529_resume, + .suspend_bias_off = true, + .controls = sta529_snd_controls, .num_controls = ARRAY_SIZE(sta529_snd_controls), }; -- cgit v1.2.3 From 4c07a43d9691ab1f264337d683dc8655b1edca46 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:39 +0200 Subject: ASoC: stac9766: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 53b810d23fea..9878534ccd16 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -254,12 +254,6 @@ static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) return 0; } -static int stac9766_codec_suspend(struct snd_soc_codec *codec) -{ - stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int stac9766_codec_resume(struct snd_soc_codec *codec) { u16 id, reset; @@ -278,7 +272,6 @@ reset: reset++; goto reset; } - stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -349,8 +342,6 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) goto codec_err; } - stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(stac9766_snd_ac97_controls)); @@ -371,9 +362,9 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { .write = stac9766_ac97_write, .read = stac9766_ac97_read, .set_bias_level = stac9766_set_bias_level, + .suspend_bias_off = true, .probe = stac9766_codec_probe, .remove = stac9766_codec_remove, - .suspend = stac9766_codec_suspend, .resume = stac9766_codec_resume, .reg_cache_size = ARRAY_SIZE(stac9766_reg), .reg_word_size = sizeof(u16), -- cgit v1.2.3 From 93932abaa3c84c2d76ce713bbbad08bad9162483 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:01 +0100 Subject: ASoC: stac9766: Cleanup printk usage Use dev_err() instead of printk(KERN_ERR. This is common practice and makes it easy to find out which device generated the message. While we are at it also align the error messages with the other AC'97 drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 9878534ccd16..e88d9ac9cbab 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -262,7 +262,7 @@ static int stac9766_codec_resume(struct snd_soc_codec *codec) /* give the codec an AC97 warm reset to start the link */ reset: if (reset > 5) { - printk(KERN_ERR "stac9766 failed to resume"); + dev_err(codec->dev, "Failed to resume\n"); return -EIO; } codec->ac97->bus->ops->warm_reset(codec->ac97); @@ -338,7 +338,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) stac9766_reset(codec, 0); ret = stac9766_reset(codec, 1); if (ret < 0) { - printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + dev_err(codec->dev, "Failed to reset: AC97 link error\n"); goto codec_err; } -- cgit v1.2.3 From 8865051d9941de905432f59f7a88662e824d5df9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:02 +0100 Subject: ASoC: stac9766: Use table based control setup Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index e88d9ac9cbab..6c62d291cde7 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -342,9 +342,6 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) goto codec_err; } - snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls, - ARRAY_SIZE(stac9766_snd_ac97_controls)); - return 0; codec_err: @@ -359,6 +356,8 @@ static int stac9766_codec_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { + .controls = stac9766_snd_ac97_controls, + .num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls), .write = stac9766_ac97_write, .read = stac9766_ac97_read, .set_bias_level = stac9766_set_bias_level, -- cgit v1.2.3 From feec843d6c4528263724ff3f4c463ea82bf63b4a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 6 Nov 2014 15:59:22 +0100 Subject: ASoC: ssm4567: Add DAC high-pass-filter control Add a switch which can be used to enable/disable the DAC high-pass-filter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm4567.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 4b5c17f8507e..e1e33d8cb55a 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -145,6 +145,8 @@ static const struct snd_kcontrol_new ssm4567_snd_controls[] = { SOC_SINGLE_TLV("Master Playback Volume", SSM4567_REG_DAC_VOLUME, 0, 0xff, 1, ssm4567_vol_tlv), SOC_SINGLE("DAC Low Power Mode Switch", SSM4567_REG_DAC_CTRL, 4, 1, 0), + SOC_SINGLE("DAC High Pass Filter Switch", SSM4567_REG_DAC_CTRL, + 5, 1, 0), }; static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { -- cgit v1.2.3 From ead99f89b7cd2b5cfe99601380a6f6f0a1ce7e53 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 6 Nov 2014 15:59:23 +0100 Subject: ASoC: ssm4567: Add support for setting the DAI format and TDM configuration The SSM4567 has support for a couple of different DAI formats. In TDM mode it is also possible to select the TDM slot. This patch adds support for this by implementing the set_fmt and set_tdm_slot callbacks. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm4567.c | 119 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 119 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index e1e33d8cb55a..217667926a77 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -69,6 +69,22 @@ #define SSM4567_DAC_FS_64000_96000 0x3 #define SSM4567_DAC_FS_128000_192000 0x4 +/* SAI_CTRL_1 */ +#define SSM4567_SAI_CTRL_1_BCLK BIT(6) +#define SSM4567_SAI_CTRL_1_TDM_BLCKS_MASK (0x3 << 4) +#define SSM4567_SAI_CTRL_1_TDM_BLCKS_32 (0x0 << 4) +#define SSM4567_SAI_CTRL_1_TDM_BLCKS_48 (0x1 << 4) +#define SSM4567_SAI_CTRL_1_TDM_BLCKS_64 (0x2 << 4) +#define SSM4567_SAI_CTRL_1_FSYNC BIT(3) +#define SSM4567_SAI_CTRL_1_LJ BIT(2) +#define SSM4567_SAI_CTRL_1_TDM BIT(1) +#define SSM4567_SAI_CTRL_1_PDM BIT(0) + +/* SAI_CTRL_2 */ +#define SSM4567_SAI_CTRL_2_AUTO_SLOT BIT(3) +#define SSM4567_SAI_CTRL_2_TDM_SLOT_MASK 0x7 +#define SSM4567_SAI_CTRL_2_TDM_SLOT(x) (x) + struct ssm4567 { struct regmap *regmap; }; @@ -194,6 +210,107 @@ static int ssm4567_mute(struct snd_soc_dai *dai, int mute) SSM4567_DAC_MUTE, val); } +static int ssm4567_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct ssm4567 *ssm4567 = snd_soc_dai_get_drvdata(dai); + unsigned int blcks; + int slot; + int ret; + + if (tx_mask == 0) + return -EINVAL; + + if (rx_mask && rx_mask != tx_mask) + return -EINVAL; + + slot = __ffs(tx_mask); + if (tx_mask != BIT(slot)) + return -EINVAL; + + switch (width) { + case 32: + blcks = SSM4567_SAI_CTRL_1_TDM_BLCKS_32; + break; + case 48: + blcks = SSM4567_SAI_CTRL_1_TDM_BLCKS_48; + break; + case 64: + blcks = SSM4567_SAI_CTRL_1_TDM_BLCKS_64; + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_2, + SSM4567_SAI_CTRL_2_AUTO_SLOT | SSM4567_SAI_CTRL_2_TDM_SLOT_MASK, + SSM4567_SAI_CTRL_2_TDM_SLOT(slot)); + if (ret) + return ret; + + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, + SSM4567_SAI_CTRL_1_TDM_BLCKS_MASK, blcks); +} + +static int ssm4567_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct ssm4567 *ssm4567 = snd_soc_dai_get_drvdata(dai); + unsigned int ctrl1 = 0; + bool invert_fclk; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_fclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1 |= SSM4567_SAI_CTRL_1_BCLK; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC; + invert_fclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1 |= SSM4567_SAI_CTRL_1_BCLK; + invert_fclk = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1 |= SSM4567_SAI_CTRL_1_LJ; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1 |= SSM4567_SAI_CTRL_1_TDM; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1 |= SSM4567_SAI_CTRL_1_TDM | SSM4567_SAI_CTRL_1_LJ; + break; + case SND_SOC_DAIFMT_PDM: + ctrl1 |= SSM4567_SAI_CTRL_1_PDM; + break; + default: + return -EINVAL; + } + + if (invert_fclk) + ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC; + + return regmap_write(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, ctrl1); +} + static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable) { int ret = 0; @@ -248,6 +365,8 @@ static int ssm4567_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops ssm4567_dai_ops = { .hw_params = ssm4567_hw_params, .digital_mute = ssm4567_mute, + .set_fmt = ssm4567_set_dai_fmt, + .set_tdm_slot = ssm4567_set_tdm_slot, }; static struct snd_soc_dai_driver ssm4567_dai = { -- cgit v1.2.3 From 5ad72152b695ba5027f9c6ec9a48a8e1a70f25dc Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 6 Nov 2014 15:59:24 +0100 Subject: ASoC: ssm4567: Add support for disabling the boost stage This patch adds a switch to enable/disable boost stage of the output amplifier. Applications that know that they do not need the output amplifier boost stage can disable it to conserve a bit of power. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm4567.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 217667926a77..a984485108cd 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -165,13 +165,20 @@ static const struct snd_kcontrol_new ssm4567_snd_controls[] = { 5, 1, 0), }; +static const struct snd_kcontrol_new ssm4567_amplifier_boost_control = + SOC_DAPM_SINGLE("Switch", SSM4567_REG_POWER_CTRL, 1, 1, 1); + static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM4567_REG_POWER_CTRL, 2, 1), + SND_SOC_DAPM_SWITCH("Amplifier Boost", SSM4567_REG_POWER_CTRL, 3, 1, + &ssm4567_amplifier_boost_control), SND_SOC_DAPM_OUTPUT("OUT"), }; static const struct snd_soc_dapm_route ssm4567_routes[] = { + { "OUT", NULL, "Amplifier Boost" }, + { "Amplifier Boost", "Switch", "DAC" }, { "OUT", NULL, "DAC" }, }; -- cgit v1.2.3