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[alsa-devel] Applied "ASoC: fsl_esai: fix register setting issue in RIGHT_J mode" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: fsl_esai: fix register setting issue in RIGHT_J mode has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From cc29ea007347f39f4c5a4d27b0b555955a0277f9 Mon Sep 17 00:00:00 2001 From: "S.j. Wang" <shengjiu.wang(a)nxp.com> Date: Mon, 18 Feb 2019 08:29:11 +0000 Subject: [PATCH] ASoC: fsl_esai: fix register setting issue in RIGHT_J mode The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to wrong register, correct it. Fixes 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Ackedy-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 57b484768a58..afe67c865330 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -398,7 +398,8 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_RIGHT_J: /* Data on rising edge of bclk, frame high, right aligned */ - xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + xcr |= ESAI_xCR_xWA; break; case SND_SOC_DAIFMT_DSP_A: /* Data on rising edge of bclk, frame high, 1clk before data */ @@ -455,12 +456,12 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR | ESAI_xCR_xWA; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | - ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); -- 2.20.1

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[alsa-devel] Applied "ASoC: wm8741: Make function 'wm8741_mute' static" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: wm8741: Make function 'wm8741_mute' static has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 76a60f312f64dc48450b15a7f167b46e6230e4d1 Mon Sep 17 00:00:00 2001 From: Wei Yongjun <weiyongjun1(a)huawei.com> Date: Sat, 16 Feb 2019 01:35:56 +0000 Subject: [PATCH] ASoC: wm8741: Make function 'wm8741_mute' static Fixes the following sparse warning: sound/soc/codecs/wm8741.c:371:5: warning: symbol 'wm8741_mute' was not declared. Should it be static? Fixes: 36b1599340b5 ("ASoC: wm8741: Add digital mute callback") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm8741.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index a4b8c459ea57..546ea735f534 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -368,7 +368,7 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -int wm8741_mute(struct snd_soc_dai *codec_dai, int mute) +static int wm8741_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_component *component = codec_dai->component; -- 2.20.1

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[alsa-devel] Applied "ASoC: samsung: odroid: Add missing DAPM routes" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: samsung: odroid: Add missing DAPM routes has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f89aea0f132142b29dc0c8cf4d445bd12db7b1a6 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Fri, 15 Feb 2019 13:04:22 +0100 Subject: [PATCH] ASoC: samsung: odroid: Add missing DAPM routes With old DTS there will be missing DAPM routes linking BE with CODECs. Add those routes in the card driver so sound works properly on Odroid XU3/4 also without DTS updates enabling the secondary PCM. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/odroid.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 941e8c3f67a4..5b2bcd1d3450 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -149,6 +149,12 @@ static const struct snd_soc_ops odroid_card_be_ops = { .trigger = odroid_card_be_trigger, }; +/* DAPM routes for backward compatibility with old DTS */ +static const struct snd_soc_dapm_route odroid_dapm_routes[] = { + { "I2S Playback", NULL, "Mixer DAI TX" }, + { "HiFi Playback", NULL, "Mixer DAI TX" }, +}; + static struct snd_soc_dai_link odroid_card_dais[] = { { /* Primary FE <-> BE link */ @@ -237,11 +243,15 @@ static int odroid_audio_probe(struct platform_device *pdev) /* * For backwards compatibility create the secondary CPU DAI link only * if there are 2 CPU DAI entries in the cpu sound-dai property in DT. + * Also add required DAPM routes not available in old DTS. */ num_pcms = of_count_phandle_with_args(cpu, "sound-dai", "#sound-dai-cells"); - if (num_pcms == 1) + if (num_pcms == 1) { + card->dapm_routes = odroid_dapm_routes; + card->num_dapm_routes = ARRAY_SIZE(odroid_dapm_routes); card->num_links--; + } for (i = 0; i < num_pcms; i++, link += 2) { ret = of_parse_phandle_with_args(cpu, "sound-dai", -- 2.20.1

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Re: [alsa-devel] [PATCH] dt-bindings: adau1977: Fix reset-gpios typo
by Mark Brown 19 Feb '19

19 Feb '19

On Fri, Feb 15, 2019 at 12:26:33PM +0200, Bogdan Togorean wrote: > This change fixes a typo in the dt-binding examples (reset_gpio -> > reset-gpios). > Even though 'reset-gpio' is a valid construct for gpiolib the naming > 'reset-gpios' is more suited for dt-bindings documentation. Please use subject lines matching the style for the subsystem. This makes it easier for people to identify relevant patches.

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[alsa-devel] Can't load alsa-project.org
by 積丹尼 Dan Jacobson 19 Feb '19

19 Feb '19

Dear ALSA team, you need to update the TLS on alsa-project.org > That's because the site apparently only offers TLS 1.0 which libssl > does not support anymore by default. See https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=921523 P.S., The side panel of https://alsa-project.org/ has a link to "Bug tracking" which is broken and needs to link to https://alsa-project.org/main/index.php/Bug_Tracking instead.

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[alsa-devel] [PATCH] ALSA: xen-front: remove unneeded switch fall-through
by Li RongQing 19 Feb '19

19 Feb '19

This case block will be terminated by a break, so not need a switch fall-through Signed-off-by: Li RongQing <lirongqing(a)baidu.com> --- sound/xen/xen_snd_front.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/xen/xen_snd_front.c b/sound/xen/xen_snd_front.c index a9e5c2cd7698..f12fb8eeaa27 100644 --- a/sound/xen/xen_snd_front.c +++ b/sound/xen/xen_snd_front.c @@ -250,7 +250,6 @@ static void sndback_changed(struct xenbus_device *xb_dev, case XenbusStateReconfigured: /* fall through */ case XenbusStateInitialised: - /* fall through */ break; case XenbusStateInitialising: -- 2.16.2

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[alsa-devel] [PATCH v2 00/21] ASoC: dmaengine updates, secondary CPU DAI for Odroid boards
by Sylwester Nawrocki 18 Feb '19

18 Feb '19

This series restores support for the secondary CPU DAI in samsung/i2s.c driver and adds support for secondary CPU DAI on Odroid boards. I have added 4 more clean up patches comparing to first version of the patch set (16/22...19/22). The first patch adds new platform device for the secondary PCM, I tried to avoid registering new platform device in probe() as it doesn't play well with deferred probing but couldn't find a way to avoid that new device, unique devices are needed to register "PCM" components for each CPU DAI. Patches 2/21...11/21 are mostly a clean up, following conversion to single component with multiple DAIs, allowing to reference both primary/secondary DAI in DT. The patch set has been tested on Odroid U3, XU3, XU4 and Mini2440 (s3c2440 non-DT) boards. Changes since v1 ((addressing Krzysztof's review comments): - added a comment, whitespace change in patch 1/21 - dropped unnecessary new line addition from patch 1/21 Sylwester Nawrocki (21): ASoC: samsung: i2s: Restore support for the secondary PCM ASoC: samsung: i2s: Move clk supplier data to common driver data structure ASoC: samsung: i2s: Add widgets and routes for DPCM support ASoC: samsung: i2s: Move core clk to the driver common data structure ASoC: samsung: i2s: Move opclk data to common driver data structure ASoC: samsung: i2s: Move registers cache to common driver data structure ASoC: samsung: i2s: Move SFR pointer to common driver data structure ASoC: samsung: i2s: Drop spinlock pointer from i2s_dai data structure ASoC: samsung: i2s: Move IP variant data to common driver data structure ASoC: samsung: i2s: Move quirks data to common driver data structure ASoC: samsung: i2s: Get rid of a static spinlock ASoC: samsung: odroid: Add support for secondary CPU DAI ASoC: samsung: Specify DMA channel names through custom DMA config ASoC: samsung: Drop DAI DMA data chan_name assignments ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition ASoC: samsung: i2s: Comments clean up ASoC: samsung: i2s: Convert to SPDX License Indentifier ARM: dts: exynos5422-odroidxu3: Add support for secondary DAI ARM: dts: exynos5422-odroidxu4: Add support for secondary DAI .../boot/dts/exynos5422-odroidxu3-audio.dtsi | 6 +- arch/arm/boot/dts/exynos5422-odroidxu4.dts | 4 +- include/sound/dmaengine_pcm.h | 6 - sound/soc/samsung/dmaengine.c | 12 +- sound/soc/samsung/i2s.c | 543 ++++++++++-------- sound/soc/samsung/odroid.c | 131 +++-- sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/soc-generic-dmaengine-pcm.c | 21 +- 9 files changed, 413 insertions(+), 318 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: qcom: Kconfig: select dmic for sdm845
by Cheng-Yi Chiang 18 Feb '19

18 Feb '19

sdm845 uses dmic on EC so it should select CROS_EC_CODEC. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 7948e993adba..8f206cb4fcc0 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -103,6 +103,7 @@ config SND_SOC_SDM845 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 select SND_SOC_MAX98927 + select SND_SOC_CROS_EC_CODEC help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.21.0.rc0.258.g878e2cd30e-goog

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[alsa-devel] [PATCH -next] ASoC: cs35l36: Make some symbols static
by Wei Yongjun 18 Feb '19

18 Feb '19

Fixes the following sparse warnings: sound/soc/codecs/cs35l36.c:135:20: warning: symbol 'cs35l36_reg' was not declared. Should it be static? sound/soc/codecs/cs35l36.c:248:6: warning: symbol 'cs35l36_readable_reg' was not declared. Should it be static? sound/soc/codecs/cs35l36.c:398:6: warning: symbol 'cs35l36_precious_reg' was not declared. Should it be static? sound/soc/codecs/cs35l36.c:410:6: warning: symbol 'cs35l36_volatile_reg' was not declared. Should it be static? Fixes: 6ba9dd6c893b ("ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/cs35l36.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index 4f880a678812..005f1adc80b8 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -132,7 +132,7 @@ static const struct cs35l36_pll_config cs35l36_pll_sysclk[] = { {27000000, 0x3F, 0x0A}, }; -struct reg_default cs35l36_reg[] = { +static struct reg_default cs35l36_reg[] = { {CS35L36_TESTKEY_CTRL, 0x00000000}, {CS35L36_USERKEY_CTL, 0x00000000}, {CS35L36_OTP_CTRL1, 0x00002460}, @@ -245,7 +245,7 @@ struct reg_default cs35l36_reg[] = { {CS35L36_PAC_INT7_CTRL, 0x00000001}, }; -bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +static bool cs35l36_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { case CS35L36_SW_RESET: @@ -395,7 +395,7 @@ bool cs35l36_readable_reg(struct device *dev, unsigned int reg) } } -bool cs35l36_precious_reg(struct device *dev, unsigned int reg) +static bool cs35l36_precious_reg(struct device *dev, unsigned int reg) { switch (reg) { case CS35L36_TESTKEY_CTRL: @@ -407,7 +407,7 @@ bool cs35l36_precious_reg(struct device *dev, unsigned int reg) } } -bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case CS35L36_SW_RESET:

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[alsa-devel] [PATCH -next] ASoC: stm32: sai: remove set but not used variables 'mask, cr1'
by YueHaibing 18 Feb '19

18 Feb '19

Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_configure_clock': sound/soc/stm/stm32_sai_sub.c:902:11: warning: variable 'mask' set but not used [-Wunused-but-set-variable] sound/soc/stm/stm32_sai_sub.c:902:6: warning: variable 'cr1' set but not used [-Wunused-but-set-variable] It's not used any more after 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/stm/stm32_sai_sub.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index d4825700b63f..a00640f1df06 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -898,7 +898,7 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, struct snd_pcm_hw_params *params) { struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); - int cr1, mask, div = 0; + int div = 0; int sai_clk_rate, mclk_ratio, den; unsigned int rate = params_rate(params); @@ -944,8 +944,7 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, if (sai->mclk_rate) { mclk_ratio = sai->mclk_rate / rate; if (mclk_ratio == 512) { - mask = SAI_XCR1_OSR; - cr1 = SAI_XCR1_OSR; + ; } else if (mclk_ratio != 256) { dev_err(cpu_dai->dev, "Wrong mclk ratio %d\n",

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[alsa-devel] [PATCH] ASoC: samsung: odroid: Ensure proper sample rate on pri/sec PCM
by Sylwester Nawrocki 18 Feb '19

18 Feb '19

Currently when playing sound with different sample rates actual sample rate will be determined by audio stream which starts first on either primary or secondary PCM. The audio root clock will be configured appropriately only for the first stream. As the hardware is limited to same sample rate on both interfaces we need to disallow streams with different sample rates. It is done by this patch by returning error in FE hw_params if there is already active stream running with different sample rate. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/odroid.c | 56 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 18bb3bfe0300..941e8c3f67a4 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -20,6 +20,11 @@ struct odroid_priv { struct snd_soc_card card; struct clk *clk_i2s_bus; struct clk *sclk_i2s; + + /* Spinlock protecting fields below */ + spinlock_t lock; + unsigned int be_sample_rate; + bool be_active; }; static int odroid_card_fe_startup(struct snd_pcm_substream *substream) @@ -31,8 +36,25 @@ static int odroid_card_fe_startup(struct snd_pcm_substream *substream) return 0; } +static int odroid_card_fe_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct odroid_priv *priv = snd_soc_card_get_drvdata(rtd->card); + unsigned long flags; + int ret = 0; + + spin_lock_irqsave(&priv->lock, flags); + if (priv->be_active && priv->be_sample_rate != params_rate(params)) + ret = -EINVAL; + spin_unlock_irqrestore(&priv->lock, flags); + + return ret; +} + static const struct snd_soc_ops odroid_card_fe_ops = { .startup = odroid_card_fe_startup, + .hw_params = odroid_card_fe_hw_params, }; static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, @@ -41,6 +63,7 @@ static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct odroid_priv *priv = snd_soc_card_get_drvdata(rtd->card); unsigned int pll_freq, rclk_freq, rfs; + unsigned long flags; int ret; switch (params_rate(params)) { @@ -87,11 +110,43 @@ static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, return ret; } + spin_lock_irqsave(&priv->lock, flags); + priv->be_sample_rate = params_rate(params); + spin_unlock_irqrestore(&priv->lock, flags); + + return 0; +} + +static int odroid_card_be_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct odroid_priv *priv = snd_soc_card_get_drvdata(rtd->card); + unsigned long flags; + + spin_lock_irqsave(&priv->lock, flags); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + priv->be_active = true; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + priv->be_active = false; + break; + } + + spin_unlock_irqrestore(&priv->lock, flags); + return 0; } static const struct snd_soc_ops odroid_card_be_ops = { .hw_params = odroid_card_be_hw_params, + .trigger = odroid_card_be_trigger, }; static struct snd_soc_dai_link odroid_card_dais[] = { @@ -150,6 +205,7 @@ static int odroid_audio_probe(struct platform_device *pdev) card->owner = THIS_MODULE; card->fully_routed = true; + spin_lock_init(&priv->lock); snd_soc_card_set_drvdata(card, priv); ret = snd_soc_of_parse_card_name(card, "model"); -- 2.17.1

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[alsa-devel] [PATCH] ASoC: simple-card: Fix refcount underflow
by Daniel Baluta 18 Feb '19

18 Feb '19

of_get_child_by_name() takes a reference we'll need to drop later so when we substitute in top we need to take a reference as well as just assigning. Without this patch we hit the following error: [ 1.246852] OF: ERROR: Bad of_node_put() on /sound-wm8524 [ 1.262261] Hardware name: NXP i.MX8MQ EVK (DT) [ 1.266807] Workqueue: events deferred_probe_work_func [ 1.271950] Call trace: [ 1.274406] dump_backtrace+0x0/0x158 [ 1.278074] show_stack+0x14/0x20 [ 1.281396] dump_stack+0xa8/0xcc [ 1.284717] of_node_release+0xb0/0xc8 [ 1.288474] kobject_put+0x74/0xf0 [ 1.291879] of_node_put+0x14/0x28 [ 1.295286] __of_get_next_child+0x44/0x70 [ 1.299387] of_get_next_child+0x3c/0x60 [ 1.303315] simple_for_each_link+0x1dc/0x230 [ 1.307676] simple_probe+0x80/0x540 [ 1.311256] platform_drv_probe+0x50/0xa0 This patch is based on an earlier version posted by Kuninori Morimoto and commit message includes explanations from Mark Brown. https://patchwork.kernel.org/patch/10814255/ Reported-by: Vicente Bergas <vicencb(a)gmail.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 08df261024cf..dc18c4492955 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -445,7 +445,7 @@ static int simple_for_each_link(struct simple_priv *priv, /* Check if it has dai-link */ node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { - node = top; + node = of_node_get(top); is_top = 1; } -- 2.17.1

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[alsa-devel] [PATCH] ASoC: simple-card: fixup refcount_t underflow
by Kuninori Morimoto 18 Feb '19

18 Feb '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card") merged simple-card and simple-scu-card. Then it had refcount underflow bug. This patch fixup it. We will get below error without this patch. OF: ERROR: Bad of_node_put() on /sound CPU: 3 PID: 237 Comm: kworker/3:1 Not tainted 5.0.0-rc6+ #1514 Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) Workqueue: events deferred_probe_work_func Call trace: dump_backtrace+0x0/0x150 show_stack+0x24/0x30 dump_stack+0xb0/0xec of_node_release+0xd0/0xd8 kobject_put+0x74/0xe8 of_node_put+0x24/0x30 __of_get_next_child+0x50/0x70 of_get_next_child+0x40/0x68 asoc_simple_card_probe+0x604/0x730 platform_drv_probe+0x58/0xa8 ... Reported-by: Vicente Bergas <vicencb(a)gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Vicente, can you please test this patch ? sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 37e001c..3fe3441 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -462,7 +462,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) conf_idx = 0; node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { - node = dev->of_node; + node = of_node_get(top); loop = 0; } -- 2.7.4

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[alsa-devel] [PATCH] ASoC: fsi: fix spelling mistake "doens't" -> "doesn't"
by Colin King 18 Feb '19

18 Feb '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 285afbafa662..3447dbdba1f1 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -780,7 +780,7 @@ static int fsi_clk_init(struct device *dev, return -EINVAL; } if (clock->div == clock->own) { - dev_err(dev, "cpu doens't support div clock\n"); + dev_err(dev, "cpu doesn't support div clock\n"); return -EINVAL; } } -- 2.20.1

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[alsa-devel] [PATCH 0/5] ASoC: codecs: ad193x: Several fixes and imprevements
by Codrin.Ciubotariu＠microchip.com 18 Feb '19

18 Feb '19

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> This patch set contains some changes I needed to make I2S and TDM (DSP_A) formats work. I tested only with ad1936, but the patches should work fine for all the codecs that use this driver. Although I checked the DS for all of them (I hope) to assure that the patches apply, I would really appreciate if someone else can test them on another codec. Codrin Ciubotariu (5): ASoC: codecs: ad193x: Remove capture support for codecs without ADC ASoC: codecs: ad193x: Set constraint to always have 32 sample bits ASoC: codecs: ad193x: Fix frame polarity for DSP_A format ASoC: codecs: ad193x: Add runtime support for DSP_A and I2S modes ASoC: codecs: ad193x: Add support to disable on-chip PLL sound/soc/codecs/ad193x.c | 76 ++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/ad193x.h | 8 +++++ 2 files changed, 80 insertions(+), 4 deletions(-) -- 2.17.1

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[alsa-devel] snd_hda_codec_hdmi: `hdaudio hdaudioC0D2: Unable to bind the codec`
by Paul Menzel 18 Feb '19

18 Feb '19

Dear Linux folks, With drm-tip (v4.19-rc3-946-g09b295662edd) Linux prints `snd_hda_codec_hdmi hdaudioC0D2: No i915 binding for Intel HDMI/DP codec` to the log. ``` […] [ 12.481788] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC671: line_outs=1 (0x21/0x0/0x0/0x0/0x0) type:line [ 12.482536] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=1 (0x17/0x0/0x0/0x0/0x0) [ 12.483184] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 12.483737] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 12.484198] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 12.484582] snd_hda_codec_realtek hdaudioC0D0: Front Mic=0x19 [ 12.485100] snd_hda_codec_realtek hdaudioC0D0: Rear Mic=0x18 [ 12.485519] snd_hda_codec_realtek hdaudioC0D0: Line=0x1a [ 12.497685] snd_hda_codec_hdmi hdaudioC0D2: No i915 binding for Intel HDMI/DP codec [ 12.498311] hdaudio hdaudioC0D2: Unable to bind the codec [ 12.498789] input: HDA Intel PCH Front Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input6 [ 12.499567] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input7 [ 12.500268] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1f.3/sound/card0/input8 [ 12.501039] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1f.3/sound/card0/input9 [ 12.501683] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input10 [ 17.743338] [drm:intel_pch_type [i915]] Found SunrisePoint PCH […] ``` Is that an error or just a notice? Kind regards, Paul

2 12

Re: [alsa-devel] [PATCH 1/2] ASoC: samsung: i2s: Fix the secondary platform device registration
by Krzysztof Kozlowski 18 Feb '19

18 Feb '19

On Fri, 15 Feb 2019 at 15:48, Sylwester Nawrocki <s.nawrocki(a)samsung.com> wrote: > > This fixes unregistration of the secondary platform device so all > resources are properly released. The test for NULL priv->pdev_sec > is not necessary and it is removed. > > Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > --- > This patch is based off of ASoC for-next and patches: > ASoC: samsung: odroid: Ensure proper sample rate on pri/sec PCM > ASoC: samsung: i2s: Prevent potential NULL platform data dereference > ASoC: samsung: odroid: Add missing DAPM routes > > sound/soc/samsung/i2s.c | 7 +++---- > 1 file changed, 3 insertions(+), 4 deletions(-) > > diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c > index 6bf0f55d1e51..e36c44e2f1bb 100644 > --- a/sound/soc/samsung/i2s.c > +++ b/sound/soc/samsung/i2s.c > @@ -1359,11 +1359,10 @@ static int i2s_create_secondary_device(struct samsung_i2s_priv *priv) > > static void i2s_delete_secondary_device(struct samsung_i2s_priv *priv) > { > - if (priv->pdev_sec) { > - platform_device_del(priv->pdev_sec); > - priv->pdev_sec = NULL; > - } > + platform_device_unregister(priv->pdev_sec); > + priv->pdev_sec = NULL; Reviewed-by: Krzysztof Kozlowski <krzk(a)kernel.org> Although I think that you might need to re-order calls in samsung_i2s_remove(). In general they should be in exact reverse order of probe(). In this case, clk_disable_unprepare(priv->clk) should be after unregistering secondary device. If order has to be different because of some reasons - could you document them in comment? Best regards, Krzysztof

2 1

Re: [alsa-devel] [PATCH 2/2] ASoC: samsung: i2s: Fix multiple "IIS multi" devices initialization
by Krzysztof Kozlowski 18 Feb '19

18 Feb '19

On Fri, 15 Feb 2019 at 15:48, Sylwester Nawrocki <s.nawrocki(a)samsung.com> wrote: > > On some SoCs (e.g. Exynos5433) there are multiple "IIS multi audio > interfaces" and the driver will try to register there multiple times > same platform device for the secondary FIFO, which of course fails > miserably. To fix this we derive the secondary platform device name > from the primary device name. The secondary device name will now > be <primary_dev_name>-sec instead of fixed "samsung-i2s-sec". > > The fixed platform_device_id table entry is removed as the secondary > device name is now dynamic and device/driver matching is done through > driver_override. > > Reported-by: Marek Szyprowski <m.szyprowski(a)samsung.com> > Suggested-by: Marek Szyprowski <m.szyprowski(a)samsung.com> > Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > --- > sound/soc/samsung/i2s.c | 49 +++++++++++++++++++++++++------------- > sound/soc/samsung/odroid.c | 2 +- > 2 files changed, 33 insertions(+), 18 deletions(-) > > diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c > index e36c44e2f1bb..4a6dd86459bc 100644 > --- a/sound/soc/samsung/i2s.c > +++ b/sound/soc/samsung/i2s.c > @@ -1339,20 +1339,34 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) > /* Create platform device for the secondary PCM */ > static int i2s_create_secondary_device(struct samsung_i2s_priv *priv) > { > - struct platform_device *pdev; > + struct platform_device *pdev_sec; > + const char *devname; > int ret; > > - pdev = platform_device_register_simple("samsung-i2s-sec", -1, NULL, 0); > - if (!pdev) > + devname = devm_kasprintf(&priv->pdev->dev, GFP_KERNEL, "%s-sec", > + dev_name(&priv->pdev->dev)); > + if (!devname) > return -ENOMEM; > > - ret = device_attach(&pdev->dev); > + pdev_sec = platform_device_alloc(devname, -1); > + if (!pdev_sec) > + return -ENOMEM; > + > + pdev_sec->driver_override = "samsung-i2s"; > + > + ret = platform_device_add(pdev_sec); > if (ret < 0) { > - dev_info(&pdev->dev, "device_attach() failed\n"); > + platform_device_put(pdev_sec); > return ret; > } > > - priv->pdev_sec = pdev; > + priv->pdev_sec = pdev_sec; > + > + ret = device_attach(&pdev_sec->dev); > + if (ret < 0) { > + dev_info(&pdev_sec->dev, "device_attach() failed\n"); Don't you need here platform_device_unregister()? Best regards, Krzysztof

2 1

Re: [alsa-devel] [PATCH 2/2] ASoC: samsung: i2s: Fix multiple "IIS multi" devices initialization
by Krzysztof Kozlowski 18 Feb '19

18 Feb '19

On Fri, 15 Feb 2019 at 15:48, Sylwester Nawrocki <s.nawrocki(a)samsung.com> wrote: > > On some SoCs (e.g. Exynos5433) there are multiple "IIS multi audio > interfaces" and the driver will try to register there multiple times > same platform device for the secondary FIFO, which of course fails > miserably. To fix this we derive the secondary platform device name > from the primary device name. The secondary device name will now > be <primary_dev_name>-sec instead of fixed "samsung-i2s-sec". > > The fixed platform_device_id table entry is removed as the secondary > device name is now dynamic and device/driver matching is done through > driver_override. > > Reported-by: Marek Szyprowski <m.szyprowski(a)samsung.com> > Suggested-by: Marek Szyprowski <m.szyprowski(a)samsung.com> > Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > --- > sound/soc/samsung/i2s.c | 49 +++++++++++++++++++++++++------------- > sound/soc/samsung/odroid.c | 2 +- > 2 files changed, 33 insertions(+), 18 deletions(-) > > diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c > index e36c44e2f1bb..4a6dd86459bc 100644 > --- a/sound/soc/samsung/i2s.c > +++ b/sound/soc/samsung/i2s.c > @@ -1339,20 +1339,34 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) > /* Create platform device for the secondary PCM */ > static int i2s_create_secondary_device(struct samsung_i2s_priv *priv) > { > - struct platform_device *pdev; > + struct platform_device *pdev_sec; > + const char *devname; > int ret; > > - pdev = platform_device_register_simple("samsung-i2s-sec", -1, NULL, 0); > - if (!pdev) > + devname = devm_kasprintf(&priv->pdev->dev, GFP_KERNEL, "%s-sec", > + dev_name(&priv->pdev->dev)); > + if (!devname) > return -ENOMEM; > > - ret = device_attach(&pdev->dev); > + pdev_sec = platform_device_alloc(devname, -1); > + if (!pdev_sec) > + return -ENOMEM; > + > + pdev_sec->driver_override = "samsung-i2s"; This is wrong, see: https://patchwork.kernel.org/project/linux-samsung-soc/list/?series=81639&s… Although there is ongoing discussion whether the drivers should use driver_override at all: https://www.mail-archive.com/linux-kernel@vger.kernel.org/msg1934411.html Best regards, Krzysztof

1 0

Re: [alsa-devel] [PATCH -next] ASoC: samsung: i2s: Fix return value check in i2s_create_secondary_device()
by Sylwester Nawrocki 18 Feb '19

18 Feb '19

On 2/16/19 02:20, Wei Yongjun wrote: > In case of error, the function platform_device_register_simple() returns > ERR_PTR() and never returns NULL. The NULL test in the return value > check should be replaced with IS_ERR(). Thanks for the patch, it looks good but I have already prepared other fix that removes the platform_device_register_simple() call entirely: "ASoC: samsung: i2s: Fix multiple "IIS multi" devices initialization" https://www.spinics.net/lists/kernel/msg3045092.html -- Thanks, Sylwester

1 0

Re: [alsa-devel] [PATCH -next] ASoC: wm8741: Make function 'wm8741_mute' static
by Charles Keepax 18 Feb '19

18 Feb '19

On Sat, Feb 16, 2019 at 01:35:56AM +0000, Wei Yongjun wrote: > Fixes the following sparse warning: > > sound/soc/codecs/wm8741.c:371:5: warning: > symbol 'wm8741_mute' was not declared. Should it be static? > > Fixes: 36b1599340b5 ("ASoC: wm8741: Add digital mute callback") > Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> > --- Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Thanks, Charles

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Expose sample resolution through proc interface
by Jussi Laako 18 Feb '19

18 Feb '19

At least some USB devices use (MSB-aligned) audio format larger than the actual resolution of the device. In order to expose the actual device resolution (bBitResolution), add extra field to the procfs stream info interface. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/card.h | 1 + sound/usb/format.c | 2 ++ sound/usb/proc.c | 1 + 3 files changed, 4 insertions(+) diff --git a/sound/usb/card.h b/sound/usb/card.h index ac785d15ced4..d9bb2c4a3bd9 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -14,6 +14,7 @@ struct audioformat { u64 formats; /* ALSA format bits */ unsigned int channels; /* # channels */ unsigned int fmt_type; /* USB audio format type (1-3) */ + unsigned int fmt_bits; /* number of significant bits */ unsigned int frame_size; /* samples per frame for non-audio */ int iface; /* interface number */ unsigned char altsetting; /* corresponding alternate setting */ diff --git a/sound/usb/format.c b/sound/usb/format.c index fd13ac11b136..d5b7cbcc2956 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -87,6 +87,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, } } + fp->fmt_bits = (unsigned int) sample_width; + if ((pcm_formats == 0) && (format == 0 || format == (1 << UAC_FORMAT_TYPE_I_UNDEFINED))) { /* some devices don't define this correctly... */ diff --git a/sound/usb/proc.c b/sound/usb/proc.c index e80c9d0749c9..ef9190530fd2 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -109,6 +109,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s if (subs->speed != USB_SPEED_FULL) snd_iprintf(buffer, " Data packet interval: %d us\n", 125 * (1 << fp->datainterval)); + snd_iprintf(buffer, " Bits: %d\n", fp->fmt_bits); // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } -- 2.17.1

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Re: [alsa-devel] Multiple codecs on one sound card for multi-channel sound card
by Yann CARDAILLAC 18 Feb '19

18 Feb '19

Hi guys, Where are we now with TDM on several codecs on the same TDM channel ? I'm sorry to dig out this old thread but I couldn't really find any proper information about that. I only plan to use 4 channel TDM with two codecs. I'm using IMX6's SSI as master for both codec. What I've got working today is 1 codec with ssi as I2S master. Do you have any pointer on what is achieved so far and how to implement it ? I'm using kernel *v4.14.78*. Any pointer to DTS achieving that or audio card doing that kind of thing would already be an amazing first hint.! Regards, Yann >Got it. Thanks Arnaud. >-Caleb > >On Thu, Sep 17, 2015 at 9:09 AM, arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel> ><arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>> wrote: >>>>>*> Le 17/09/2015 17:34, Caleb Crome a écrit : *>>>>>>* On Thu, Sep 17, 2015 at 7:33 AM, arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel> *>>>* <arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>> wrote: *>>>>>>>>* hello, *>>>>>>>*> Le 17/09/2015 15:38, Caleb Crome a écrit : *>>>>>>>>>>* On Thu, Sep 17, 2015 at 1:51 AM, arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel> *>>>>>* <arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>> wrote: *>>>>>>>>>>>>* Hello Caleb, *>>>>>>>>>>>>* freescale SSI is fine with more than 2 channels. *>>>>>>* I have planed to publish a set of patch in this direction but we still *>>>>>>* have *>>>>>>* some corner cases to fix first (rare issues with channels alignment) *>>>>>>>>>>* I'd love to see anything you're willing to show :-) *>>>>>>>>>>>* Yet, this following is far enough to have it working. *>>>>>>* And since the max/min rate is at the end the intersection of what the *>>>>>>* SSI, *>>>>>>* the codec and the card are declaring, it will not change anything until *>>>>>>* you *>>>>>>* connect the SSI to a codec with more than 2 channels, or a multi-codec *>>>>>>* solution. *>>>>*> *>>>>* What if I change channels_max to 16 for capture and playback? Last *>>>>>* time I tried, it would not keep the channels synchronized on the TDM *>>>>>* bus. I tried playing with the FIFO settings, which helped, but it *>>>>>* still would not start each channel in the right slot reliably. *>>>>>* Perhaps I was using an older kernel. *>>>>>>>>* you put the finger on what's hurting ! ;) *>>>>>>>>* In fact, we made 8 channels working in reliable way on imx50, with *>>>>* predictive latency ... with an old kernel (3.17) ... based on freescale *>>>>* BSP... on sound/soc/imx/imx-ssi.c (something that doesn't exist outside *>>>>* freescale BSP) *>>>>>>>>* Well, just to say it is doable on any recent imx since the hardware is *>>>>* not *>>>>* the issue. But the job was a set of awful hacks. *>>>>>>* Sorry, I don't quite understand. Are you saying that on a recent *>>>* kernel (say 4.1 or 4.2), you think it will just work. Or are yo *>>>* saying that even on a modern kernel it's a set of awful hacks? And *>>>* should I use fsl_ssi.c or imx-ssi.c (device tree/non device tree) *>>>* version? *>>>>* - I'm saying it is working on recent upstream kernel, using *>>* sound/soc/fsl/fsl_ssi.c, but with *>>* some channel sync issues in rare conditions (at least 4 channels case, *>>* opening multiple SSI at the same time). *>>* So, you should test on your side with you setup, and simply be prepared to *>>* not see it working as a charm. *>>>>* - we made it work perfectly, but on an old freescale BSP, where freescale *>>* rewrites a SSI driver different from the already 2 drivers (device tree/non *>>* device tree) available today in upstream + lot of awful hacks => so, this is *>>* not a hardware issue. *>>>>* - when I will have time, I will start from the upstream driver, and make it *>>* work correctly with 8 channels (or more) *>>>>* arnaud *>>>>>>>>* In short, what's your best current recommendation to get it working today? *>>>>>>>* My plan was to re-do all this job on an upstream kernel, in a way it can *>>>>* be *>>>>* accepted. *>>>>* Unfortunately you will have to wait, I'm under pressure for other stuff *>>>>* right now. *>>>>>>* Heh, understood. If there's anything I can do to help, let me know. *>>>>>>* Thanks, *>>>* -Caleb *>>>>>>>* Regards, *>>>>* arnaud *>>>>>>>>>>>>>* I'll give it a try again. Like I said, we'd love to get our *>>>>>* 16-channel board up and running on the MX6, so I'm definitely *>>>>>* motivated to put some work in to get it reliable. *>>>>>>>>>>* Thanks, *>>>>>* -Caleb *>>>>>>> -- [image: SMILE] <http://www.smile.eu/> 20 rue des Jardins 92600 Asnières-sur-Seine *Yann CARDAILLAC* Ingénieur Linux Embarqué [image: email] yann.cardaillac(a)smile.fr [image: url] http://www.smile.eu [image: Twitter] <https://twitter.com/GroupeSmile> [image: Facebook] <https://www.facebook.com/smileopensource> [image: LinkedIn] <https://www.linkedin.com/company/smile> [image: Github] <https://github.com/Smile-SA> [image: eco] Pour la planète, n'imprimez ce mail que si c'est nécessaire

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[alsa-devel] [PATCH V3] ASoC: fsl_esai: fix register setting issue in RIGHT_J mode
by S.j. Wang 18 Feb '19

18 Feb '19

The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to wrong register, correct it. Fixes 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Ackedy-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changes in v3 - Add Cc: <stable(a)vger.kernel.org> sound/soc/fsl/fsl_esai.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 57b484768a58..afe67c865330 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -398,7 +398,8 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_RIGHT_J: /* Data on rising edge of bclk, frame high, right aligned */ - xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + xcr |= ESAI_xCR_xWA; break; case SND_SOC_DAIFMT_DSP_A: /* Data on rising edge of bclk, frame high, 1clk before data */ @@ -455,12 +456,12 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR | ESAI_xCR_xWA; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | - ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); -- 1.9.1

1 0

[alsa-devel] [PATCH V2] ASoC: fsl_esai: fix register setting issue in RIGHT_J mode
by S.j. Wang 18 Feb '19

18 Feb '19

Fixes commit 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to wrong register, correct it. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Ackedy-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changes in v2 - add Fixes tag and cc stable kernel. sound/soc/fsl/fsl_esai.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 57b484768a58..afe67c865330 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -398,7 +398,8 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_RIGHT_J: /* Data on rising edge of bclk, frame high, right aligned */ - xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + xcr |= ESAI_xCR_xWA; break; case SND_SOC_DAIFMT_DSP_A: /* Data on rising edge of bclk, frame high, 1clk before data */ @@ -455,12 +456,12 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR | ESAI_xCR_xWA; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | - ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); -- 1.9.1

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Re: [alsa-devel] [PATCH -next] ASoC: samsung: i2s: Fix return value check in i2s_create_secondary_device()
by Krzysztof Kozlowski 18 Feb '19

18 Feb '19

On Sat, 16 Feb 2019 at 02:07, Wei Yongjun <weiyongjun1(a)huawei.com> wrote: > > In case of error, the function platform_device_register_simple() returns > ERR_PTR() and never returns NULL. The NULL test in the return value > check should be replaced with IS_ERR(). > > Fixes: 7196c64c7d0c ("ASoC: samsung: i2s: Restore support for the secondary PCM") > Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> > --- > sound/soc/samsung/i2s.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) Reviewed-by: Krzysztof Kozlowski <krzk(a)kernel.org> Best regards, Krzysztof

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Re: [alsa-devel] [PATCH] ASoC: simple-card: fixup refcount_t underflow
by Daniel Baluta 18 Feb '19

18 Feb '19

On Fri, Feb 15, 2019 at 8:43 PM Takashi Iwai <tiwai(a)suse.de> wrote: > > On Fri, 15 Feb 2019 18:31:08 +0100, > Mark Brown wrote: > > > > On Fri, Feb 15, 2019 at 06:21:04PM +0100, Takashi Iwai wrote: > > > > > Actually it's not all. As mentioned in my earlier post, there are > > > more of refcount unbalance in the code. Lots of them, really. > > > > > Below is a patch I've fixed quickly over the whole scanning the sound > > > tree. Some might be incorrect, so more detailed review for each piece > > > of change is needed... > > > > Extra references aren't such a problem for practical systems since we > > don't actually ever free any of the DT (there's no mainline way to use > > overlays yet...), that's how none of this stuff ever gets noticed. I'm > > not surprised there's loads of missing frees TBH. In contrast if we've > > got extra frees they will get noticed and cause problems. > > Yeah, the rest are just leaks, and we have already tons of them :) > > The point is that the commit in question introduced multiple such > issues, so they should be addressed as well. > > But, the biggest problem is that OF api is very hard to use right... Hello Mark/Takashi, Just sent this https://lkml.org/lkml/2019/2/16/52 to fix the problem that was bugging me and to update the patch from Kuninori. Feel free to send a patch for the rest of the refcount issues :). Kuninori, if you consider necessary please add your Signed-off-by to the patch as it is mostly your work but I didn't know exactly how to give you credit for it :(. thanks, Daniel.

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Re: [alsa-devel] [PATCH] ASoC: simple-card: fixup refcount_t underflow
by Kuninori Morimoto 18 Feb '19

18 Feb '19

Hi Takashi-san > > commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card") > > merged simple-card and simple-scu-card. Then it had refcount > > underflow bug. This patch fixup it. > > We will get below error without this patch. > > > > OF: ERROR: Bad of_node_put() on /sound > > CPU: 3 PID: 237 Comm: kworker/3:1 Not tainted 5.0.0-rc6+ #1514 > > Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) > > Workqueue: events deferred_probe_work_func > > Call trace: > > dump_backtrace+0x0/0x150 > > show_stack+0x24/0x30 > > dump_stack+0xb0/0xec > > of_node_release+0xd0/0xd8 > > kobject_put+0x74/0xe8 > > of_node_put+0x24/0x30 > > __of_get_next_child+0x50/0x70 > > of_get_next_child+0x40/0x68 > > asoc_simple_card_probe+0x604/0x730 > > platform_drv_probe+0x58/0xa8 > > ... > > Reported-by: Vicente Bergas <vicencb(a)gmail.com> > > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > > Please don't forget to put Fixes tag for a regression fix. Oops, indeed. Thanks. will add it on v2 patch. > And, looking at the code in 5.0-rc, it seems still leaving the > reference in some condition. The 5.0-rc code corrected your patch > looks like: > > node = of_get_child_by_name(top, PREFIX "dai-link"); > if (!node) { > node = of_node_get(top); > loop = 0; > } > > do { > .... > node = of_get_next_child(top, node); > } while (loop && node); > > So when loop=0, it aborts the loop at the first iteration. If node is > non-NULL at that point, it still keeps the refcount taken in > of_get_next_child(). > > That is, you'd need to call of_node_put(node) after the loop. Ah.. yes.. > And, this leads me checking the whole code, and I'm afraid that we > have many similar unblanced refcounts. For example, a tricky one is > for_each_child_of_node() loop. When you abort at the middle of the > loop, you'd need to unreference the remaining node. But we return or > abort immediately without unreferencing in many places. > > Hrm... Yes, I had noticed this refcounts issue. The code can be very confusable for this refcounts... Best regards --- Kuninori Morimoto

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Re: [alsa-devel] [PATCH] ASoC: fsl_esai: fix register setting issue in RIGHT_J mode
by Fabio Estevam 16 Feb '19

16 Feb '19

Hi Shengjiu, On Fri, Feb 15, 2019 at 9:04 AM S.j. Wang <shengjiu.wang(a)nxp.com> wrote: > > The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to > wrong register, correct it. > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Please add a Fixes tag and Cc stable. Also, it seems that Mark Brown is not on Cc. Please Cc him in v2. Thanks

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Re: [alsa-devel] Saffire Pro 26 IO Restarting Issue
by Takashi Sakamoto 15 Feb '19

15 Feb '19

Hi, On Fri, Feb 15, 2019 at 12:49:19PM +0000, ackolonges fds wrote: > Hi All, > > I was wondering if there might be anyone on here that could give me any > pointers in getting my Focusrite Saffire Pro 26 i/o interface working with > Linux via FFADO? > > It's a new desktop PC (Ryzen 5 2600, 8GB RAM), dual booting Windows 10 and > Ubuntu 18.10, with a PCI Express, Texas Instruments XIO2213 Firewire card. > > I have tested this Saffire interface successfully on a MacBook Pro (it was > plug and play), and also on this very desktop computer in Windows 10. > However, when the interface is connected in Ubuntu 18.10, it repeatedly > and periodically resets itself about 10-15 seconds after booting up, in a > loop. If I run ffado-dbus-server while it's booting, it connects > successfully to the interface, and quickly running FFADO mixer afterwards > lets me see the controls available, for a short time until the interface > resets itself, cutting all the connections. > > Has anybody here ever heard of this kind of issue before? Or perhaps > someone may have some ideas that might aid in troubleshooting the issue? > > Thanks so much for any advice you may have, In case of Focusrite Saffire Pro I/O 10 and 26, a combination of ALSA bebob driver and pulseaudio brings this problem due to quirk of these two models reported in bugzilla.kernel.org[1]. I've already committed a solution and it was merged to pulseaudio[2], however it has never included into any releases of pulseaudio at present. Therefore you can see this issue on any desktop environment in released distribution such as the latest Ubuntu. You can avoid this issue by two handy solutions at a quick glance: 1. suppress pulseaudio to handle your device 2. suppress your system to load ALSA bebob driver For solution 1, it's the best way to edit 90-pulseaudio.rules in your system. $ systemctl --user stop pulseaudio.service pulseaudio.socket $ sudo nano /lib/udev/rules.d/90-pulseaudio.rules (modify the above file according to my patch[2].) $ systemctl --user start pulseaudio.service pulseaudio.socket For solution 2, it's the easiest way to add ALSA bebob driver to blacklist. $ sudo nano /etc/modprobe.d/blacklist-bebob.conf (add entry of 'blacklist snd-bebob') (then unplug and plugin your device) Sorry for the inconvenience. [1] https://bugzilla.kernel.org/show_bug.cgi?id=199365 [2] https://gitlab.freedesktop.org/pulseaudio/pulseaudio/commit/37358e42c49abe3… Regards Takashi Sakamoto

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Re: [alsa-devel] [PATCH v5 0/3] Add NXP AUDMIX device and machine drivers
by Nicolin Chen 15 Feb '19

15 Feb '19

On Fri, Feb 15, 2019 at 02:01:32PM +0000, Viorel Suman wrote: > The patchset adds NXP Audio Mixer (AUDMIX) device and machine > drivers and related DT bindings documentation. For this series, Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> And Rob gave his at the previous version already. Thanks. > Changes since V4: > 1. Removed "model" attribute from device driver DT bindings documentation > as suggested by Nicolin. > > Changes since V3: > 1. Removed machine driver DT bindings documentation. > 2. Trigger machine driver probe from device driver as suggested by Nicolin. > > Changes since V2: > 1. Moved "dais" node from machine driver DTS node to device driver DTS node > as suggested by Rob. > > Changes since V1: > 1. Original patch split into distinct patches for the device driver and > DT binding documentation. > 2. Replaced AMIX with AUDMIX in both code and file names as it looks more > RM-compliant. > 3. Removed polarity control from CPU DAI driver as suggested by Nicolin. > 4. Added machine driver and related DT binding documentation. > > Viorel Suman (3): > ASoC: fsl: Add Audio Mixer CPU DAI driver > ASoC: add fsl_audmix DT binding documentation > ASoC: fsl: Add Audio Mixer machine driver > > .../devicetree/bindings/sound/fsl,audmix.txt | 50 ++ > sound/soc/fsl/Kconfig | 16 + > sound/soc/fsl/Makefile | 5 + > sound/soc/fsl/fsl_audmix.c | 578 +++++++++++++++++++++ > sound/soc/fsl/fsl_audmix.h | 102 ++++ > sound/soc/fsl/imx-audmix.c | 327 ++++++++++++ > 6 files changed, 1078 insertions(+) > create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt > create mode 100644 sound/soc/fsl/fsl_audmix.c > create mode 100644 sound/soc/fsl/fsl_audmix.h > create mode 100644 sound/soc/fsl/imx-audmix.c > > -- > 2.7.4 >

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Re: [alsa-devel] [PATCH] ASoC: fsl_esai: fix register setting issue in RIGHT_J mode
by Nicolin Chen 15 Feb '19

15 Feb '19

On Fri, Feb 15, 2019 at 11:04:38AM +0000, S.j. Wang wrote: > The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to > wrong register, correct it. > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Would need this for stable kernel too. Ackedy-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Thanks. > --- > sound/soc/fsl/fsl_esai.c | 7 ++++--- > 1 file changed, 4 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c > index 57b484768a58..afe67c865330 100644 > --- a/sound/soc/fsl/fsl_esai.c > +++ b/sound/soc/fsl/fsl_esai.c > @@ -398,7 +398,8 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) > break; > case SND_SOC_DAIFMT_RIGHT_J: > /* Data on rising edge of bclk, frame high, right aligned */ > - xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; > + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; > + xcr |= ESAI_xCR_xWA; > break; > case SND_SOC_DAIFMT_DSP_A: > /* Data on rising edge of bclk, frame high, 1clk before data */ > @@ -455,12 +456,12 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) > return -EINVAL; > } > > - mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; > + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR | ESAI_xCR_xWA; > regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); > regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); > > mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | > - ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; > + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; > regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); > regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); > > -- > 1.9.1 >

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[alsa-devel] Applied "ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag" to the asoc tree
by Mark Brown 15 Feb '19

15 Feb '19

The patch ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 76d9c68b360f852e784170f10cb431e4713c7d0b Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 16:45:55 +0100 Subject: [PATCH] ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag There is now no users of this flag so remove it together with related code. The chan_name field of snd_dmaengine_dai_dma_data data structure is not removed as it is still in use by the PXA platform. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/dmaengine_pcm.h | 4 ---- sound/soc/soc-generic-dmaengine-pcm.c | 21 ++++----------------- 2 files changed, 4 insertions(+), 21 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 2c4cfaa135a6..c679f6116580 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -99,10 +99,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) -/* - * The PCM streams have custom channel names specified. - */ -#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 1b44e363c50c..f1ab6285a085 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -265,7 +265,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) struct dmaengine_pcm *pcm = soc_component_to_pcm(component); const struct snd_dmaengine_pcm_config *config = pcm->config; struct device *dev = component->dev; - struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; size_t prealloc_buffer_size; size_t max_buffer_size; @@ -285,19 +284,9 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (!pcm->chan[i] && - ((pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME) || - (config && config->chan_names[i]))) { - const char *chan_name = dma_data->chan_name; - - if (config && config->chan_names[i]) - chan_name = config->chan_names[i]; - + if (!pcm->chan[i] && config && config->chan_names[i]) pcm->chan[i] = dma_request_slave_channel(dev, - chan_name); - } + config->chan_names[i]); if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, @@ -420,10 +409,8 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, const char *name; struct dma_chan *chan; - if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || - (!dev->of_node && !(config && config->dma_dev && - config->dma_dev->of_node))) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || (!dev->of_node && + !(config && config->dma_dev && config->dma_dev->of_node))) return 0; if (config && config->dma_dev) { -- 2.20.1

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[alsa-devel] [PATCH] ASoC: samsung: i2s: Prevent potential NULL platform data dereference
by Sylwester Nawrocki 15 Feb '19

15 Feb '19

When np is NULL i2s_pdata could also be NULL but i2s_pdata is now being dereferenced without proper check. Fix this and shorten the error message so we don't exceed 80 characters limit. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 03fff1c657be..02472f576e17 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1369,7 +1369,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) struct i2s_dai *pri_dai, *sec_dai = NULL; struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; struct resource *res; - u32 regs_base, quirks = 0, idma_addr = 0; + u32 regs_base, idma_addr = 0; struct device_node *np = pdev->dev.of_node; const struct samsung_i2s_dai_data *i2s_dai_data; int num_dais, ret; @@ -1389,11 +1389,19 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - quirks = np ? i2s_dai_data->quirks : i2s_pdata->type.quirks; - num_dais = (quirks & QUIRK_SEC_DAI) ? 2 : 1; + if (np) { + priv->quirks = i2s_dai_data->quirks; + } else { + if (!i2s_pdata) { + dev_err(&pdev->dev, "Missing platform data\n"); + return -EINVAL; + } + priv->quirks = i2s_pdata->type.quirks; + } + + num_dais = (priv->quirks & QUIRK_SEC_DAI) ? 2 : 1; priv->pdev = pdev; priv->variant_regs = i2s_dai_data->i2s_variant_regs; - priv->quirks = quirks; ret = i2s_alloc_dais(priv, i2s_dai_data, num_dais); if (ret < 0) @@ -1405,11 +1413,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) spin_lock_init(&priv->pcm_lock); if (!np) { - if (i2s_pdata == NULL) { - dev_err(&pdev->dev, "Can't work without s3c_audio_pdata\n"); - return -EINVAL; - } - pri_dai->dma_playback.filter_data = i2s_pdata->dma_playback; pri_dai->dma_capture.filter_data = i2s_pdata->dma_capture; pri_dai->filter = i2s_pdata->dma_filter; @@ -1418,7 +1421,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } else { if (of_property_read_u32(np, "samsung,idma-addr", &idma_addr)) { - if (quirks & QUIRK_SUPPORTS_IDMA) { + if (priv->quirks & QUIRK_SUPPORTS_IDMA) { dev_info(&pdev->dev, "idma address is not"\ "specified"); } @@ -1451,7 +1454,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_capture.addr_width = 4; pri_dai->priv = priv; - if (quirks & QUIRK_PRI_6CHAN) + if (priv->quirks & QUIRK_PRI_6CHAN) pri_dai->drv->playback.channels_max = 6; ret = samsung_asoc_dma_platform_register(&pdev->dev, pri_dai->filter, @@ -1459,7 +1462,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (ret < 0) goto err_disable_clk; - if (quirks & QUIRK_SEC_DAI) { + if (priv->quirks & QUIRK_SEC_DAI) { sec_dai = &priv->dai[SAMSUNG_I2S_ID_SECONDARY - 1]; sec_dai->dma_playback.addr = regs_base + I2STXDS; -- 2.17.1

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Re: [alsa-devel] [PATCH] ASoC: simple-card: fixup refcount_t underflow
by Mark Brown 15 Feb '19

15 Feb '19

On Fri, Feb 15, 2019 at 06:21:04PM +0100, Takashi Iwai wrote: > Actually it's not all. As mentioned in my earlier post, there are > more of refcount unbalance in the code. Lots of them, really. > Below is a patch I've fixed quickly over the whole scanning the sound > tree. Some might be incorrect, so more detailed review for each piece > of change is needed... Extra references aren't such a problem for practical systems since we don't actually ever free any of the DT (there's no mainline way to use overlays yet...), that's how none of this stuff ever gets noticed. I'm not surprised there's loads of missing frees TBH. In contrast if we've got extra frees they will get noticed and cause problems.

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[alsa-devel] [regression] snd_soc_simple_card: refcount_t: underflow; use-after-free.
by Vicente Bergas 15 Feb '19

15 Feb '19

Hi, in linux 5.0-rc there is a regression regarding snd_soc_simple_card. Since updated from 4.20 there is a new error appearing in the kernel log, although sound works fine and the system is stable. The issue has not been bisected, but it probably lies at or arround this commit: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… The DT node referred in the trace below is: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… The kernel comes from the ArchLinuxArm distribution, its configuration is: https://archlinuxarm.org/packages/aarch64/linux-aarch64-rc/files/config This trace is from "dmesg -tl1,2,3,4": OF: ERROR: Bad of_node_put() on /hdmi-sound CPU: 2 PID: 370 Comm: kworker/2:2 Tainted: G C 5.0.0-rc6-1-ARCH #1 Hardware name: Sapphire-RK3399 Board (DT) Workqueue: events deferred_probe_work_func Call trace: dump_backtrace+0x0/0x1b8 show_stack+0x24/0x30 dump_stack+0x98/0xbc of_node_release+0xd0/0xd8 kobject_put+0x8c/0x1f0 of_node_put+0x24/0x30 __of_get_next_child+0x50/0x70 of_get_next_child+0x64/0x90 asoc_simple_card_probe+0xe4/0x6b0 [snd_soc_simple_card] platform_drv_probe+0x58/0xa8 really_probe+0x1f0/0x3d8 driver_probe_device+0xe4/0x138 __device_attach_driver+0xb4/0x140 bus_for_each_drv+0x8c/0xd8 __device_attach+0xdc/0x158 device_initial_probe+0x24/0x30 bus_probe_device+0x9c/0xa8 deferred_probe_work_func+0xa0/0xf0 process_one_work+0x1ac/0x400 worker_thread+0x50/0x488 kthread+0x130/0x138 ret_from_fork+0x10/0x1c ------------[ cut here ]------------ refcount_t: underflow; use-after-free. WARNING: CPU: 2 PID: 370 at lib/refcount.c:187 refcount_sub_and_test_checked+0xb8/0xd0 Modules linked in: snd_soc_hdmi_codec rockchip_vpu(C+) rockchip_rga videobuf2_dma_contig videobuf2_dma_sg v4l2_mem2mem videobuf2_vmalloc videobuf2_memops videobuf2_v4l2 videobuf2_common rc_cec snd_soc_simple_card realtek snd_soc_rockchip_i2s snd_soc_simple_card_utils snd_soc_rockchip_pcm dw_hdmi_cec dw_hdmi_i2s_audio dw_wdt videodev rtc_rk808 media hid_kensington dwmac_rk rockchip_saradc rockchip_thermal stmmac_platform stmmac squashfs loop crypto_user gpio_keys rockchipdrm analogix_dp dw_hdmi cec rc_core dw_mipi_dsi drm_kms_helper syscopyarea sysfillrect sysimgblt fb_sys_fops drm drm_panel_orientation_quirks CPU: 2 PID: 370 Comm: kworker/2:2 Tainted: G C 5.0.0-rc6-1-ARCH #1 Hardware name: Sapphire-RK3399 Board (DT) Workqueue: events deferred_probe_work_func pstate: 80000085 (Nzcv daIf -PAN -UAO) pc : refcount_sub_and_test_checked+0xb8/0xd0 lr : refcount_sub_and_test_checked+0xb8/0xd0 sp : ffff000012d9ba20 x29: ffff000012d9ba20 x28: 0000000000000000 x27: 0000000000000002 x26: 0000000000000001 x25: ffff0000115ad6c8 x24: ffff0000090bb428 x23: ffff8000f781a740 x22: 0000000000000000 x21: ffff8000f781a740 x20: ffff8000f781a740 x19: ffff8000f781a790 x18: 0000000000000000 x17: 0000000000000000 x16: 0000000000000000 x15: ffffffffffffffff x14: 0000000000000003 x13: 0000000000000000 x12: ffff000011810000 x11: ffff0000115d6000 x10: ffff000011810f48 x9 : 0000000000000000 x8 : ffff00001181f170 x7 : 0000000000000000 x6 : 0000000000000001 x5 : 0000000000000000 x4 : 0000000000000001 x3 : 0000000000000007 x2 : 0000000000000007 x1 : 86b2dbbfc7425b00 x0 : 0000000000000000 Call trace: refcount_sub_and_test_checked+0xb8/0xd0 refcount_dec_and_test_checked+0x14/0x20 kobject_put+0x24/0x1f0 of_node_put+0x24/0x30 __of_get_next_child+0x50/0x70 of_get_next_child+0x64/0x90 asoc_simple_card_probe+0x544/0x6b0 [snd_soc_simple_card] platform_drv_probe+0x58/0xa8 really_probe+0x1f0/0x3d8 driver_probe_device+0xe4/0x138 __device_attach_driver+0xb4/0x140 bus_for_each_drv+0x8c/0xd8 __device_attach+0xdc/0x158 device_initial_probe+0x24/0x30 bus_probe_device+0x9c/0xa8 deferred_probe_work_func+0xa0/0xf0 process_one_work+0x1ac/0x400 worker_thread+0x50/0x488 kthread+0x130/0x138 ret_from_fork+0x10/0x1c ---[ end trace ae290e9394a14a2f ]--- asoc-simple-card hdmi-sound: ASoC: no DMI vendor name! Regards, Vicente.

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[alsa-devel] [PATCH v4] ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag
by Sylwester Nawrocki 14 Feb '19

14 Feb '19

There is now no users of this flag so remove it together with related code. The chan_name field of snd_dmaengine_dai_dma_data data structure is not removed as it is still in use by the PXA platform. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Changes since v3: - dropped removal of chan_name field. include/sound/dmaengine_pcm.h | 4 ---- sound/soc/soc-generic-dmaengine-pcm.c | 21 ++++----------------- 2 files changed, 4 insertions(+), 21 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 2c4cfaa135a6..c679f6116580 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -99,10 +99,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) -/* - * The PCM streams have custom channel names specified. - */ -#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 1b44e363c50c..f1ab6285a085 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -265,7 +265,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) struct dmaengine_pcm *pcm = soc_component_to_pcm(component); const struct snd_dmaengine_pcm_config *config = pcm->config; struct device *dev = component->dev; - struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; size_t prealloc_buffer_size; size_t max_buffer_size; @@ -285,19 +284,9 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (!pcm->chan[i] && - ((pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME) || - (config && config->chan_names[i]))) { - const char *chan_name = dma_data->chan_name; - - if (config && config->chan_names[i]) - chan_name = config->chan_names[i]; - + if (!pcm->chan[i] && config && config->chan_names[i]) pcm->chan[i] = dma_request_slave_channel(dev, - chan_name); - } + config->chan_names[i]); if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, @@ -420,10 +409,8 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, const char *name; struct dma_chan *chan; - if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || - (!dev->of_node && !(config && config->dma_dev && - config->dma_dev->of_node))) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || (!dev->of_node && + !(config && config->dma_dev && config->dma_dev->of_node))) return 0; if (config && config->dma_dev) { -- 2.17.1

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[alsa-devel] Applied "ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From eb540d3988d93d2b0231a5b36012aa0b3abaec81 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:44 +0100 Subject: [PATCH] ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage If the probe call is on the primary DAI we can use 'other' in place of i2s->sec_dai, if the probe call is on the secondary DAI we can use 'i2s' in place of other->sec_dai. While at it fix one whitespace issue. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 6ab99e38e6dd..967c1b22ac35 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1057,18 +1057,17 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(dai->dev); if (is_secondary(i2s)) { /* If this is probe on the secondary DAI */ - snd_soc_dai_init_dma_data(dai, &other->sec_dai->dma_playback, - NULL); + snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, NULL); } else { snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, - &i2s->dma_capture); + &i2s->dma_capture); if (priv->quirks & QUIRK_NEED_RSTCLR) writel(CON_RSTCLR, priv->addr + I2SCON); if (priv->quirks & QUIRK_SUPPORTS_IDMA) idma_reg_addr_init(priv->addr, - i2s->sec_dai->idma_playback.addr); + other->idma_playback.addr); } /* Reset any constraint on RFS and BFS */ -- 2.20.1

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[alsa-devel] Applied "ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c5ba619247391527248c4a8fb27e68f7cece8d0d Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:45 +0100 Subject: [PATCH] ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition Change indentation so this macro definition spans 2 rows and looks more consistent with surrounding code. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 967c1b22ac35..07f815a730df 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1150,9 +1150,8 @@ static const struct snd_soc_component_driver samsung_i2s_component = { .num_dapm_routes = ARRAY_SIZE(samsung_i2s_dapm_routes), }; -#define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \ - SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE) +#define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) static int i2s_alloc_dais(struct samsung_i2s_priv *priv, const struct samsung_i2s_dai_data *i2s_dai_data, -- 2.20.1

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[alsa-devel] Applied "ASoC: samsung: i2s: Comments clean up" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Comments clean up has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 9f9f8a5b79b0855d162153de41ffda687fd2241f Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:46 +0100 Subject: [PATCH] ASoC: samsung: i2s: Comments clean up Spelling error fixes, upper/lower case letter changes. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 29 +++++++++++++++-------------- 1 file changed, 15 insertions(+), 14 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 07f815a730df..84cfa2c0ba68 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1,5 +1,4 @@ -/* sound/soc/samsung/i2s.c - * +/* * ALSA SoC Audio Layer - Samsung I2S Controller driver * * Copyright (c) 2010 Samsung Electronics Co. Ltd. @@ -61,10 +60,10 @@ struct i2s_dai { /* Platform device for this DAI */ struct platform_device *pdev; - /* Frame Clock */ + /* Frame clock */ unsigned frmclk; /* - * Specifically requested RCLK,BCLK by MACHINE Driver. + * Specifically requested RCLK, BCLK by machine driver. * 0 indicates CPU driver is free to choose any value. */ unsigned rfs, bfs; @@ -72,8 +71,9 @@ struct i2s_dai { struct i2s_dai *pri_dai; /* Pointer to the Secondary_Fifo if it has one, NULL otherwise */ struct i2s_dai *sec_dai; -#define DAI_OPENED (1 << 0) /* Dai is opened */ -#define DAI_MANAGER (1 << 1) /* Dai is the manager */ + +#define DAI_OPENED (1 << 0) /* DAI is opened */ +#define DAI_MANAGER (1 << 1) /* DAI is the manager */ unsigned mode; /* Driver for this DAI */ @@ -98,7 +98,7 @@ struct samsung_i2s_priv { /* Spinlock protecting access to the device's registers */ spinlock_t lock; - /* Lock for cross i/f checks */ + /* Lock for cross interface checks */ spinlock_t pcm_lock; /* CPU DAIs and their corresponding drivers */ @@ -309,7 +309,7 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) writel(mod, priv->addr + I2SMOD); } -/* Read Bit-Clock of I2S (in multiples of LRCLK) */ +/* Read bit-clock of I2S (in multiples of LRCLK) */ static inline unsigned get_bfs(struct i2s_dai *i2s) { struct samsung_i2s_priv *priv = i2s->priv; @@ -331,7 +331,7 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) } } -/* Write Bit-Clock of I2S (in multiples of LRCLK) */ +/* Write bit-clock of I2S (in multiples of LRCLK) */ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { struct samsung_i2s_priv *priv = i2s->priv; @@ -383,7 +383,7 @@ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) writel(mod, priv->addr + I2SMOD); } -/* Sample-Size */ +/* Sample size */ static inline int get_blc(struct i2s_dai *i2s) { int blc = readl(i2s->priv->addr + I2SMOD); @@ -397,7 +397,7 @@ static inline int get_blc(struct i2s_dai *i2s) } } -/* TX Channel Control */ +/* TX channel control */ static void i2s_txctrl(struct i2s_dai *i2s, int on) { struct samsung_i2s_priv *priv = i2s->priv; @@ -742,7 +742,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, switch (params_channels(params)) { case 6: val |= MOD_DC2_EN; - /* fall through */ + /* Fall through */ case 4: val |= MOD_DC1_EN; break; @@ -821,7 +821,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -/* We set constraints on the substream acc to the version of I2S */ +/* We set constraints on the substream according to the version of I2S */ static int i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1056,7 +1056,8 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(dai->dev); - if (is_secondary(i2s)) { /* If this is probe on the secondary DAI */ + if (is_secondary(i2s)) { + /* If this is probe on the secondary DAI */ snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, NULL); } else { snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, -- 2.20.1

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[alsa-devel] [PATCH v3 Resend 00/13] ASoC: dmaengine updates, secondary CPU DAI for Odroid
by Sylwester Nawrocki 14 Feb '19

14 Feb '19

This is a rebased version with the build break fixed and added Krzysztof's Acks. My apologies, I double checked there is no build errors between patches this time. Below is cover letter text from v2. This series restores support for the secondary CPU DAI in samsung/i2s.c driver and adds support for secondary CPU DAI on Odroid boards. I have added 4 more clean up patches comparing to first version of the patch set (16/22...19/22). The first patch adds new platform device for the secondary PCM, I tried to avoid registering new platform device in probe() as it doesn't play well with deferred probing but couldn't find a way to avoid that new device, unique devices are needed to register "PCM" components for each CPU DAI. Patches 2/21...11/21 are mostly a clean up, following conversion to single component with multiple DAIs, allowing to reference both primary/secondary DAI in DT. The patch set has been tested on Odroid U3, XU3, XU4 and Mini2440 (s3c2440 non-DT) boards. Changes since v1 ((addressing Krzysztof's review comments): - added a comment, whitespace change in patch 1/21 - dropped unnecessary new line addition from patch 1/21 Sylwester Nawrocki (13): ASoC: samsung: i2s: Move SFR pointer to common driver data structure ASoC: samsung: i2s: Drop spinlock pointer from i2s_dai data structure ASoC: samsung: i2s: Move IP variant data to common driver data structure ASoC: samsung: i2s: Move quirks data to common driver data structure ASoC: samsung: i2s: Get rid of a static spinlock ASoC: samsung: odroid: Add support for secondary CPU DAI ASoC: samsung: Specify DMA channel names through custom DMA config ASoC: samsung: Drop DAI DMA data chan_name assignments ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition ASoC: samsung: i2s: Comments clean up ASoC: samsung: i2s: Convert to SPDX License Indentifier include/sound/dmaengine_pcm.h | 6 - sound/soc/samsung/dmaengine.c | 12 +- sound/soc/samsung/i2s.c | 297 +++++++++++++------------- sound/soc/samsung/odroid.c | 131 ++++++++---- sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/soc-generic-dmaengine-pcm.c | 21 +- 7 files changed, 250 insertions(+), 225 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH v2 0/2] ASoC: da7219: BCLK and TDM flexibility improvements
by Adam Thomson 14 Feb '19

14 Feb '19

This patch set updates the driver to be more flexible regarding the BCLK and TDM configurations of the device. Previously the BCLK rate was fixed at 64 periods per WCLK, when the codec is DAI clock master, but to satisfy devices which prefer a lower rate, and to save power, the BCLK rate is now calculated based on hw_params() data. As a fallout of the BCLK efforts it was also apparent that the TDM code could be made more flexible as well. The device is capable of automatically detecting the BCLK rate when it is clock slave, so previous limitations imposed were not necessary. The mask handling is now used as an input to determine the BCLK offset to align closer with other examples within ALSA. Changes in v2: - Update 0001 patch to use '&&' instead of '&' in hw_params() to decide if BCLK rate should be configured or not. Adam Thomson (2): ASoC: da7219: Add support for master mode BCLK rate adjustment ASoC: da7219: Update TDM usage to be more flexible sound/soc/codecs/da7219.c | 116 +++++++++++++++++++++++++++++----------------- sound/soc/codecs/da7219.h | 1 + 2 files changed, 74 insertions(+), 43 deletions(-) -- 1.9.1

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[alsa-devel] Applied "ASoC: da7219: Update TDM usage to be more flexible" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: da7219: Update TDM usage to be more flexible has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 541ccdc113f000d51858ee7e135889e4096a3316 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Thu, 14 Feb 2019 10:13:30 +0000 Subject: [PATCH] ASoC: da7219: Update TDM usage to be more flexible The previous implementatation was restrictive with regards to BCLK rates for slave mode where the driver would not allow rates the codec couldn't provide itself as clock master. The codec is able to automatically determine and handle whatever rate is provided so this restriction isn't necessary for slave mode. The code was also flawed with regards to setting of the frame offset as using rx_mask to explicitly set the offset has the knock on effect of impacting the min and max channels for the codec, in soc_pcm_hw_params() through the call to soc_pcm_codec_params_fixup(). With this update, the driver now only limits frame size if codec is clock master, and dynamically determines the BCLK offset relating to WCLK using the tx_mask for slot offset along with the slot width provided. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7219.c | 80 +++++++++++++++++++++++---------------- 1 file changed, 47 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c599aa9f609b..121a8190f93e 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1391,8 +1391,10 @@ static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, { struct snd_soc_component *component = dai->component; struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); - u8 dai_bclks_per_wclk; - __le16 offset; + unsigned int ch_mask; + u8 dai_bclks_per_wclk, slot_offset; + u16 offset; + __le16 dai_offset; u32 frame_size; /* No channels enabled so disable TDM */ @@ -1405,51 +1407,63 @@ static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, } /* Check we have valid slots */ - if (fls(tx_mask) > DA7219_DAI_TDM_MAX_SLOTS) { - dev_err(component->dev, "Invalid number of slots, max = %d\n", + slot_offset = ffs(tx_mask) - 1; + ch_mask = (tx_mask >> slot_offset); + if (fls(ch_mask) > DA7219_DAI_TDM_MAX_SLOTS) { + dev_err(component->dev, + "Invalid number of slots, max = %d\n", DA7219_DAI_TDM_MAX_SLOTS); return -EINVAL; } - /* Check we have a valid offset given */ - if (rx_mask > DA7219_DAI_OFFSET_MAX) { - dev_err(component->dev, "Invalid slot offset, max = %d\n", - DA7219_DAI_OFFSET_MAX); + /* + * Ensure we have a valid offset into the frame, based on slot width + * and slot offset of first slot we're interested in. + */ + offset = slot_offset * slot_width; + if (offset > DA7219_DAI_OFFSET_MAX) { + dev_err(component->dev, "Invalid frame offset %d\n", offset); return -EINVAL; } - /* Calculate & validate frame size based on slot info provided. */ - frame_size = slots * slot_width; - switch (frame_size) { - case 32: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; - break; - case 64: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; - break; - case 128: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128; - break; - case 256: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256; - break; - default: - dev_err(component->dev, "Invalid frame size %d\n", frame_size); - return -EINVAL; - } + /* + * If we're master, calculate & validate frame size based on slot info + * provided as we have a limited set of rates available. + */ + if (da7219->master) { + frame_size = slots * slot_width; + switch (frame_size) { + case 32: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; + break; + case 64: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; + break; + case 128: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128; + break; + case 256: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256; + break; + default: + dev_err(component->dev, "Invalid frame size %d\n", + frame_size); + return -EINVAL; + } - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, - DA7219_DAI_BCLKS_PER_WCLK_MASK, - dai_bclks_per_wclk); + snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + dai_bclks_per_wclk); + } - offset = cpu_to_le16(rx_mask); + dai_offset = cpu_to_le16(offset); regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER, - &offset, sizeof(offset)); + &dai_offset, sizeof(dai_offset)); snd_soc_component_update_bits(component, DA7219_DAI_TDM_CTRL, DA7219_DAI_TDM_CH_EN_MASK | DA7219_DAI_TDM_MODE_EN_MASK, - (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) | + (ch_mask << DA7219_DAI_TDM_CH_EN_SHIFT) | DA7219_DAI_TDM_MODE_EN_MASK); da7219->tdm_en = true; -- 2.20.1

1 0

[alsa-devel] [PATCH v2 00/20] ASoC: Sound Open Firmware (SOF) - Intel support
by Pierre-Louis Bossart 14 Feb '19

14 Feb '19

This patch series introduces the support for Intel devices (Baytrail, CherryTrail, Broadwell/Haswell, ApolloLake, GeminiLake, CannonLake, IceLake). Support for Skylake and KabyLake is incomplete at the moment but the basic building blocks required are already present. The series starts with the 'legacy' devices, then all the more recent ones and concludes with ACPI/PCI hooks. SKL+ platforms can be compiled without HDaudio link/codec support, in which case the hdac library is not used. When support for HDMI or HDaudio codecs is selected, parts of this library are used. The machine drivers supported by SOF are not provided as part of this series. The build will only work in 'nocodec' mode for now. Support for SPI-based devices, compressed streams is also not provided for now since it's not ready for upstream just yet. Thank you for reviews and comments, we appreciate the time spent commenting on this large patchset. Thanks in particular to Alan Cox and Andy Shevchenko for their comments on an earlier version. Changes since v2: Addressed dozens of comments from Takashi Iwai, Mark Brown, Andy Shevchenko, Daniel Baluta (Thanks!) Transition to new display power management Fixed suspend issues Optimized register polling, better handling of timeouts and firmware download retries Detect number of HDaudio codec and select machine driver accordingly Added workqueue based probe for HDaudio (dependency on modules) Fixed platform name override and conflicts with DT deferred probe changes Added prefix for firmware and topology files and module arguments Split Baytrail and Edison Fixed warnings (smatch, cppcheck) Keyon Jie (2): ASoC: SOF: Intel: Add hda-bus support and initialization ASoC: SOF: Intel: SKL, CNL, APL platform DAIs Liam Girdwood (15): ASoC: SOF: Intel: Add BYT, CHT and BSW DSP HW support. ASoC: SOF: Intel: Add HSW HW DSP support ASoC: SOF: Intel: Add BDW HW DSP support ASoC: SOF: Intel: Add APL/CNL HW DSP support ASoC: SOF: Intel: Add HDA controller for Intel DSP ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations ASoC: SOF: Intel: Add Intel specific HDA IPC mechanisms. ASoC: SOF: Intel: Add Intel specific HDA firmware loader ASoC: SOF: Intel: Add Intel specific HDA PCM operations ASoC: SOF: Intel: Add Intel specific HDA stream operations ASoC: SOF: Intel: Add Intel specific HDA trace operations ASoC: SOF: Intel: Add platform differentiation for SKL, APL and CNL ASoC: SOF: Add ACPI device support ASoC: SOF: Add PCI device support ASoC: SOF: Add Build support for SOF core and Intel drivers Pierre-Louis Bossart (3): ASoC: SOF: Intel: Add support for HDAudio codecs ASoC: SOF: Intel: Add SKL-specific code loader ASoC: Intel: Kconfig: expose common option between SST and SOF drivers sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/intel/Kconfig | 7 +- sound/soc/sof/Kconfig | 141 +++++ sound/soc/sof/Makefile | 18 + sound/soc/sof/intel/Kconfig | 253 ++++++++ sound/soc/sof/intel/Makefile | 17 + sound/soc/sof/intel/apl.c | 111 ++++ sound/soc/sof/intel/bdw.c | 716 +++++++++++++++++++++ sound/soc/sof/intel/byt.c | 893 +++++++++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 269 ++++++++ sound/soc/sof/intel/hda-bus.c | 108 ++++ sound/soc/sof/intel/hda-codec.c | 171 +++++ sound/soc/sof/intel/hda-ctrl.c | 165 +++++ sound/soc/sof/intel/hda-dai.c | 362 +++++++++++ sound/soc/sof/intel/hda-dsp.c | 440 +++++++++++++ sound/soc/sof/intel/hda-ipc.c | 403 ++++++++++++ sound/soc/sof/intel/hda-loader-skl.c | 526 ++++++++++++++++ sound/soc/sof/intel/hda-loader.c | 359 +++++++++++ sound/soc/sof/intel/hda-pcm.c | 235 +++++++ sound/soc/sof/intel/hda-stream.c | 663 ++++++++++++++++++++ sound/soc/sof/intel/hda-trace.c | 98 +++ sound/soc/sof/intel/hda.c | 680 ++++++++++++++++++++ sound/soc/sof/intel/hda.h | 549 ++++++++++++++++ sound/soc/sof/intel/hsw.c | 717 +++++++++++++++++++++ sound/soc/sof/intel/shim.h | 177 ++++++ sound/soc/sof/intel/skl.c | 104 ++++ sound/soc/sof/sof-acpi-dev.c | 310 ++++++++++ sound/soc/sof/sof-pci-dev.c | 351 +++++++++++ 29 files changed, 8844 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/intel/Kconfig create mode 100644 sound/soc/sof/intel/Makefile create mode 100644 sound/soc/sof/intel/apl.c create mode 100644 sound/soc/sof/intel/bdw.c create mode 100644 sound/soc/sof/intel/byt.c create mode 100644 sound/soc/sof/intel/cnl.c create mode 100644 sound/soc/sof/intel/hda-bus.c create mode 100644 sound/soc/sof/intel/hda-codec.c create mode 100644 sound/soc/sof/intel/hda-ctrl.c create mode 100644 sound/soc/sof/intel/hda-dai.c create mode 100644 sound/soc/sof/intel/hda-dsp.c create mode 100644 sound/soc/sof/intel/hda-ipc.c create mode 100644 sound/soc/sof/intel/hda-loader-skl.c create mode 100644 sound/soc/sof/intel/hda-loader.c create mode 100644 sound/soc/sof/intel/hda-pcm.c create mode 100644 sound/soc/sof/intel/hda-stream.c create mode 100644 sound/soc/sof/intel/hda-trace.c create mode 100644 sound/soc/sof/intel/hda.c create mode 100644 sound/soc/sof/intel/hda.h create mode 100644 sound/soc/sof/intel/hsw.c create mode 100644 sound/soc/sof/intel/shim.h create mode 100644 sound/soc/sof/intel/skl.c create mode 100644 sound/soc/sof/sof-acpi-dev.c create mode 100644 sound/soc/sof/sof-pci-dev.c -- 2.17.1

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[alsa-devel] [PATCH] ASoC: stm32: sai: add missing put_device()
by Wen Yang 14 Feb '19

14 Feb '19

The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Fixes: 7dd0d835582f ("ASoC: stm32: sai: simplify sync modes management") Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> --- sound/soc/stm/stm32_sai.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index bcb35ca..14c9591 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -112,16 +112,21 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client, if (!sai_provider) { dev_err(&sai_client->pdev->dev, "SAI sync provider data not found\n"); - return -EINVAL; + ret = -EINVAL; + goto out_put_dev; } /* Configure sync client */ ret = stm32_sai_sync_conf_client(sai_client, synci); if (ret < 0) - return ret; + goto out_put_dev; /* Configure sync provider */ - return stm32_sai_sync_conf_provider(sai_provider, synco); + ret = stm32_sai_sync_conf_provider(sai_provider, synco); + +out_put_dev: + put_device(&pdev->dev); + return ret; } static int stm32_sai_probe(struct platform_device *pdev) -- 2.7.4

3 4

[alsa-devel] Fwd: right way to enumerate pcm devices
by Joël Krähemann 14 Feb '19

14 Feb '19

---------- Forwarded message --------- From: Joël Krähemann <jkraehemann(a)gmail.com> Date: Thu, Feb 14, 2019 at 12:11 AM Subject: Re: [alsa-devel] right way to enumerate pcm devices To: <sylvain.bertrand(a)gmail.com> Hi all, Thank you for the hint about alsa hints. It was very helpful. Just updated my code: http://git.savannah.nongnu.org/cgit/gsequencer.git/tree/ags/audio/ags_devou… bests, Joël On Wed, Feb 13, 2019 at 11:44 PM <sylvain.bertrand(a)gmail.com> wrote: > > On Wed, Feb 13, 2019 at 10:25:18PM +0000, sylvain.bertrand(a)gmail.com wrote: > > ... > > Actually, I would just give a way to police the access to buffers. For > instance the user application would be in charge of handling the actual > mixing (I guess mmap-ed dma buffers). > > -- > Sylvain > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 1

[alsa-devel] Applied "ASoC: samsung: i2s: Move SFR pointer to common driver data structure" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Move SFR pointer to common driver data structure has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e2e16fa6a32dcf0a340cb2b6155a44d1bf5858ef Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:35 +0100 Subject: [PATCH] ASoC: samsung: i2s: Move SFR pointer to common driver data structure The SFR region is common for both DAIs so move related data structure field from struct i2s_dai to the common driver data structure. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 106 ++++++++++++++++++++++------------------ 1 file changed, 59 insertions(+), 47 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 72f0cb7abb30..3f8955608a94 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -60,8 +60,7 @@ struct samsung_i2s_dai_data { struct i2s_dai { /* Platform device for this DAI */ struct platform_device *pdev; - /* Memory mapped SFR region */ - void __iomem *addr; + /* Frame Clock */ unsigned frmclk; /* @@ -100,6 +99,9 @@ struct samsung_i2s_priv { struct platform_device *pdev; struct platform_device *pdev_sec; + /* Memory mapped SFR region */ + void __iomem *addr; + /* Spinlock protecting access to the device's registers */ spinlock_t spinlock; @@ -143,7 +145,9 @@ static inline bool is_secondary(struct i2s_dai *i2s) /* If operating in SoC-Slave mode */ static inline bool is_slave(struct i2s_dai *i2s) { - u32 mod = readl(i2s->addr + I2SMOD); + struct samsung_i2s_priv *priv = i2s->priv; + + u32 mod = readl(priv->addr + I2SMOD); return (mod & (1 << i2s->variant_regs->mss_off)) ? true : false; } @@ -155,7 +159,7 @@ static inline bool tx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SCON); + active = readl(i2s->priv->addr + I2SCON); if (is_secondary(i2s)) active &= CON_TXSDMA_ACTIVE; @@ -193,7 +197,7 @@ static inline bool rx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE; + active = readl(i2s->priv->addr + I2SCON) & CON_RXDMA_ACTIVE; return active ? true : false; } @@ -256,8 +260,10 @@ static inline bool is_manager(struct i2s_dai *i2s) /* Read RCLK of I2S (in multiples of LRCLK) */ static inline unsigned get_rfs(struct i2s_dai *i2s) { + struct samsung_i2s_priv *priv = i2s->priv; u32 rfs; - rfs = readl(i2s->addr + I2SMOD) >> i2s->variant_regs->rfs_off; + + rfs = readl(priv->addr + I2SMOD) >> i2s->variant_regs->rfs_off; rfs &= i2s->variant_regs->rfs_mask; switch (rfs) { @@ -275,7 +281,8 @@ static inline unsigned get_rfs(struct i2s_dai *i2s) /* Write RCLK of I2S (in multiples of LRCLK) */ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) { - u32 mod = readl(i2s->addr + I2SMOD); + struct samsung_i2s_priv *priv = i2s->priv; + u32 mod = readl(priv->addr + I2SMOD); int rfs_shift = i2s->variant_regs->rfs_off; mod &= ~(i2s->variant_regs->rfs_mask << rfs_shift); @@ -307,14 +314,16 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) break; } - writel(mod, i2s->addr + I2SMOD); + writel(mod, priv->addr + I2SMOD); } /* Read Bit-Clock of I2S (in multiples of LRCLK) */ static inline unsigned get_bfs(struct i2s_dai *i2s) { + struct samsung_i2s_priv *priv = i2s->priv; u32 bfs; - bfs = readl(i2s->addr + I2SMOD) >> i2s->variant_regs->bfs_off; + + bfs = readl(priv->addr + I2SMOD) >> i2s->variant_regs->bfs_off; bfs &= i2s->variant_regs->bfs_mask; switch (bfs) { @@ -333,7 +342,8 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) /* Write Bit-Clock of I2S (in multiples of LRCLK) */ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { - u32 mod = readl(i2s->addr + I2SMOD); + struct samsung_i2s_priv *priv = i2s->priv; + u32 mod = readl(priv->addr + I2SMOD); int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM; int bfs_shift = i2s->variant_regs->bfs_off; @@ -378,13 +388,13 @@ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) return; } - writel(mod, i2s->addr + I2SMOD); + writel(mod, priv->addr + I2SMOD); } /* Sample-Size */ static inline int get_blc(struct i2s_dai *i2s) { - int blc = readl(i2s->addr + I2SMOD); + int blc = readl(i2s->priv->addr + I2SMOD); blc = (blc >> 13) & 0x3; @@ -398,7 +408,8 @@ static inline int get_blc(struct i2s_dai *i2s) /* TX Channel Control */ static void i2s_txctrl(struct i2s_dai *i2s, int on) { - void __iomem *addr = i2s->addr; + struct samsung_i2s_priv *priv = i2s->priv; + void __iomem *addr = priv->addr; int txr_off = i2s->variant_regs->txr_off; u32 con = readl(addr + I2SCON); u32 mod = readl(addr + I2SMOD) & ~(3 << txr_off); @@ -448,7 +459,8 @@ static void i2s_txctrl(struct i2s_dai *i2s, int on) /* RX Channel Control */ static void i2s_rxctrl(struct i2s_dai *i2s, int on) { - void __iomem *addr = i2s->addr; + struct samsung_i2s_priv *priv = i2s->priv; + void __iomem *addr = priv->addr; int txr_off = i2s->variant_regs->txr_off; u32 con = readl(addr + I2SCON); u32 mod = readl(addr + I2SMOD) & ~(3 << txr_off); @@ -485,9 +497,9 @@ static inline void i2s_fifo(struct i2s_dai *i2s, u32 flush) return; if (is_secondary(i2s)) - fic = i2s->addr + I2SFICS; + fic = i2s->priv->addr + I2SFICS; else - fic = i2s->addr + I2SFIC; + fic = i2s->priv->addr + I2SFIC; /* Flush the FIFO */ writel(readl(fic) | flush, fic); @@ -516,7 +528,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs, pm_runtime_get_sync(dai->dev); spin_lock_irqsave(i2s->lock, flags); - mod = readl(i2s->addr + I2SMOD); + mod = readl(priv->addr + I2SMOD); spin_unlock_irqrestore(i2s->lock, flags); switch (clk_id) { @@ -613,9 +625,9 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs, } spin_lock_irqsave(i2s->lock, flags); - mod = readl(i2s->addr + I2SMOD); + mod = readl(priv->addr + I2SMOD); mod = (mod & ~mask) | val; - writel(mod, i2s->addr + I2SMOD); + writel(mod, priv->addr + I2SMOD); spin_unlock_irqrestore(i2s->lock, flags); done: pm_runtime_put(dai->dev); @@ -698,7 +710,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) pm_runtime_get_sync(dai->dev); spin_lock_irqsave(i2s->lock, flags); - mod = readl(i2s->addr + I2SMOD); + mod = readl(priv->addr + I2SMOD); /* * Don't change the I2S mode if any controller is active on this * channel. @@ -714,7 +726,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) mod &= ~(sdf_mask | lrp_rlow | mod_slave); mod |= tmp; - writel(mod, i2s->addr + I2SMOD); + writel(mod, priv->addr + I2SMOD); spin_unlock_irqrestore(i2s->lock, flags); pm_runtime_put(dai->dev); @@ -801,9 +813,9 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, } spin_lock_irqsave(i2s->lock, flags); - mod = readl(i2s->addr + I2SMOD); + mod = readl(priv->addr + I2SMOD); mod = (mod & ~mask) | val; - writel(mod, i2s->addr + I2SMOD); + writel(mod, priv->addr + I2SMOD); spin_unlock_irqrestore(i2s->lock, flags); snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); @@ -837,7 +849,7 @@ static int i2s_startup(struct snd_pcm_substream *substream, i2s->mode |= DAI_MANAGER; if (!any_active(i2s) && (i2s->quirks & QUIRK_NEED_RSTCLR)) - writel(CON_RSTCLR, i2s->addr + I2SCON); + writel(CON_RSTCLR, i2s->priv->addr + I2SCON); spin_unlock_irqrestore(&lock, flags); @@ -920,7 +932,7 @@ static int config_setup(struct i2s_dai *i2s) if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { psr = priv->rclk_srcrate / i2s->frmclk / rfs; - writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR); + writel(((psr - 1) << 8) | PSR_PSREN, priv->addr + I2SPSR); dev_dbg(&i2s->pdev->dev, "RCLK_SRC=%luHz PSR=%u, RCLK=%dfs, BCLK=%dfs\n", priv->rclk_srcrate, psr, rfs, bfs); @@ -1008,8 +1020,9 @@ static int i2s_set_clkdiv(struct snd_soc_dai *dai, static snd_pcm_sframes_t i2s_delay(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); - u32 reg = readl(i2s->addr + I2SFIC); + u32 reg = readl(priv->addr + I2SFIC); snd_pcm_sframes_t delay; const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs; @@ -1018,7 +1031,7 @@ i2s_delay(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) delay = FIC_RXCOUNT(reg); else if (is_secondary(i2s)) - delay = FICS_TXCOUNT(readl(i2s->addr + I2SFICS)); + delay = FICS_TXCOUNT(readl(priv->addr + I2SFICS)); else delay = (reg >> i2s_regs->ftx0cnt_off) & 0x7f; @@ -1042,6 +1055,7 @@ static int i2s_resume(struct snd_soc_dai *dai) static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = get_other_dai(i2s); unsigned long flags; @@ -1056,10 +1070,10 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) &i2s->dma_capture); if (i2s->quirks & QUIRK_NEED_RSTCLR) - writel(CON_RSTCLR, i2s->addr + I2SCON); + writel(CON_RSTCLR, priv->addr + I2SCON); if (i2s->quirks & QUIRK_SUPPORTS_IDMA) - idma_reg_addr_init(i2s->addr, + idma_reg_addr_init(priv->addr, i2s->sec_dai->idma_playback.addr); } @@ -1086,6 +1100,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); unsigned long flags; @@ -1094,7 +1109,7 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) if (!is_secondary(i2s)) { if (i2s->quirks & QUIRK_NEED_RSTCLR) { spin_lock_irqsave(i2s->lock, flags); - writel(0, i2s->addr + I2SCON); + writel(0, priv->addr + I2SCON); spin_unlock_irqrestore(i2s->lock, flags); } } @@ -1205,11 +1220,10 @@ static int i2s_alloc_dais(struct samsung_i2s_priv *priv, static int i2s_runtime_suspend(struct device *dev) { struct samsung_i2s_priv *priv = dev_get_drvdata(dev); - struct i2s_dai *i2s = samsung_i2s_get_pri_dai(dev); - priv->suspend_i2smod = readl(i2s->addr + I2SMOD); - priv->suspend_i2scon = readl(i2s->addr + I2SCON); - priv->suspend_i2spsr = readl(i2s->addr + I2SPSR); + priv->suspend_i2smod = readl(priv->addr + I2SMOD); + priv->suspend_i2scon = readl(priv->addr + I2SCON); + priv->suspend_i2spsr = readl(priv->addr + I2SPSR); if (priv->op_clk) clk_disable_unprepare(priv->op_clk); @@ -1221,7 +1235,6 @@ static int i2s_runtime_suspend(struct device *dev) static int i2s_runtime_resume(struct device *dev) { struct samsung_i2s_priv *priv = dev_get_drvdata(dev); - struct i2s_dai *i2s = samsung_i2s_get_pri_dai(dev); int ret; ret = clk_prepare_enable(priv->clk); @@ -1236,9 +1249,9 @@ static int i2s_runtime_resume(struct device *dev) } } - writel(priv->suspend_i2scon, i2s->addr + I2SCON); - writel(priv->suspend_i2smod, i2s->addr + I2SMOD); - writel(priv->suspend_i2spsr, i2s->addr + I2SPSR); + writel(priv->suspend_i2scon, priv->addr + I2SCON); + writel(priv->suspend_i2smod, priv->addr + I2SMOD); + writel(priv->suspend_i2spsr, priv->addr + I2SPSR); return 0; } @@ -1296,21 +1309,21 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { /* Activate the prescaler */ - u32 val = readl(i2s->addr + I2SPSR); - writel(val | PSR_PSREN, i2s->addr + I2SPSR); + u32 val = readl(priv->addr + I2SPSR); + writel(val | PSR_PSREN, priv->addr + I2SPSR); priv->clk_table[CLK_I2S_RCLK_SRC] = clk_register_mux(dev, i2s_clk_name[CLK_I2S_RCLK_SRC], p_names, ARRAY_SIZE(p_names), CLK_SET_RATE_NO_REPARENT | CLK_SET_RATE_PARENT, - i2s->addr + I2SMOD, reg_info->rclksrc_off, + priv->addr + I2SMOD, reg_info->rclksrc_off, 1, 0, i2s->lock); priv->clk_table[CLK_I2S_RCLK_PSR] = clk_register_divider(dev, i2s_clk_name[CLK_I2S_RCLK_PSR], i2s_clk_name[CLK_I2S_RCLK_SRC], CLK_SET_RATE_PARENT, - i2s->addr + I2SPSR, 8, 6, 0, i2s->lock); + priv->addr + I2SPSR, 8, 6, 0, i2s->lock); p_names[0] = i2s_clk_name[CLK_I2S_RCLK_PSR]; priv->clk_data.clk_num = 2; @@ -1319,7 +1332,7 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) priv->clk_table[CLK_I2S_CDCLK] = clk_register_gate(dev, i2s_clk_name[CLK_I2S_CDCLK], p_names[0], CLK_SET_RATE_PARENT, - i2s->addr + I2SMOD, reg_info->cdclkcon_off, + priv->addr + I2SMOD, reg_info->cdclkcon_off, CLK_GATE_SET_TO_DISABLE, i2s->lock); priv->clk_data.clk_num += 1; @@ -1423,9 +1436,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - pri_dai->addr = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(pri_dai->addr)) - return PTR_ERR(pri_dai->addr); + priv->addr = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(priv->addr)) + return PTR_ERR(priv->addr); regs_base = res->start; @@ -1472,7 +1485,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.addr_width = 4; - sec_dai->addr = pri_dai->addr; sec_dai->quirks = quirks; sec_dai->idma_playback.addr = idma_addr; sec_dai->pri_dai = pri_dai; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: i2s: Drop spinlock pointer from i2s_dai data structure" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Drop spinlock pointer from i2s_dai data structure has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 9d7939c929413d0f9effef599a0ca73300b494be Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:36 +0100 Subject: [PATCH] ASoC: samsung: i2s: Drop spinlock pointer from i2s_dai data structure As we now have the 'priv' pointer in most of the places we can use priv->lock directly, dropping extra indirection in the SFR region spinlock access. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 51 +++++++++++++++++++---------------------- 1 file changed, 24 insertions(+), 27 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 3f8955608a94..cf8615e23879 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -87,8 +87,6 @@ struct i2s_dai { u32 quirks; const struct samsung_i2s_variant_regs *variant_regs; - spinlock_t *lock; - struct samsung_i2s_priv *priv; }; @@ -103,7 +101,7 @@ struct samsung_i2s_priv { void __iomem *addr; /* Spinlock protecting access to the device's registers */ - spinlock_t spinlock; + spinlock_t lock; /* CPU DAIs and their corresponding drivers */ struct i2s_dai *dai; @@ -527,9 +525,9 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs, pm_runtime_get_sync(dai->dev); - spin_lock_irqsave(i2s->lock, flags); + spin_lock_irqsave(&priv->lock, flags); mod = readl(priv->addr + I2SMOD); - spin_unlock_irqrestore(i2s->lock, flags); + spin_unlock_irqrestore(&priv->lock, flags); switch (clk_id) { case SAMSUNG_I2S_OPCLK: @@ -624,11 +622,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs, goto err; } - spin_lock_irqsave(i2s->lock, flags); + spin_lock_irqsave(&priv->lock, flags); mod = readl(priv->addr + I2SMOD); mod = (mod & ~mask) | val; writel(mod, priv->addr + I2SMOD); - spin_unlock_irqrestore(i2s->lock, flags); + spin_unlock_irqrestore(&priv->lock, flags); done: pm_runtime_put(dai->dev); @@ -709,7 +707,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } pm_runtime_get_sync(dai->dev); - spin_lock_irqsave(i2s->lock, flags); + spin_lock_irqsave(&priv->lock, flags); mod = readl(priv->addr + I2SMOD); /* * Don't change the I2S mode if any controller is active on this @@ -717,7 +715,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) */ if (any_active(i2s) && ((mod & (sdf_mask | lrp_rlow | mod_slave)) != tmp)) { - spin_unlock_irqrestore(i2s->lock, flags); + spin_unlock_irqrestore(&priv->lock, flags); pm_runtime_put(dai->dev); dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); @@ -727,7 +725,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) mod &= ~(sdf_mask | lrp_rlow | mod_slave); mod |= tmp; writel(mod, priv->addr + I2SMOD); - spin_unlock_irqrestore(i2s->lock, flags); + spin_unlock_irqrestore(&priv->lock, flags); pm_runtime_put(dai->dev); return 0; @@ -812,11 +810,11 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - spin_lock_irqsave(i2s->lock, flags); + spin_lock_irqsave(&priv->lock, flags); mod = readl(priv->addr + I2SMOD); mod = (mod & ~mask) | val; writel(mod, priv->addr + I2SMOD); - spin_unlock_irqrestore(i2s->lock, flags); + spin_unlock_irqrestore(&priv->lock, flags); snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); @@ -944,6 +942,7 @@ static int config_setup(struct i2s_dai *i2s) static int i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); struct snd_soc_pcm_runtime *rtd = substream->private_data; struct i2s_dai *i2s = to_info(rtd->cpu_dai); @@ -954,10 +953,10 @@ static int i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: pm_runtime_get_sync(dai->dev); - spin_lock_irqsave(i2s->lock, flags); + spin_lock_irqsave(&priv->lock, flags); if (config_setup(i2s)) { - spin_unlock_irqrestore(i2s->lock, flags); + spin_unlock_irqrestore(&priv->lock, flags); return -EINVAL; } @@ -966,12 +965,12 @@ static int i2s_trigger(struct snd_pcm_substream *substream, else i2s_txctrl(i2s, 1); - spin_unlock_irqrestore(i2s->lock, flags); + spin_unlock_irqrestore(&priv->lock, flags); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - spin_lock_irqsave(i2s->lock, flags); + spin_lock_irqsave(&priv->lock, flags); if (capture) { i2s_rxctrl(i2s, 0); @@ -981,7 +980,7 @@ static int i2s_trigger(struct snd_pcm_substream *substream, i2s_fifo(i2s, FIC_TXFLUSH); } - spin_unlock_irqrestore(i2s->lock, flags); + spin_unlock_irqrestore(&priv->lock, flags); pm_runtime_put(dai->dev); break; } @@ -1081,13 +1080,13 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) i2s->rfs = 0; i2s->bfs = 0; - spin_lock_irqsave(i2s->lock, flags); + spin_lock_irqsave(&priv->lock, flags); i2s_txctrl(i2s, 0); i2s_rxctrl(i2s, 0); i2s_fifo(i2s, FIC_TXFLUSH); i2s_fifo(other, FIC_TXFLUSH); i2s_fifo(i2s, FIC_RXFLUSH); - spin_unlock_irqrestore(i2s->lock, flags); + spin_unlock_irqrestore(&priv->lock, flags); /* Gate CDCLK by default */ if (!is_opened(other)) @@ -1108,9 +1107,9 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) if (!is_secondary(i2s)) { if (i2s->quirks & QUIRK_NEED_RSTCLR) { - spin_lock_irqsave(i2s->lock, flags); + spin_lock_irqsave(&priv->lock, flags); writel(0, priv->addr + I2SCON); - spin_unlock_irqrestore(i2s->lock, flags); + spin_unlock_irqrestore(&priv->lock, flags); } } @@ -1317,13 +1316,13 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) ARRAY_SIZE(p_names), CLK_SET_RATE_NO_REPARENT | CLK_SET_RATE_PARENT, priv->addr + I2SMOD, reg_info->rclksrc_off, - 1, 0, i2s->lock); + 1, 0, &priv->lock); priv->clk_table[CLK_I2S_RCLK_PSR] = clk_register_divider(dev, i2s_clk_name[CLK_I2S_RCLK_PSR], i2s_clk_name[CLK_I2S_RCLK_SRC], CLK_SET_RATE_PARENT, - priv->addr + I2SPSR, 8, 6, 0, i2s->lock); + priv->addr + I2SPSR, 8, 6, 0, &priv->lock); p_names[0] = i2s_clk_name[CLK_I2S_RCLK_PSR]; priv->clk_data.clk_num = 2; @@ -1333,7 +1332,7 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) i2s_clk_name[CLK_I2S_CDCLK], p_names[0], CLK_SET_RATE_PARENT, priv->addr + I2SMOD, reg_info->cdclkcon_off, - CLK_GATE_SET_TO_DISABLE, i2s->lock); + CLK_GATE_SET_TO_DISABLE, &priv->lock); priv->clk_data.clk_num += 1; priv->clk_data.clks = priv->clk_table; @@ -1411,8 +1410,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai = &priv->dai[SAMSUNG_I2S_ID_PRIMARY - 1]; - spin_lock_init(&priv->spinlock); - pri_dai->lock = &priv->spinlock; + spin_lock_init(&priv->lock); if (!np) { if (i2s_pdata == NULL) { @@ -1474,7 +1472,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (quirks & QUIRK_SEC_DAI) { sec_dai = &priv->dai[SAMSUNG_I2S_ID_SECONDARY - 1]; - sec_dai->lock = &priv->spinlock; sec_dai->variant_regs = pri_dai->variant_regs; sec_dai->dma_playback.addr = regs_base + I2STXDS; sec_dai->dma_playback.chan_name = "tx-sec"; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: i2s: Move IP variant data to common driver data structure" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Move IP variant data to common driver data structure has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5bfaeddc269401677d61f6d7d40eec76f40e6d4c Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:37 +0100 Subject: [PATCH] ASoC: samsung: i2s: Move IP variant data to common driver data structure The IP variant data is another thing common for both DAIs, move it to the driver's common data structure. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 42 ++++++++++++++++++++--------------------- 1 file changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index cf8615e23879..0c4c4de8c7e9 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -85,7 +85,6 @@ struct i2s_dai { struct snd_dmaengine_dai_dma_data idma_playback; dma_filter_fn filter; u32 quirks; - const struct samsung_i2s_variant_regs *variant_regs; struct samsung_i2s_priv *priv; }; @@ -122,6 +121,8 @@ struct samsung_i2s_priv { u32 suspend_i2scon; u32 suspend_i2spsr; + const struct samsung_i2s_variant_regs *variant_regs; + /* The clock provider's data */ struct clk *clk_table[3]; struct clk_onecell_data clk_data; @@ -146,7 +147,7 @@ static inline bool is_slave(struct i2s_dai *i2s) struct samsung_i2s_priv *priv = i2s->priv; u32 mod = readl(priv->addr + I2SMOD); - return (mod & (1 << i2s->variant_regs->mss_off)) ? true : false; + return (mod & (1 << priv->variant_regs->mss_off)) ? true : false; } /* If this interface of the controller is transmitting data */ @@ -261,8 +262,8 @@ static inline unsigned get_rfs(struct i2s_dai *i2s) struct samsung_i2s_priv *priv = i2s->priv; u32 rfs; - rfs = readl(priv->addr + I2SMOD) >> i2s->variant_regs->rfs_off; - rfs &= i2s->variant_regs->rfs_mask; + rfs = readl(priv->addr + I2SMOD) >> priv->variant_regs->rfs_off; + rfs &= priv->variant_regs->rfs_mask; switch (rfs) { case 7: return 192; @@ -281,9 +282,9 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) { struct samsung_i2s_priv *priv = i2s->priv; u32 mod = readl(priv->addr + I2SMOD); - int rfs_shift = i2s->variant_regs->rfs_off; + int rfs_shift = priv->variant_regs->rfs_off; - mod &= ~(i2s->variant_regs->rfs_mask << rfs_shift); + mod &= ~(priv->variant_regs->rfs_mask << rfs_shift); switch (rfs) { case 192: @@ -321,8 +322,8 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) struct samsung_i2s_priv *priv = i2s->priv; u32 bfs; - bfs = readl(priv->addr + I2SMOD) >> i2s->variant_regs->bfs_off; - bfs &= i2s->variant_regs->bfs_mask; + bfs = readl(priv->addr + I2SMOD) >> priv->variant_regs->bfs_off; + bfs &= priv->variant_regs->bfs_mask; switch (bfs) { case 8: return 256; @@ -343,7 +344,7 @@ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) struct samsung_i2s_priv *priv = i2s->priv; u32 mod = readl(priv->addr + I2SMOD); int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM; - int bfs_shift = i2s->variant_regs->bfs_off; + int bfs_shift = priv->variant_regs->bfs_off; /* Non-TDM I2S controllers do not support BCLK > 48 * FS */ if (!tdm && bfs > 48) { @@ -351,7 +352,7 @@ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) return; } - mod &= ~(i2s->variant_regs->bfs_mask << bfs_shift); + mod &= ~(priv->variant_regs->bfs_mask << bfs_shift); switch (bfs) { case 48: @@ -408,7 +409,7 @@ static void i2s_txctrl(struct i2s_dai *i2s, int on) { struct samsung_i2s_priv *priv = i2s->priv; void __iomem *addr = priv->addr; - int txr_off = i2s->variant_regs->txr_off; + int txr_off = priv->variant_regs->txr_off; u32 con = readl(addr + I2SCON); u32 mod = readl(addr + I2SMOD) & ~(3 << txr_off); @@ -459,7 +460,7 @@ static void i2s_rxctrl(struct i2s_dai *i2s, int on) { struct samsung_i2s_priv *priv = i2s->priv; void __iomem *addr = priv->addr; - int txr_off = i2s->variant_regs->txr_off; + int txr_off = priv->variant_regs->txr_off; u32 con = readl(addr + I2SCON); u32 mod = readl(addr + I2SMOD) & ~(3 << txr_off); @@ -516,7 +517,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs, struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = get_other_dai(i2s); - const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs; + const struct samsung_i2s_variant_regs *i2s_regs = priv->variant_regs; unsigned int cdcon_mask = 1 << i2s_regs->cdclkcon_off; unsigned int rsrc_mask = 1 << i2s_regs->rclksrc_off; u32 mod, mask, val = 0; @@ -644,9 +645,9 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) u32 mod, tmp = 0; unsigned long flags; - lrp_shift = i2s->variant_regs->lrp_off; - sdf_shift = i2s->variant_regs->sdf_off; - mod_slave = 1 << i2s->variant_regs->mss_off; + lrp_shift = priv->variant_regs->lrp_off; + sdf_shift = priv->variant_regs->sdf_off; + mod_slave = 1 << priv->variant_regs->mss_off; sdf_mask = MOD_SDF_MASK << sdf_shift; lrp_rlow = MOD_LR_RLOW << lrp_shift; @@ -1023,7 +1024,6 @@ i2s_delay(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); u32 reg = readl(priv->addr + I2SFIC); snd_pcm_sframes_t delay; - const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs; WARN_ON(!pm_runtime_active(dai->dev)); @@ -1032,7 +1032,7 @@ i2s_delay(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) else if (is_secondary(i2s)) delay = FICS_TXCOUNT(readl(priv->addr + I2SFICS)); else - delay = (reg >> i2s_regs->ftx0cnt_off) & 0x7f; + delay = (reg >> priv->variant_regs->ftx0cnt_off) & 0x7f; return delay; } @@ -1281,7 +1281,7 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) const char *p_names[2] = { NULL }; struct device *dev = &priv->pdev->dev; struct i2s_dai *i2s = samsung_i2s_get_pri_dai(dev); - const struct samsung_i2s_variant_regs *reg_info = i2s->variant_regs; + const struct samsung_i2s_variant_regs *reg_info = priv->variant_regs; const char *i2s_clk_name[ARRAY_SIZE(i2s_clk_desc)]; struct clk *rclksrc; int ret, i; @@ -1400,6 +1400,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; + priv->variant_regs = i2s_dai_data->i2s_variant_regs; + quirks = np ? i2s_dai_data->quirks : i2s_pdata->type.quirks; num_dais = (quirks & QUIRK_SEC_DAI) ? 2 : 1; priv->pdev = pdev; @@ -1458,7 +1460,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_playback.addr_width = 4; pri_dai->dma_capture.addr_width = 4; pri_dai->quirks = quirks; - pri_dai->variant_regs = i2s_dai_data->i2s_variant_regs; pri_dai->priv = priv; if (quirks & QUIRK_PRI_6CHAN) @@ -1472,7 +1473,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (quirks & QUIRK_SEC_DAI) { sec_dai = &priv->dai[SAMSUNG_I2S_ID_SECONDARY - 1]; - sec_dai->variant_regs = pri_dai->variant_regs; sec_dai->dma_playback.addr = regs_base + I2STXDS; sec_dai->dma_playback.chan_name = "tx-sec"; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: i2s: Move quirks data to common driver data structure" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Move quirks data to common driver data structure has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5944170f497c8d8c93704c40d18e794351673a11 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:38 +0100 Subject: [PATCH] ASoC: samsung: i2s: Move quirks data to common driver data structure The quirk flags are common for the primary and the secondary DAI so move respective field from struct i2s_dai to common driver data structure. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 33 ++++++++++++--------------------- 1 file changed, 12 insertions(+), 21 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0c4c4de8c7e9..8f0af4b0f25a 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -84,7 +84,6 @@ struct i2s_dai { struct snd_dmaengine_dai_dma_data dma_capture; struct snd_dmaengine_dai_dma_data idma_playback; dma_filter_fn filter; - u32 quirks; struct samsung_i2s_priv *priv; }; @@ -122,19 +121,13 @@ struct samsung_i2s_priv { u32 suspend_i2spsr; const struct samsung_i2s_variant_regs *variant_regs; + u32 quirks; /* The clock provider's data */ struct clk *clk_table[3]; struct clk_onecell_data clk_data; }; -struct i2s_dai *samsung_i2s_get_pri_dai(struct device *dev) -{ - struct samsung_i2s_priv *priv = dev_get_drvdata(dev); - - return &priv->dai[SAMSUNG_I2S_ID_PRIMARY - 1]; -} - /* Returns true if this is the 'overlay' stereo DAI */ static inline bool is_secondary(struct i2s_dai *i2s) { @@ -343,7 +336,7 @@ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { struct samsung_i2s_priv *priv = i2s->priv; u32 mod = readl(priv->addr + I2SMOD); - int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM; + int tdm = priv->quirks & QUIRK_SUPPORTS_TDM; int bfs_shift = priv->variant_regs->bfs_off; /* Non-TDM I2S controllers do not support BCLK > 48 * FS */ @@ -563,7 +556,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs, case SAMSUNG_I2S_RCLKSRC_1: /* clock corrsponding to IISMOD[10] := 1 */ mask = 1 << i2s_regs->rclksrc_off; - if ((i2s->quirks & QUIRK_NO_MUXPSR) + if ((priv->quirks & QUIRK_NO_MUXPSR) || (clk_id == SAMSUNG_I2S_RCLKSRC_0)) clk_id = 0; else @@ -832,6 +825,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, static int i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = get_other_dai(i2s); unsigned long flags; @@ -847,7 +841,7 @@ static int i2s_startup(struct snd_pcm_substream *substream, else i2s->mode |= DAI_MANAGER; - if (!any_active(i2s) && (i2s->quirks & QUIRK_NEED_RSTCLR)) + if (!any_active(i2s) && (priv->quirks & QUIRK_NEED_RSTCLR)) writel(CON_RSTCLR, i2s->priv->addr + I2SCON); spin_unlock_irqrestore(&lock, flags); @@ -929,7 +923,7 @@ static int config_setup(struct i2s_dai *i2s) if (is_slave(i2s)) return 0; - if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { + if (!(priv->quirks & QUIRK_NO_MUXPSR)) { psr = priv->rclk_srcrate / i2s->frmclk / rfs; writel(((psr - 1) << 8) | PSR_PSREN, priv->addr + I2SPSR); dev_dbg(&i2s->pdev->dev, @@ -1068,10 +1062,10 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); - if (i2s->quirks & QUIRK_NEED_RSTCLR) + if (priv->quirks & QUIRK_NEED_RSTCLR) writel(CON_RSTCLR, priv->addr + I2SCON); - if (i2s->quirks & QUIRK_SUPPORTS_IDMA) + if (priv->quirks & QUIRK_SUPPORTS_IDMA) idma_reg_addr_init(priv->addr, i2s->sec_dai->idma_playback.addr); } @@ -1106,7 +1100,7 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) pm_runtime_get_sync(dai->dev); if (!is_secondary(i2s)) { - if (i2s->quirks & QUIRK_NEED_RSTCLR) { + if (priv->quirks & QUIRK_NEED_RSTCLR) { spin_lock_irqsave(&priv->lock, flags); writel(0, priv->addr + I2SCON); spin_unlock_irqrestore(&priv->lock, flags); @@ -1280,7 +1274,6 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) const char *clk_name[2] = { "i2s_opclk0", "i2s_opclk1" }; const char *p_names[2] = { NULL }; struct device *dev = &priv->pdev->dev; - struct i2s_dai *i2s = samsung_i2s_get_pri_dai(dev); const struct samsung_i2s_variant_regs *reg_info = priv->variant_regs; const char *i2s_clk_name[ARRAY_SIZE(i2s_clk_desc)]; struct clk *rclksrc; @@ -1306,7 +1299,7 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) return -ENOMEM; } - if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { + if (!(priv->quirks & QUIRK_NO_MUXPSR)) { /* Activate the prescaler */ u32 val = readl(priv->addr + I2SPSR); writel(val | PSR_PSREN, priv->addr + I2SPSR); @@ -1400,11 +1393,11 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - priv->variant_regs = i2s_dai_data->i2s_variant_regs; - quirks = np ? i2s_dai_data->quirks : i2s_pdata->type.quirks; num_dais = (quirks & QUIRK_SEC_DAI) ? 2 : 1; priv->pdev = pdev; + priv->variant_regs = i2s_dai_data->i2s_variant_regs; + priv->quirks = quirks; ret = i2s_alloc_dais(priv, i2s_dai_data, num_dais); if (ret < 0) @@ -1459,7 +1452,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_capture.chan_name = "rx"; pri_dai->dma_playback.addr_width = 4; pri_dai->dma_capture.addr_width = 4; - pri_dai->quirks = quirks; pri_dai->priv = priv; if (quirks & QUIRK_PRI_6CHAN) @@ -1482,7 +1474,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.addr_width = 4; - sec_dai->quirks = quirks; sec_dai->idma_playback.addr = idma_addr; sec_dai->pri_dai = pri_dai; sec_dai->priv = priv; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: i2s: Get rid of a static spinlock" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Get rid of a static spinlock has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From defc67c6e3638020cc6189d056e0bc187b297068 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:39 +0100 Subject: [PATCH] ASoC: samsung: i2s: Get rid of a static spinlock This patch makes the spinlock serializing access to the primary/secondary PCM a per I2S controller lock, rather than a global one. There is no need to have a global lock across multiple I2S controllers in the SoC. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 8f0af4b0f25a..692a752b194c 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -88,9 +88,6 @@ struct i2s_dai { struct samsung_i2s_priv *priv; }; -/* Lock for cross i/f checks */ -static DEFINE_SPINLOCK(lock); - struct samsung_i2s_priv { struct platform_device *pdev; struct platform_device *pdev_sec; @@ -101,6 +98,9 @@ struct samsung_i2s_priv { /* Spinlock protecting access to the device's registers */ spinlock_t lock; + /* Lock for cross i/f checks */ + spinlock_t pcm_lock; + /* CPU DAIs and their corresponding drivers */ struct i2s_dai *dai; struct snd_soc_dai_driver *dai_drv; @@ -832,7 +832,7 @@ static int i2s_startup(struct snd_pcm_substream *substream, pm_runtime_get_sync(dai->dev); - spin_lock_irqsave(&lock, flags); + spin_lock_irqsave(&priv->pcm_lock, flags); i2s->mode |= DAI_OPENED; @@ -844,7 +844,7 @@ static int i2s_startup(struct snd_pcm_substream *substream, if (!any_active(i2s) && (priv->quirks & QUIRK_NEED_RSTCLR)) writel(CON_RSTCLR, i2s->priv->addr + I2SCON); - spin_unlock_irqrestore(&lock, flags); + spin_unlock_irqrestore(&priv->pcm_lock, flags); return 0; } @@ -852,11 +852,12 @@ static int i2s_startup(struct snd_pcm_substream *substream, static void i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = get_other_dai(i2s); unsigned long flags; - spin_lock_irqsave(&lock, flags); + spin_lock_irqsave(&priv->pcm_lock, flags); i2s->mode &= ~DAI_OPENED; i2s->mode &= ~DAI_MANAGER; @@ -868,7 +869,7 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, i2s->rfs = 0; i2s->bfs = 0; - spin_unlock_irqrestore(&lock, flags); + spin_unlock_irqrestore(&priv->pcm_lock, flags); pm_runtime_put(dai->dev); } @@ -1406,6 +1407,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai = &priv->dai[SAMSUNG_I2S_ID_PRIMARY - 1]; spin_lock_init(&priv->lock); + spin_lock_init(&priv->pcm_lock); if (!np) { if (i2s_pdata == NULL) { -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: odroid: Add support for secondary CPU DAI" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: odroid: Add support for secondary CPU DAI has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bc3cf17b575a7a97b4af7ddcf86133175da7a582 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:40 +0100 Subject: [PATCH] ASoC: samsung: odroid: Add support for secondary CPU DAI This patch adds DPCM links in order to support the secondary I2S interface. For the secondary PCM interface to be actually available one more entry should be added to the sound-dai property in sound/cpu node in DT. The changes in driver are done in a way so we are backwards compatible with existing DTS/DTB, i.e. if the cpu sound-dai property contains only one entry only one PCM will be registered. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/odroid.c | 131 +++++++++++++++++++++++++++---------- 1 file changed, 95 insertions(+), 36 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index e7b371b07230..18bb3bfe0300 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -7,6 +7,7 @@ */ #include <linux/clk.h> +#include <linux/clk-provider.h> #include <linux/of.h> #include <linux/of_device.h> #include <linux/module.h> @@ -17,21 +18,24 @@ struct odroid_priv { struct snd_soc_card card; - struct snd_soc_dai_link dai_link; - struct clk *clk_i2s_bus; struct clk *sclk_i2s; }; -static int odroid_card_startup(struct snd_pcm_substream *substream) +static int odroid_card_fe_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2); + return 0; } -static int odroid_card_hw_params(struct snd_pcm_substream *substream, +static const struct snd_soc_ops odroid_card_fe_ops = { + .startup = odroid_card_fe_startup, +}; + +static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -86,19 +90,55 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_ops odroid_card_ops = { - .startup = odroid_card_startup, - .hw_params = odroid_card_hw_params, +static const struct snd_soc_ops odroid_card_be_ops = { + .hw_params = odroid_card_be_hw_params, +}; + +static struct snd_soc_dai_link odroid_card_dais[] = { + { + /* Primary FE <-> BE link */ + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .ops = &odroid_card_fe_ops, + .name = "Primary", + .stream_name = "Primary", + .platform_name = "3830000.i2s", + .dynamic = 1, + .dpcm_playback = 1, + }, { + /* BE <-> CODECs link */ + .name = "I2S Mixer", + .cpu_name = "snd-soc-dummy", + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .ops = &odroid_card_be_ops, + .no_pcm = 1, + .dpcm_playback = 1, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, { + /* Secondary FE <-> BE link */ + .playback_only = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .ops = &odroid_card_fe_ops, + .name = "Secondary", + .stream_name = "Secondary", + .platform_name = "samsung-i2s-sec", + .dynamic = 1, + .dpcm_playback = 1, + } }; static int odroid_audio_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; - struct device_node *cpu, *codec; + struct device_node *cpu, *cpu_dai, *codec; struct odroid_priv *priv; - struct snd_soc_dai_link *link; struct snd_soc_card *card; - int ret; + struct snd_soc_dai_link *link, *codec_link; + int num_pcms, ret, i; + struct of_phandle_args args = {}; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -130,45 +170,67 @@ static int odroid_audio_probe(struct platform_device *pdev) return ret; } - link = &priv->dai_link; - - link->ops = &odroid_card_ops; - link->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS; - - card->dai_link = &priv->dai_link; - card->num_links = 1; + card->dai_link = odroid_card_dais; + card->num_links = ARRAY_SIZE(odroid_card_dais); cpu = of_get_child_by_name(dev->of_node, "cpu"); codec = of_get_child_by_name(dev->of_node, "codec"); + link = card->dai_link; + codec_link = &card->dai_link[1]; - link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); - if (!link->cpu_of_node) { - dev_err(dev, "Failed parsing cpu/sound-dai property\n"); - return -EINVAL; + /* + * For backwards compatibility create the secondary CPU DAI link only + * if there are 2 CPU DAI entries in the cpu sound-dai property in DT. + */ + num_pcms = of_count_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells"); + if (num_pcms == 1) + card->num_links--; + + for (i = 0; i < num_pcms; i++, link += 2) { + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", i, &args); + if (ret < 0) + return ret; + + if (!args.np) { + dev_err(dev, "sound-dai property parse error: %d\n", ret); + return -EINVAL; + } + + ret = snd_soc_get_dai_name(&args, &link->cpu_dai_name); + of_node_put(args.np); + + if (ret < 0) + return ret; } - ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); + cpu_dai = of_parse_phandle(cpu, "sound-dai", 0); + of_node_put(cpu); + of_node_put(codec); + + ret = snd_soc_of_get_dai_link_codecs(dev, codec, codec_link); if (ret < 0) goto err_put_codec_n; - link->platform_of_node = link->cpu_of_node; - - link->name = "Primary"; - link->stream_name = link->name; - + /* Set capture capability only for boards with the MAX98090 CODEC */ + if (codec_link->num_codecs > 1) { + card->dai_link[0].dpcm_capture = 1; + card->dai_link[1].dpcm_capture = 1; + } - priv->sclk_i2s = of_clk_get_by_name(link->cpu_of_node, "i2s_opclk1"); + priv->sclk_i2s = of_clk_get_by_name(cpu_dai, "i2s_opclk1"); if (IS_ERR(priv->sclk_i2s)) { ret = PTR_ERR(priv->sclk_i2s); - goto err_put_i2s_n; + goto err_put_codec_n; } - priv->clk_i2s_bus = of_clk_get_by_name(link->cpu_of_node, "iis"); + priv->clk_i2s_bus = of_clk_get_by_name(cpu_dai, "iis"); if (IS_ERR(priv->clk_i2s_bus)) { ret = PTR_ERR(priv->clk_i2s_bus); goto err_put_sclk; } + of_node_put(cpu_dai); ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { @@ -182,10 +244,8 @@ static int odroid_audio_probe(struct platform_device *pdev) clk_put(priv->clk_i2s_bus); err_put_sclk: clk_put(priv->sclk_i2s); -err_put_i2s_n: - of_node_put(link->cpu_of_node); err_put_codec_n: - snd_soc_of_put_dai_link_codecs(link); + snd_soc_of_put_dai_link_codecs(codec_link); return ret; } @@ -193,8 +253,7 @@ static int odroid_audio_remove(struct platform_device *pdev) { struct odroid_priv *priv = platform_get_drvdata(pdev); - of_node_put(priv->dai_link.cpu_of_node); - snd_soc_of_put_dai_link_codecs(&priv->dai_link); + snd_soc_of_put_dai_link_codecs(&priv->card.dai_link[1]); clk_put(priv->sclk_i2s); clk_put(priv->clk_i2s_bus); -- 2.20.1

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[alsa-devel] Applied "ASoC: samsung: Specify DMA channel names through custom DMA config" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: Specify DMA channel names through custom DMA config has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0f928c19b646f6af39ccf7481a546e5da616bb78 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:41 +0100 Subject: [PATCH] ASoC: samsung: Specify DMA channel names through custom DMA config This is a part of conversion of Samsung platforms to use the custom DMA config for specifying DMA channel names, in addition to passing custom DMA device for the secondary CPU DAI's "PCM" component for some variants of the I2S controller. We also don't set the SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME any more as setting it wouldn't allow to specify DMA channels through the custom DMA config. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/dmaengine.c | 12 ++++-------- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/s3c2412-i2s.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- 4 files changed, 7 insertions(+), 11 deletions(-) diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index 84601fa9aa46..302871974cb3 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -28,7 +28,6 @@ int samsung_asoc_dma_platform_register(struct device *dev, dma_filter_fn filter, const char *tx, const char *rx, struct device *dma_dev) { - unsigned int flags = SND_DMAENGINE_PCM_FLAG_COMPAT; struct snd_dmaengine_pcm_config *pcm_conf; pcm_conf = devm_kzalloc(dev, sizeof(*pcm_conf), GFP_KERNEL); @@ -39,14 +38,11 @@ int samsung_asoc_dma_platform_register(struct device *dev, dma_filter_fn filter, pcm_conf->compat_filter_fn = filter; pcm_conf->dma_dev = dma_dev; - if (dev->of_node) { - pcm_conf->chan_names[SNDRV_PCM_STREAM_PLAYBACK] = tx; - pcm_conf->chan_names[SNDRV_PCM_STREAM_CAPTURE] = rx; - } else { - flags |= SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME; - } + pcm_conf->chan_names[SNDRV_PCM_STREAM_PLAYBACK] = tx; + pcm_conf->chan_names[SNDRV_PCM_STREAM_CAPTURE] = rx; - return devm_snd_dmaengine_pcm_register(dev, pcm_conf, flags); + return devm_snd_dmaengine_pcm_register(dev, pcm_conf, + SND_DMAENGINE_PCM_FLAG_COMPAT); } EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 692a752b194c..6ab99e38e6dd 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1460,7 +1460,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->drv->playback.channels_max = 6; ret = samsung_asoc_dma_platform_register(&pdev->dev, pri_dai->filter, - NULL, NULL, NULL); + "tx", "rx", NULL); if (ret < 0) goto err_disable_clk; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 67dfa27ae321..c08638b0e458 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -177,7 +177,7 @@ static int s3c2412_iis_dev_probe(struct platform_device *pdev) ret = samsung_asoc_dma_platform_register(&pdev->dev, pdata->dma_filter, - NULL, NULL, NULL); + "tx", "rx", NULL); if (ret) { pr_err("failed to register the DMA: %d\n", ret); return ret; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index ba0f2b94f8d4..a8026b640c95 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -446,7 +446,7 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev) s3c24xx_i2s_pcm_stereo_in.addr = res->start + S3C2410_IISFIFO; ret = samsung_asoc_dma_platform_register(&pdev->dev, NULL, - NULL, NULL, NULL); + "tx", "rx", NULL); if (ret) { dev_err(&pdev->dev, "Failed to register the DMA: %d\n", ret); return ret; -- 2.20.1

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[alsa-devel] [bug report] ASoC: samsung: i2s: Convert to single component with multiple DAIs
by Dan Carpenter 14 Feb '19

14 Feb '19

Hello Sylwester Nawrocki, This is a semi-automatic email about new static checker warnings. The patch a404b72d2bdd: "ASoC: samsung: i2s: Convert to single component with multiple DAIs" from Feb 7, 2019, leads to the following Smatch complaint: sound/soc/samsung/i2s.c:1352 samsung_i2s_probe() warn: variable dereferenced before check 'i2s_pdata' (see line 1336) sound/soc/samsung/i2s.c 1315 static int samsung_i2s_probe(struct platform_device *pdev) 1316 { 1317 struct i2s_dai *pri_dai, *sec_dai = NULL; 1318 struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; 1319 struct resource *res; 1320 u32 regs_base, quirks = 0, idma_addr = 0; 1321 struct device_node *np = pdev->dev.of_node; ^^^^^^^^^^^^^^^^^^^^^^ 1322 const struct samsung_i2s_dai_data *i2s_dai_data; 1323 int num_dais, ret; 1324 struct samsung_i2s_priv *priv; 1325 1326 if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) 1327 i2s_dai_data = of_device_get_match_data(&pdev->dev); 1328 else 1329 i2s_dai_data = (struct samsung_i2s_dai_data *) 1330 platform_get_device_id(pdev)->driver_data; 1331 1332 priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); 1333 if (!priv) 1334 return -ENOMEM; 1335 1336 quirks = np ? i2s_dai_data->quirks : i2s_pdata->type.quirks; ^^^^^^^^^^^^^^^^^^^^^^ If "np" is NULL then we dereference "i2s_pdata". 1337 quirks &= ~(QUIRK_SEC_DAI | QUIRK_SUPPORTS_IDMA); 1338 1339 num_dais = (quirks & QUIRK_SEC_DAI) ? 2 : 1; 1340 priv->pdev = pdev; 1341 1342 ret = i2s_alloc_dais(priv, i2s_dai_data, num_dais); 1343 if (ret < 0) 1344 return ret; 1345 1346 pri_dai = &priv->dai[SAMSUNG_I2S_ID_PRIMARY - 1]; 1347 1348 spin_lock_init(&priv->spinlock); 1349 pri_dai->lock = &priv->spinlock; 1350 1351 if (!np) { 1352 if (i2s_pdata == NULL) { ^^^^^^^^^^^^^^^^^ But here "np" is NULL and we're checking if i2s_pdata is also NULL. 1353 dev_err(&pdev->dev, "Can't work without s3c_audio_pdata\n"); 1354 return -EINVAL; regards, dan carpenter

1 0

[alsa-devel] [PATCH v3 00/13] ASoC: dmaengine updates, secondary CPU DAI for Odroid boards
by Sylwester Nawrocki 14 Feb '19

14 Feb '19

This is a rebased version with the build break fixed and added Krzysztof's Acks. My apologies, I double checked there is no build errors between patches this time. Below is cover letter text from v2. This series restores support for the secondary CPU DAI in samsung/i2s.c driver and adds support for secondary CPU DAI on Odroid boards. I have added 4 more clean up patches comparing to first version of the patch set (16/22...19/22). The first patch adds new platform device for the secondary PCM, I tried to avoid registering new platform device in probe() as it doesn't play well with deferred probing but couldn't find a way to avoid that new device, unique devices are needed to register "PCM" components for each CPU DAI. Patches 2/21...11/21 are mostly a clean up, following conversion to single component with multiple DAIs, allowing to reference both primary/secondary DAI in DT. The patch set has been tested on Odroid U3, XU3, XU4 and Mini2440 (s3c2440 non-DT) boards. Changes since v1 ((addressing Krzysztof's review comments): - added a comment, whitespace change in patch 1/21 - dropped unnecessary new line addition from patch 1/21 Sylwester Nawrocki (13): ASoC: samsung: i2s: Move SFR pointer to common driver data structure ASoC: samsung: i2s: Drop spinlock pointer from i2s_dai data structure ASoC: samsung: i2s: Move IP variant data to common driver data structure ASoC: samsung: i2s: Move quirks data to common driver data structure ASoC: samsung: i2s: Get rid of a static spinlock ASoC: samsung: odroid: Add support for secondary CPU DAI ASoC: samsung: Specify DMA channel names through custom DMA config ASoC: samsung: Drop DAI DMA data chan_name assignments ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition ASoC: samsung: i2s: Comments clean up ASoC: samsung: i2s: Convert to SPDX License Indentifier include/sound/dmaengine_pcm.h | 6 - sound/soc/samsung/dmaengine.c | 12 +- sound/soc/samsung/i2s.c | 297 +++++++++++++------------- sound/soc/samsung/odroid.c | 131 ++++++++---- sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/soc-generic-dmaengine-pcm.c | 21 +- 7 files changed, 250 insertions(+), 225 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH 0/2] ASoC: da7219: BCLK and TDM flexibility improvements
by Adam Thomson 14 Feb '19

14 Feb '19

This patch set updates the driver to be more flexible regarding the BCLK and TDM configurations of the device. Previously the BCLK rate was fixed at 64 periods per WCLK, when the codec is DAI clock master, but to satisfy devices which prefer a lower rate, and to save power, the BCLK rate is now calculated based on hw_params() data. As a fallout of the BCLK efforts it was also apparent that the TDM code could be made more flexible as well. The device is capable of automatically detecting the BCLK rate when it is clock slave, so previous limitations imposed were not necessary. The mask handling is now used as an input to determine the BCLK offset to align closer with other examples within ALSA. Adam Thomson (2): ASoC: da7219: Add support for master mode BCLK rate adjustment ASoC: da7219: Update TDM usage to be more flexible sound/soc/codecs/da7219.c | 116 +++++++++++++++++++++++++++++----------------- sound/soc/codecs/da7219.h | 1 + 2 files changed, 74 insertions(+), 43 deletions(-) -- 1.9.1

1 3

[alsa-devel] [PATCH -next] ALSA: es1688: Remove set but not used variable 'hw'
by YueHaibing 14 Feb '19

14 Feb '19

Fixes gcc '-Wunused-but-set-variable' warning: sound/isa/es1688/es1688_lib.c: In function 'snd_es1688_probe': sound/isa/es1688/es1688_lib.c:124:31: warning: variable 'hw' set but not used [-Wunused-but-set-variable] unsigned short major, minor, hw; ^ Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/isa/es1688/es1688_lib.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index da341969e650..1d9556c045e9 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -121,7 +121,7 @@ EXPORT_SYMBOL(snd_es1688_reset); static int snd_es1688_probe(struct snd_es1688 *chip) { unsigned long flags; - unsigned short major, minor, hw; + unsigned short major, minor; int i; /* @@ -166,14 +166,12 @@ static int snd_es1688_probe(struct snd_es1688 *chip) if (!chip->version) return -ENODEV; /* probably SB */ - hw = ES1688_HW_AUTO; switch (chip->version & 0xfff0) { case 0x4880: snd_printk(KERN_ERR "[0x%lx] ESS: AudioDrive ES488 detected, " "but driver is in another place\n", chip->port); return -ENODEV; case 0x6880: - hw = (chip->version & 0x0f) >= 8 ? ES1688_HW_1688 : ES1688_HW_688; break; default: snd_printk(KERN_ERR "[0x%lx] ESS: unknown AudioDrive chip "

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek: Disable PC beep in passthrough on alc285
by Hui Wang 14 Feb '19

14 Feb '19

It is reported that there's a constant background "hum/whitenoise" in the headset on the Lenovo X1 machines with the codec alc285, and it is confirmed that if we run the command below, the noise will stop. sudo hda-verb /dev/snd/hwC0D0 0x1d SET_PIN_WIDGET_CONTROL 0x0 Then I consulted this issue with Kailang, he told me the pin 0x1d on this codec is used for PC beep in, the noise probably comes from this pin and we can also disable the PC beep in passthrough, then the PC beep in will not affect other sound playback. Fixes: c4cfcf6f4297 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops") Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1660581 Cc: <stable(a)vger.kernel.org> Signed-off-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ba3d4bee7f85..6cd8d392106f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5662,6 +5662,7 @@ enum { ALC294_FIXUP_ASUS_SPK, ALC225_FIXUP_HEADSET_JACK, ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE, + ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6617,6 +6618,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Disable PCBEEP-IN passthrough */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x36 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_LENOVO_HEADPHONE_NOISE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7302,7 +7314,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x19, 0x03a11020}, {0x21, 0x0321101f}), - SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, + SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, {0x12, 0x90a60130}, {0x14, 0x90170110}, {0x19, 0x04a11040}, -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: rt5682: Check JD status when system resume
by shumingf＠realtek.com 14 Feb '19

14 Feb '19

From: Shuming Fan <shumingf(a)realtek.com> The IRQ function may not work when system suspend. We remove snd_soc_dapm_force_enable_pin function call to make sure the bias off when idle and run into suspend/resume function. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 9d5acd2..b7e2e99 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -910,13 +910,20 @@ static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); unsigned int val, count; if (jack_insert) { - snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); - snd_soc_dapm_sync(dapm); + + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2, RT5682_PWR_VREF2); + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + usleep_range(15000, 20000); + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, RT5682_PWR_CBJ); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH); @@ -944,8 +951,10 @@ static int rt5682_headset_detect(struct snd_soc_component *component, rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - snd_soc_dapm_disable_pin(dapm, "CBJ Power"); - snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2, 0); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, 0); rt5682->jack_type = 0; } @@ -1591,8 +1600,6 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -1627,9 +1634,6 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3, - RT5682_PWR_CBJ_BIT, 0, NULL, 0), - /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix, ARRAY_SIZE(rt5682_rec1_l_mix)), @@ -1792,17 +1796,13 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { /*Vref*/ {"MICBIAS1", NULL, "Vref1"}, - {"MICBIAS1", NULL, "Vref2"}, {"MICBIAS2", NULL, "Vref1"}, - {"MICBIAS2", NULL, "Vref2"}, {"CLKDET SYS", NULL, "CLKDET"}, {"IN1P", NULL, "LDO2"}, {"BST1 CBJ", NULL, "IN1P"}, - {"BST1 CBJ", NULL, "CBJ Power"}, - {"CBJ Power", NULL, "Vref2"}, {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, {"RECMIX1L", NULL, "RECMIX1L Power"}, @@ -1912,9 +1912,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Capless"}, {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, - {"HP Amp", NULL, "CBJ Power"}, {"HP Amp", NULL, "Vref1"}, - {"HP Amp", NULL, "Vref2"}, {"HPOL Playback", "Switch", "HP Amp"}, {"HPOR Playback", "Switch", "HP Amp"}, {"HPOL", NULL, "HPOL Playback"}, @@ -2363,6 +2361,8 @@ static int rt5682_resume(struct snd_soc_component *component) regcache_cache_only(rt5682->regmap, false); regcache_sync(rt5682->regmap); + rt5682_irq(0, rt5682); + return 0; } #else -- 2.7.4

1 0

[alsa-devel] right way to enumerate pcm devices
by sylvain.bertrand＠gmail.com 13 Feb '19

13 Feb '19

Hi, I got some code based on hints which, unfortunately, seems not sufficient. I can list the hardware cards, but what is the proper way to enumerate the pcm devices? For instance, I don't get basic user defined pcm devices, I get all the surround pcm devices for the USB mic from my webcam etc... The real question: how I build a clean list of pcm devices I would present to a user to choose from without confusing him/her? Wherever I look I get only this hint based code. I presume I am looking the wrong way. regards, -- Sylvain

3 8

[alsa-devel] [PATCH] Headset microphone and internal speaker support for System76 oryp5
by Jeremy Soller 13 Feb '19

13 Feb '19

On the System76 Oryx Pro (oryp5), there is a headset microphone input attached to 0x19 that does not have a jack detect. In order to get it working, the pin configuration needs to be set correctly, and the ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied. This is similar to the MIC_NO_PRESENCE fixups for some Dell laptops, except we have a separate microphone jack that is already configured correctly. Since the ALC1220 does not have a fixup similar to ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, I have exposed the fixup from the ALC269 in a way that it can be accessed from the alc1220_fixup_system76_oryp5 function. In addition, the alc1220_fixup_clevo_p950 needs to be applied to gain speaker output. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2df7c380765b..496bcd153a24 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1855,6 +1855,8 @@ enum { ALC887_FIXUP_BASS_CHMAP, ALC1220_FIXUP_GB_DUAL_CODECS, ALC1220_FIXUP_CLEVO_P950, + ALC1220_FIXUP_SYSTEM76_ORYP5, + ALC1220_FIXUP_SYSTEM76_ORYP5_PINS, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2056,6 +2058,17 @@ static void alc1220_fixup_clevo_p950(struct hda_codec *codec, snd_hda_override_conn_list(codec, 0x1b, 1, conn1); } +static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action); + +static void alc1220_fixup_system76_oryp5(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + alc1220_fixup_clevo_p950(codec, fix, action); + alc_fixup_headset_mode_no_hp_mic(codec, fix, action); +} + static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = HDA_FIXUP_PINS, @@ -2300,6 +2313,19 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc1220_fixup_clevo_p950, }, + [ALC1220_FIXUP_SYSTEM76_ORYP5] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc1220_fixup_system76_oryp5, + }, + [ALC1220_FIXUP_SYSTEM76_ORYP5_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + {} + }, + .chained = true, + .chain_id = ALC1220_FIXUP_SYSTEM76_ORYP5, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2376,6 +2402,8 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x95e1, "Clevo P95xER", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x95e2, "Clevo P950ER", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0x96e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_SYSTEM76_ORYP5_PINS), + SND_PCI_QUIRK(0x1558, 0x97e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_SYSTEM76_ORYP5_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530),

2 1

[alsa-devel] Applied "ASoC: samsung: i2s: Move clk supplier data to common driver data structure" to the asoc tree
by Mark Brown 13 Feb '19

13 Feb '19

The patch ASoC: samsung: i2s: Move clk supplier data to common driver data structure has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 89d2e831487682b567525275f89d679793dd53da Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Tue, 12 Feb 2019 19:03:23 +0100 Subject: [PATCH] ASoC: samsung: i2s: Move clk supplier data to common driver data structure Having the clocks provider data in struct samsung_i2s_priv, i.e. per the I2S controller instance, rather than per CPU DAI better models the hardware and simplifies the code a little. The clock provider is common for both DAIs. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 68 ++++++++++++++++++++--------------------- 1 file changed, 33 insertions(+), 35 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 273620914471..fffc76ab60da 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -99,9 +99,7 @@ struct i2s_dai { spinlock_t *lock; - /* Below fields are only valid if this is the primary FIFO */ - struct clk *clk_table[3]; - struct clk_onecell_data clk_data; + struct samsung_i2s_priv *priv; }; /* Lock for cross i/f checks */ @@ -118,6 +116,10 @@ struct samsung_i2s_priv { struct i2s_dai *dai; struct snd_soc_dai_driver *dai_drv; int num_dais; + + /* The clock provider's data */ + struct clk *clk_table[3]; + struct clk_onecell_data clk_data; }; struct i2s_dai *samsung_i2s_get_pri_dai(struct device *dev) @@ -625,11 +627,10 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, return ret; } -static int i2s_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) +static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); - struct i2s_dai *other = get_other_dai(i2s); int lrp_shift, sdf_shift, sdf_mask, lrp_rlow, mod_slave; u32 mod, tmp = 0; unsigned long flags; @@ -687,8 +688,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, * CLK_I2S_RCLK_SRC clock is not exposed so we ensure any * clock configuration assigned in DT is not overwritten. */ - if (i2s->rclk_srcrate == 0 && i2s->clk_data.clks == NULL && - other->clk_data.clks == NULL) + if (i2s->rclk_srcrate == 0 && priv->clk_data.clks == NULL) i2s_set_sysclk(dai, SAMSUNG_I2S_RCLKSRC_0, 0, SND_SOC_CLOCK_IN); break; @@ -725,8 +725,8 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, static int i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); - struct i2s_dai *other = get_other_dai(i2s); u32 mod, mask = 0, val = 0; struct clk *rclksrc; unsigned long flags; @@ -811,10 +811,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, i2s->frmclk = params_rate(params); - rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC]; - if (!rclksrc || IS_ERR(rclksrc)) - rclksrc = other->clk_table[CLK_I2S_RCLK_SRC]; - + rclksrc = priv->clk_table[CLK_I2S_RCLK_SRC]; if (rclksrc && !IS_ERR(rclksrc)) i2s->rclk_srcrate = clk_get_rate(rclksrc); @@ -1221,31 +1218,30 @@ static int i2s_runtime_resume(struct device *dev) } #endif /* CONFIG_PM */ -static void i2s_unregister_clocks(struct i2s_dai *i2s) +static void i2s_unregister_clocks(struct samsung_i2s_priv *priv) { int i; - for (i = 0; i < i2s->clk_data.clk_num; i++) { - if (!IS_ERR(i2s->clk_table[i])) - clk_unregister(i2s->clk_table[i]); + for (i = 0; i < priv->clk_data.clk_num; i++) { + if (!IS_ERR(priv->clk_table[i])) + clk_unregister(priv->clk_table[i]); } } -static void i2s_unregister_clock_provider(struct platform_device *pdev) +static void i2s_unregister_clock_provider(struct samsung_i2s_priv *priv) { - struct i2s_dai *i2s = samsung_i2s_get_pri_dai(&pdev->dev); - - of_clk_del_provider(pdev->dev.of_node); - i2s_unregister_clocks(i2s); + of_clk_del_provider(priv->pdev->dev.of_node); + i2s_unregister_clocks(priv); } -static int i2s_register_clock_provider(struct platform_device *pdev) + +static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) { const char * const i2s_clk_desc[] = { "cdclk", "rclk_src", "prescaler" }; const char *clk_name[2] = { "i2s_opclk0", "i2s_opclk1" }; const char *p_names[2] = { NULL }; - struct device *dev = &pdev->dev; + struct device *dev = &priv->pdev->dev; struct i2s_dai *i2s = samsung_i2s_get_pri_dai(dev); const struct samsung_i2s_variant_regs *reg_info = i2s->variant_regs; const char *i2s_clk_name[ARRAY_SIZE(i2s_clk_desc)]; @@ -1277,37 +1273,37 @@ static int i2s_register_clock_provider(struct platform_device *pdev) u32 val = readl(i2s->addr + I2SPSR); writel(val | PSR_PSREN, i2s->addr + I2SPSR); - i2s->clk_table[CLK_I2S_RCLK_SRC] = clk_register_mux(dev, + priv->clk_table[CLK_I2S_RCLK_SRC] = clk_register_mux(dev, i2s_clk_name[CLK_I2S_RCLK_SRC], p_names, ARRAY_SIZE(p_names), CLK_SET_RATE_NO_REPARENT | CLK_SET_RATE_PARENT, i2s->addr + I2SMOD, reg_info->rclksrc_off, 1, 0, i2s->lock); - i2s->clk_table[CLK_I2S_RCLK_PSR] = clk_register_divider(dev, + priv->clk_table[CLK_I2S_RCLK_PSR] = clk_register_divider(dev, i2s_clk_name[CLK_I2S_RCLK_PSR], i2s_clk_name[CLK_I2S_RCLK_SRC], CLK_SET_RATE_PARENT, i2s->addr + I2SPSR, 8, 6, 0, i2s->lock); p_names[0] = i2s_clk_name[CLK_I2S_RCLK_PSR]; - i2s->clk_data.clk_num = 2; + priv->clk_data.clk_num = 2; } - i2s->clk_table[CLK_I2S_CDCLK] = clk_register_gate(dev, + priv->clk_table[CLK_I2S_CDCLK] = clk_register_gate(dev, i2s_clk_name[CLK_I2S_CDCLK], p_names[0], CLK_SET_RATE_PARENT, i2s->addr + I2SMOD, reg_info->cdclkcon_off, CLK_GATE_SET_TO_DISABLE, i2s->lock); - i2s->clk_data.clk_num += 1; - i2s->clk_data.clks = i2s->clk_table; + priv->clk_data.clk_num += 1; + priv->clk_data.clks = priv->clk_table; ret = of_clk_add_provider(dev->of_node, of_clk_src_onecell_get, - &i2s->clk_data); + &priv->clk_data); if (ret < 0) { dev_err(dev, "failed to add clock provider: %d\n", ret); - i2s_unregister_clocks(i2s); + i2s_unregister_clocks(priv); } return ret; @@ -1426,6 +1422,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_capture.addr_width = 4; pri_dai->quirks = quirks; pri_dai->variant_regs = i2s_dai_data->i2s_variant_regs; + pri_dai->priv = priv; if (quirks & QUIRK_PRI_6CHAN) pri_dai->drv->playback.channels_max = 6; @@ -1454,6 +1451,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai->quirks = quirks; sec_dai->idma_playback.addr = idma_addr; sec_dai->pri_dai = pri_dai; + sec_dai->priv = priv; pri_dai->sec_dai = sec_dai; ret = i2s_create_secondary_device(priv); @@ -1485,11 +1483,11 @@ static int samsung_i2s_probe(struct platform_device *pdev) pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - ret = i2s_register_clock_provider(pdev); + ret = i2s_register_clock_provider(priv); if (ret < 0) goto err_disable_pm; - pri_dai->op_clk = clk_get_parent(pri_dai->clk_table[CLK_I2S_RCLK_SRC]); + pri_dai->op_clk = clk_get_parent(priv->clk_table[CLK_I2S_RCLK_SRC]); return 0; @@ -1513,7 +1511,7 @@ static int samsung_i2s_remove(struct platform_device *pdev) pm_runtime_get_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); - i2s_unregister_clock_provider(pdev); + i2s_unregister_clock_provider(priv); clk_disable_unprepare(pri_dai->clk); pm_runtime_put_noidle(&pdev->dev); i2s_delete_secondary_device(priv); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: i2s: Add widgets and routes for DPCM support" to the asoc tree
by Mark Brown 13 Feb '19

13 Feb '19

The patch ASoC: samsung: i2s: Add widgets and routes for DPCM support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 64aba9bca5bd8d0957b0410bdfa192afb1fcb267 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Tue, 12 Feb 2019 19:03:24 +0100 Subject: [PATCH] ASoC: samsung: i2s: Add widgets and routes for DPCM support This patch adds DAPM widgets required to model the internal mixer of the I2S controller merging audio streams from the primary and from the secondary PCM interface. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index fffc76ab60da..29bcfca20572 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1116,8 +1116,31 @@ static const struct snd_soc_dai_ops samsung_i2s_dai_ops = { .delay = i2s_delay, }; +static const struct snd_soc_dapm_widget samsung_i2s_widgets[] = { + /* Backend DAI */ + SND_SOC_DAPM_AIF_OUT("Mixer DAI TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("Mixer DAI RX", NULL, 0, SND_SOC_NOPM, 0, 0), + + /* Playback Mixer */ + SND_SOC_DAPM_MIXER("Playback Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route samsung_i2s_dapm_routes[] = { + { "Playback Mixer", NULL, "Primary" }, + { "Playback Mixer", NULL, "Secondary" }, + + { "Mixer DAI TX", NULL, "Playback Mixer" }, + { "Playback Mixer", NULL, "Mixer DAI RX" }, +}; + static const struct snd_soc_component_driver samsung_i2s_component = { - .name = "samsung-i2s", + .name = "samsung-i2s", + + .dapm_widgets = samsung_i2s_widgets, + .num_dapm_widgets = ARRAY_SIZE(samsung_i2s_widgets), + + .dapm_routes = samsung_i2s_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(samsung_i2s_dapm_routes), }; #define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \ @@ -1129,6 +1152,7 @@ static int i2s_alloc_dais(struct samsung_i2s_priv *priv, int num_dais) { static const char *dai_names[] = { "samsung-i2s", "samsung-i2s-sec" }; + static const char *stream_names[] = { "Primary", "Secondary" }; struct snd_soc_dai_driver *dai_drv; struct i2s_dai *dai; int i; @@ -1158,6 +1182,7 @@ static int i2s_alloc_dais(struct samsung_i2s_priv *priv, dai_drv->playback.channels_max = 2; dai_drv->playback.rates = i2s_dai_data->pcm_rates; dai_drv->playback.formats = SAMSUNG_I2S_FMTS; + dai_drv->playback.stream_name = stream_names[i]; dai_drv->id = i + 1; dai_drv->name = dai_names[i]; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: i2s: Move opclk data to common driver data structure" to the asoc tree
by Mark Brown 13 Feb '19

13 Feb '19

The patch ASoC: samsung: i2s: Move opclk data to common driver data structure has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3b0fa51ffd827b66f4362397acdfb9742c609b13 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Tue, 12 Feb 2019 19:03:26 +0100 Subject: [PATCH] ASoC: samsung: i2s: Move opclk data to common driver data structure The clock for generating I2S signals is also common for both CPU DAIs so move it to the driver's common data structure. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 70 +++++++++++++++++++---------------------- 1 file changed, 33 insertions(+), 37 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 159c19fdb662..d8414d781f83 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -62,8 +62,6 @@ struct i2s_dai { struct platform_device *pdev; /* Memory mapped SFR region */ void __iomem *addr; - /* Rate of RCLK source clock */ - unsigned long rclk_srcrate; /* Frame Clock */ unsigned frmclk; /* @@ -71,8 +69,6 @@ struct i2s_dai { * 0 indicates CPU driver is free to choose any value. */ unsigned rfs, bfs; - /* Clock for generating I2S signals */ - struct clk *op_clk; /* Pointer to the Primary_Fifo if this is Sec_Fifo, NULL otherwise */ struct i2s_dai *pri_dai; /* Pointer to the Secondary_Fifo if it has one, NULL otherwise */ @@ -118,6 +114,12 @@ struct samsung_i2s_priv { /* The I2S controller's core clock */ struct clk *clk; + /* Clock for generating I2S signals */ + struct clk *op_clk; + + /* Rate of RCLK source clock */ + unsigned long rclk_srcrate; + /* The clock provider's data */ struct clk *clk_table[3]; struct clk_onecell_data clk_data; @@ -496,9 +498,10 @@ static inline void i2s_fifo(struct i2s_dai *i2s, u32 flush) writel(readl(fic) & ~flush, fic); } -static int i2s_set_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int rfs, int dir) +static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs, + int dir) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = get_other_dai(i2s); const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs; @@ -554,44 +557,39 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, clk_id = 1; if (!any_active(i2s)) { - if (i2s->op_clk && !IS_ERR(i2s->op_clk)) { + if (priv->op_clk && !IS_ERR(priv->op_clk)) { if ((clk_id && !(mod & rsrc_mask)) || (!clk_id && (mod & rsrc_mask))) { - clk_disable_unprepare(i2s->op_clk); - clk_put(i2s->op_clk); + clk_disable_unprepare(priv->op_clk); + clk_put(priv->op_clk); } else { - i2s->rclk_srcrate = - clk_get_rate(i2s->op_clk); + priv->rclk_srcrate = + clk_get_rate(priv->op_clk); goto done; } } if (clk_id) - i2s->op_clk = clk_get(&i2s->pdev->dev, + priv->op_clk = clk_get(&i2s->pdev->dev, "i2s_opclk1"); else - i2s->op_clk = clk_get(&i2s->pdev->dev, + priv->op_clk = clk_get(&i2s->pdev->dev, "i2s_opclk0"); - if (WARN_ON(IS_ERR(i2s->op_clk))) { - ret = PTR_ERR(i2s->op_clk); - i2s->op_clk = NULL; + if (WARN_ON(IS_ERR(priv->op_clk))) { + ret = PTR_ERR(priv->op_clk); + priv->op_clk = NULL; goto err; } - ret = clk_prepare_enable(i2s->op_clk); + ret = clk_prepare_enable(priv->op_clk); if (ret) { - clk_put(i2s->op_clk); - i2s->op_clk = NULL; + clk_put(priv->op_clk); + priv->op_clk = NULL; goto err; } - i2s->rclk_srcrate = clk_get_rate(i2s->op_clk); + priv->rclk_srcrate = clk_get_rate(priv->op_clk); - /* Over-ride the other's */ - if (other) { - other->op_clk = i2s->op_clk; - other->rclk_srcrate = i2s->rclk_srcrate; - } } else if ((!clk_id && (mod & rsrc_mask)) || (clk_id && !(mod & rsrc_mask))) { dev_err(&i2s->pdev->dev, @@ -600,8 +598,6 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, goto err; } else { /* Call can't be on the active DAI */ - i2s->op_clk = other->op_clk; - i2s->rclk_srcrate = other->rclk_srcrate; goto done; } @@ -689,7 +685,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) * CLK_I2S_RCLK_SRC clock is not exposed so we ensure any * clock configuration assigned in DT is not overwritten. */ - if (i2s->rclk_srcrate == 0 && priv->clk_data.clks == NULL) + if (priv->rclk_srcrate == 0 && priv->clk_data.clks == NULL) i2s_set_sysclk(dai, SAMSUNG_I2S_RCLKSRC_0, 0, SND_SOC_CLOCK_IN); break; @@ -814,7 +810,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, rclksrc = priv->clk_table[CLK_I2S_RCLK_SRC]; if (rclksrc && !IS_ERR(rclksrc)) - i2s->rclk_srcrate = clk_get_rate(rclksrc); + priv->rclk_srcrate = clk_get_rate(rclksrc); return 0; } @@ -872,6 +868,7 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, static int config_setup(struct i2s_dai *i2s) { + struct samsung_i2s_priv *priv = i2s->priv; struct i2s_dai *other = get_other_dai(i2s); unsigned rfs, bfs, blc; u32 psr; @@ -920,11 +917,11 @@ static int config_setup(struct i2s_dai *i2s) return 0; if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { - psr = i2s->rclk_srcrate / i2s->frmclk / rfs; + psr = priv->rclk_srcrate / i2s->frmclk / rfs; writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR); dev_dbg(&i2s->pdev->dev, "RCLK_SRC=%luHz PSR=%u, RCLK=%dfs, BCLK=%dfs\n", - i2s->rclk_srcrate, psr, rfs, bfs); + priv->rclk_srcrate, psr, rfs, bfs); } return 0; @@ -1067,7 +1064,6 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; i2s->bfs = 0; - i2s->rclk_srcrate = 0; spin_lock_irqsave(i2s->lock, flags); i2s_txctrl(i2s, 0); @@ -1213,8 +1209,8 @@ static int i2s_runtime_suspend(struct device *dev) i2s->suspend_i2scon = readl(i2s->addr + I2SCON); i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); - if (i2s->op_clk) - clk_disable_unprepare(i2s->op_clk); + if (priv->op_clk) + clk_disable_unprepare(priv->op_clk); clk_disable_unprepare(priv->clk); return 0; @@ -1230,8 +1226,8 @@ static int i2s_runtime_resume(struct device *dev) if (ret) return ret; - if (i2s->op_clk) { - ret = clk_prepare_enable(i2s->op_clk); + if (priv->op_clk) { + ret = clk_prepare_enable(priv->op_clk); if (ret) { clk_disable_unprepare(priv->clk); return ret; @@ -1514,7 +1510,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (ret < 0) goto err_disable_pm; - pri_dai->op_clk = clk_get_parent(priv->clk_table[CLK_I2S_RCLK_SRC]); + priv->op_clk = clk_get_parent(priv->clk_table[CLK_I2S_RCLK_SRC]); return 0; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: i2s: Move registers cache to common driver data structure" to the asoc tree
by Mark Brown 13 Feb '19

13 Feb '19

The patch ASoC: samsung: i2s: Move registers cache to common driver data structure has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 81bcbf2c72948d36ba431ac0812d7d7c3d8da0ce Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Tue, 12 Feb 2019 19:03:27 +0100 Subject: [PATCH] ASoC: samsung: i2s: Move registers cache to common driver data structure There is no need to keep the PM suspend/resume register cache separate for each DAI as those registers are common, move related i2s_dai data structure to the driver's common data structure. This will allow us to simplify the code a little eventually and to make it easier to follow. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d8414d781f83..72f0cb7abb30 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -86,9 +86,6 @@ struct i2s_dai { struct snd_dmaengine_dai_dma_data idma_playback; dma_filter_fn filter; u32 quirks; - u32 suspend_i2smod; - u32 suspend_i2scon; - u32 suspend_i2spsr; const struct samsung_i2s_variant_regs *variant_regs; spinlock_t *lock; @@ -120,6 +117,11 @@ struct samsung_i2s_priv { /* Rate of RCLK source clock */ unsigned long rclk_srcrate; + /* Cache of selected I2S registers for system suspend */ + u32 suspend_i2smod; + u32 suspend_i2scon; + u32 suspend_i2spsr; + /* The clock provider's data */ struct clk *clk_table[3]; struct clk_onecell_data clk_data; @@ -1205,9 +1207,9 @@ static int i2s_runtime_suspend(struct device *dev) struct samsung_i2s_priv *priv = dev_get_drvdata(dev); struct i2s_dai *i2s = samsung_i2s_get_pri_dai(dev); - i2s->suspend_i2smod = readl(i2s->addr + I2SMOD); - i2s->suspend_i2scon = readl(i2s->addr + I2SCON); - i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR); + priv->suspend_i2smod = readl(i2s->addr + I2SMOD); + priv->suspend_i2scon = readl(i2s->addr + I2SCON); + priv->suspend_i2spsr = readl(i2s->addr + I2SPSR); if (priv->op_clk) clk_disable_unprepare(priv->op_clk); @@ -1234,9 +1236,9 @@ static int i2s_runtime_resume(struct device *dev) } } - writel(i2s->suspend_i2scon, i2s->addr + I2SCON); - writel(i2s->suspend_i2smod, i2s->addr + I2SMOD); - writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR); + writel(priv->suspend_i2scon, i2s->addr + I2SCON); + writel(priv->suspend_i2smod, i2s->addr + I2SMOD); + writel(priv->suspend_i2spsr, i2s->addr + I2SPSR); return 0; } -- 2.20.1

1 0

[alsa-devel] [PATCH -next] ASoC: cs35l36: Remove unused including <linux/version.h>
by YueHaibing 13 Feb '19

13 Feb '19

Remove including <linux/version.h> that don't need it. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs35l36.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index 4f880a678812..e374fffb7e17 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -8,7 +8,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h>

2 1

[alsa-devel] [PATCH -next] ASoC: mediatek: mt8183: make some functions static
by Wei Yongjun 13 Feb '19

13 Feb '19

Fixes the following sparse warnings: sound/soc/mediatek/mt8183/mt8183-dai-i2s.c:966:5: warning: symbol 'mt8183_dai_i2s_get_share' was not declared. Should it be static? sound/soc/mediatek/mt8183/mt8183-dai-i2s.c:986:5: warning: symbol 'mt8183_dai_i2s_set_priv' was not declared. Should it be static? Fixes: a94aec035a12 ("ASoC: mediatek: mt8183: add platform driver") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-i2s.c b/sound/soc/mediatek/mt8183/mt8183-dai-i2s.c index c25024f72e72..777e93d70bea 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-i2s.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-i2s.c @@ -963,7 +963,7 @@ static const struct mtk_afe_i2s_priv mt8183_i2s_priv[DAI_I2S_NUM] = { }, }; -int mt8183_dai_i2s_get_share(struct mtk_base_afe *afe) +static int mt8183_dai_i2s_get_share(struct mtk_base_afe *afe) { struct mt8183_afe_private *afe_priv = afe->platform_priv; const struct device_node *of_node = afe->dev->of_node; @@ -983,7 +983,7 @@ int mt8183_dai_i2s_get_share(struct mtk_base_afe *afe) return 0; } -int mt8183_dai_i2s_set_priv(struct mtk_base_afe *afe) +static int mt8183_dai_i2s_set_priv(struct mtk_base_afe *afe) { struct mt8183_afe_private *afe_priv = afe->platform_priv; struct mtk_afe_i2s_priv *i2s_priv;

2 1

[alsa-devel] [PATCH -next] ASoC: cros_ec_codec: Make symbol 'cros_ec_dai' static
by Wei Yongjun 13 Feb '19

13 Feb '19

Fixes the following sparse warning: sound/soc/codecs/cros_ec_codec.c:209:27: warning: symbol 'cros_ec_dai' was not declared. Should it be static? Fixes: b291f42a3718 ("ASoC: cros_ec_codec: Add codec driver for Cros EC") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/cros_ec_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index b14100b6a939..99a3af8a15ff 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -206,7 +206,7 @@ static const struct snd_soc_dai_ops cros_ec_i2s_dai_ops = { .set_fmt = cros_ec_i2s_set_dai_fmt, }; -struct snd_soc_dai_driver cros_ec_dai[] = { +static struct snd_soc_dai_driver cros_ec_dai[] = { { .name = "cros_ec_codec I2S", .id = 0,

3 2

[alsa-devel] [PATCH] ALSA: pcm: Comment why read blocks when PCM is not running
by Ricardo Biehl Pasquali 13 Feb '19

13 Feb '19

This avoids bringing back the problem introduced by 62ba568f7aef ("ALSA: pcm: Return 0 when size < start_threshold in capture") and fixed in 00a399cad1a0 ("ALSA: pcm: Revert capture stream behavior change in blocking mode"), which prevented the user from starting capture from another thread. Signed-off-by: Ricardo Biehl Pasquali <pasqualirb(a)gmail.com> --- sound/core/pcm_lib.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index bcb06bd3d..345ab1ab2 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2176,6 +2176,10 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) snd_pcm_update_hw_ptr(substream); + /* + * If size < start_threshold, wait indefinitely. Another + * thread may start capture + */ if (!is_playback && runtime->status->state == SNDRV_PCM_STATE_PREPARED && size >= runtime->start_threshold) { -- 2.19.1

2 1

Re: [alsa-devel] [PATCH v3 0/3] ASoC: Intel: Update broxton machine driver
by Pierre-Louis Bossart 12 Feb '19

12 Feb '19

On 1/23/19 4:23 PM, Pierre-Louis Bossart wrote: > > > On 1/23/19 12:38 PM, Yong Zhi wrote: >> Hello, >> >> This patchset is an update to the following v2 posted by Naveen last year: >> >> https://patchwork.kernel.org/patch/10335295/ >> https://patchwork.kernel.org/patch/10335293/ >> https://patchwork.kernel.org/patch/10335297/ >> >> 1/5 and 5/5 of original series were merged. The changes have been tested for >> both bxt_da7219_max98357a and glk_da7219_max98357a. >> >> Yong Zhi (3): >> ASoC: Intel: Headset button support in broxton machine driver >> ASoC: Intel: Add Geminilake Dialog Maxim machine driver >> ASoC: Intel: glk: Add DAI links for Multi-Playback >> >> sound/soc/intel/boards/bxt_da7219_max98357a.c | 101 +++++++++++++++++++++++--- >> 1 file changed, 89 insertions(+), 12 deletions(-) > > All 3 patches > > Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Yong Zhi, looks like this ack never made it outside of my draft folder on Jan 23 and now unfortunately the patches don't apply cleanly any longer. Would you mind sending a v4? Sorry for the delayed response and generating extra work :-( Thanks! -Pierre

2 1

[alsa-devel] [PATCH] ALSA: PCM: check if ops are defined before suspending PCM
by Pierre-Louis Bossart 12 Feb '19

12 Feb '19

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> BE dai links only have internal PCM's and their substream ops may not be set. Suspending these PCM's will result in their ops->trigger() being invoked and cause a kernel oops. So skip suspending PCM's if their ops are NULL. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/core/pcm_native.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 818dff1de545..b6e158ce6650 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1506,6 +1506,14 @@ int snd_pcm_suspend_all(struct snd_pcm *pcm) /* FIXME: the open/close code should lock this as well */ if (substream->runtime == NULL) continue; + + /* + * Skip BE dai link PCM's that are internal and may + * not have their substream ops set. + */ + if (!substream->ops) + continue; + err = snd_pcm_suspend(substream); if (err < 0 && err != -EBUSY) return err; -- 2.17.1

4 6

[alsa-devel] [PATCH 00/22] ASoC: dmaengine updates, secondary CPU DAI for Odroid boards
by Sylwester Nawrocki 12 Feb '19

12 Feb '19

This series restores support for the secondary CPU DAI in samsung/i2s.c driver and adds support for secondary CPU DAI on Odroid boards. First two patches introduce changes in ASoC generic dmaengine driver so as to use custom DMA config as introduced in commit 194c7dea00c6 "ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config" for handling custom DMA channel names on all Samsung platforms. Patches 7/22...16/22 are mostly clean up, following conversion to single component with multiple DAIs in patch 5/22, allowing to reference both primary/secondary DAI in DT. I tried to avoid registering new platform device in probe() as it doesn't play well with deferred probing but couldn't find a way to avoid that new device, unique devices are needed to register "PCM" components for each CPU DAI. I am not entirely sure we should be removing the SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag like this, it might all be a bit more explicit with the flag. The patch set has been tested on Odroid U3, XU3, XU4 and Mini2440 (s3c2440 non-DT) boards. This series depends on patch "ASoC: samsung: Prevent clk_get_rate() calls in atomic context" Sylwester Nawrocki (22): ASoC: dmaengine: Improve of_node test in dmaengine_pcm_request_chan_of() ASoC: dmaengine: Extend use of chan_names provided in custom DMA config ASoC: samsung: dmaengine: Allow to specify custom DMA device ASoC: samsung: i2s: Fix prescaler setting for the secondary DAI ASoC: samsung: i2s: Convert to single component with multiple DAIs ASoC: samsung: i2s: Restore support for the secondary PCM ASoC: samsung: i2s: Move clk supplier data to common driver data structure ASoC: samsung: i2s: Add widgets and routes for DPCM support ASoC: samsung: i2s: Move core clk to the driver common data structure ASoC: samsung: i2s: Move opclk data to common driver data structure ASoC: samsung: i2s: Move registers cache to common driver data structure ASoC: samsung: i2s: Move SFR pointer to common driver data structure ASoC: samsung: i2s: Drop spinlock pointer from i2s_dai data structure ASoC: samsung: i2s: Move IP variant data to common driver data structure ASoC: samsung: i2s: Move quirks data to common driver data structure ASoC: samsung: i2s: Get rid of a static spinlock ASoC: samsung: odroid: Add support for secondary CPU DAI ASoC: samsung: Specify DMA channel names through custom DMA config ASoC: samsung: Drop DAI DMA data chan_name assignments ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag ARM: dts: exynos5422-odroidxu3: Add support for secondary DAI ARM: dts: exynos5422-odroidxu4: Add support for secondary DAI .../boot/dts/exynos5422-odroidxu3-audio.dtsi | 6 +- arch/arm/boot/dts/exynos5422-odroidxu4.dts | 4 +- include/sound/dmaengine_pcm.h | 6 - sound/soc/samsung/dma.h | 3 +- sound/soc/samsung/dmaengine.c | 16 +- sound/soc/samsung/i2s.c | 630 ++++++++++-------- sound/soc/samsung/odroid.c | 132 +++- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/samsung/spdif.c | 2 +- sound/soc/soc-generic-dmaengine-pcm.c | 13 +- 12 files changed, 489 insertions(+), 333 deletions(-) -- 2.20.1

3 52

[alsa-devel] don't pass a NULL struct device to DMA API functions
by Christoph Hellwig 12 Feb '19

12 Feb '19

We still have a few drivers which pass a NULL struct device pointer to DMA API functions, which generally is a bad idea as the API implementations rely on the device not only for ops selection, but also the dma mask and various other attributes. This series contains all easy conversions to pass a struct device, besides that there also is some arch code that needs separate handling, a driver that should not use the DMA API at all, and one that is a complete basket case to be deal with separately.

7 40

[alsa-devel] [PATCH 00/17] ASoC: soc-acpi: remove SOF file prefix and unused fields
by Pierre-Louis Bossart 12 Feb '19

12 Feb '19

Remove the hard-coded "intel" for all SOF files. The SOF code now adds a different default path for firmware and topology, both of which can be overridden by kernel module arguments. The intent is to make releases more manageable since for the same platform there will be different releases (e.g. public and public+non-public components). This change should have no impact since pre-production devices with early versions of SOF will be updated with these patches. Also remove fields that have not been used by SOF in a long time, Marie Kondo style. In hindsight no one seems to recall why they were necessary. Pierre-Louis Bossart (17): ASoC: Intel: bxt-match: remove prefix for SOF files ASoC: Intel: byt-match.c: remove prefix for SOF files ASoC: Intel: cht-match: remove prefix for SOF files ASoC: Intel: cnl-match: remove prefix for SOF files ASoC: Intel: glk-match: remove prefix for SOF files ASoC: Intel: hda-match: remove prefix for SOF files ASoC: Intel: hsw-bdw-match: remove prefix for SOF files ASoC: Intel: icl-match: remove prefix for SOF files ASoC: Intel: soc-acpi: bxt-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: byt-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: cht-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: glk-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: hsw-bdw-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: icl-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: cnl-match.c: remove asoc_plat_name field ASoC: soc-acpi: remove asoc_plat_name field ASoC: soc-acpi: remove new_mach_data field include/sound/soc-acpi.h | 5 -- .../intel/common/soc-acpi-intel-bxt-match.c | 25 +++---- .../intel/common/soc-acpi-intel-byt-match.c | 60 +++++++--------- .../intel/common/soc-acpi-intel-cht-match.c | 70 ++++++++----------- .../intel/common/soc-acpi-intel-cnl-match.c | 5 +- .../intel/common/soc-acpi-intel-glk-match.c | 10 ++- .../intel/common/soc-acpi-intel-hda-match.c | 2 +- .../common/soc-acpi-intel-hsw-bdw-match.c | 20 +++--- .../intel/common/soc-acpi-intel-icl-match.c | 5 +- 9 files changed, 79 insertions(+), 123 deletions(-) -- 2.17.1

2 34

[alsa-devel] [RFC PATCH 0/2] ASoC: core/Intel: fix module-in-use error
by Pierre-Louis Bossart 12 Feb '19

12 Feb '19

Recent simplifications to the SOF driver, which now follows the same device hierarchy as the Skylake driver, expose a conceptual issue with the module refcount that blocks module rmmod/modprobe loop tests for both drivers. Sending as an RFC since I don't fully understand the initial design and might have missed a simpler solution. Pierre-Louis Bossart (2): ASoC: core: don't increase component module refcount unconditionally ASoC: Intel: Skylake: set .ignore_module_refcount field in component include/sound/soc.h | 3 +++ sound/soc/intel/skylake/skl-pcm.c | 1 + sound/soc/soc-core.c | 6 ++++-- 3 files changed, 8 insertions(+), 2 deletions(-) -- 2.17.1

4 9

[alsa-devel] [PATCH 1/2] ASoC: pcm3060: Add soft reset on probe
by Kirill Marinushkin 12 Feb '19

12 Feb '19

Softly reset registers values on module probe Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> --- sound/soc/codecs/pcm3060.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 6714aa8d9026..543cb86fd764 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -287,6 +287,14 @@ int pcm3060_probe(struct device *dev) int rc; struct pcm3060_priv *priv = dev_get_drvdata(dev); + /* soft reset */ + rc = regmap_update_bits(priv->regmap, PCM3060_REG64, + PCM3060_REG_MRST, 0); + if (rc) { + dev_err(dev, "failed to reset component, rc=%d\n", rc); + return rc; + } + if (dev->of_node) pcm3060_parse_dt(dev->of_node, priv); -- 2.13.6

2 3

[alsa-devel] [PATCH] ASoC: mediatek: use %pOFn instead of device_node.name
by KaiChieh Chuang 12 Feb '19

12 Feb '19

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index e408c1b..4b613d5 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1269,7 +1269,7 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) /* irq */ irq_id = platform_get_irq(pdev, 0); if (irq_id <= 0) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return irq_id < 0 ? irq_id : -ENXIO; } diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index ff3111e..4e045dd 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1139,7 +1139,7 @@ static int mt8183_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); if (!irq_id) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return -ENXIO; } ret = devm_request_irq(dev, irq_id, mt8183_afe_irq_handler, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: hdac_hdmi: use devm_kzalloc for all structures
by Pierre-Louis Bossart 12 Feb '19

12 Feb '19

Loading/unloading modules exposes issues with memory allocation, which is a mix of devm_kzalloc and manual kzalloc. Move to devm_k routines everywhere to simplify all this. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/hdac_hdmi.c | 87 ++++++++---------------------------- 1 file changed, 18 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index b19d7a3e7a2c..5eeb0fe836a9 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1176,13 +1176,15 @@ static int hdac_hdmi_add_cvt(struct hdac_device *hdev, hda_nid_t nid) struct hdac_hdmi_cvt *cvt; char name[NAME_SIZE]; - cvt = kzalloc(sizeof(*cvt), GFP_KERNEL); + cvt = devm_kzalloc(&hdev->dev, sizeof(*cvt), GFP_KERNEL); if (!cvt) return -ENOMEM; cvt->nid = nid; sprintf(name, "cvt %d", cvt->nid); - cvt->name = kstrdup(name, GFP_KERNEL); + cvt->name = devm_kstrdup(&hdev->dev, name, GFP_KERNEL); + if (!cvt->name) + return -ENOMEM; list_add_tail(&cvt->head, &hdmi->cvt_list); hdmi->num_cvt++; @@ -1287,8 +1289,8 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, mutex_unlock(&hdmi->pin_mutex); } -static int hdac_hdmi_add_ports(struct hdac_hdmi_priv *hdmi, - struct hdac_hdmi_pin *pin) +static int hdac_hdmi_add_ports(struct hdac_device *hdev, + struct hdac_hdmi_pin *pin) { struct hdac_hdmi_port *ports; int max_ports = HDA_MAX_PORTS; @@ -1300,7 +1302,7 @@ static int hdac_hdmi_add_ports(struct hdac_hdmi_priv *hdmi, * implemented. */ - ports = kcalloc(max_ports, sizeof(*ports), GFP_KERNEL); + ports = devm_kcalloc(&hdev->dev, max_ports, sizeof(*ports), GFP_KERNEL); if (!ports) return -ENOMEM; @@ -1319,14 +1321,14 @@ static int hdac_hdmi_add_pin(struct hdac_device *hdev, hda_nid_t nid) struct hdac_hdmi_pin *pin; int ret; - pin = kzalloc(sizeof(*pin), GFP_KERNEL); + pin = devm_kzalloc(&hdev->dev, sizeof(*pin), GFP_KERNEL); if (!pin) return -ENOMEM; pin->nid = nid; pin->mst_capable = false; pin->hdev = hdev; - ret = hdac_hdmi_add_ports(hdmi, pin); + ret = hdac_hdmi_add_ports(hdev, pin); if (ret < 0) return ret; @@ -1468,8 +1470,6 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_device *hdev, { hda_nid_t nid; int i, num_nodes; - struct hdac_hdmi_cvt *temp_cvt, *cvt_next; - struct hdac_hdmi_pin *temp_pin, *pin_next; struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); int ret; @@ -1497,51 +1497,35 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_device *hdev, case AC_WID_AUD_OUT: ret = hdac_hdmi_add_cvt(hdev, nid); if (ret < 0) - goto free_widgets; + return ret; break; case AC_WID_PIN: ret = hdac_hdmi_add_pin(hdev, nid); if (ret < 0) - goto free_widgets; + return ret; break; } } if (!hdmi->num_pin || !hdmi->num_cvt) { ret = -EIO; - goto free_widgets; + dev_err(&hdev->dev, "Bad pin/cvt setup in %s\n", __func__); + return ret; } ret = hdac_hdmi_create_dais(hdev, dais, hdmi, hdmi->num_cvt); if (ret) { dev_err(&hdev->dev, "Failed to create dais with err: %d\n", - ret); - goto free_widgets; + ret); + return ret; } *num_dais = hdmi->num_cvt; ret = hdac_hdmi_init_dai_map(hdev); if (ret < 0) - goto free_widgets; - - return ret; - -free_widgets: - list_for_each_entry_safe(temp_cvt, cvt_next, &hdmi->cvt_list, head) { - list_del(&temp_cvt->head); - kfree(temp_cvt->name); - kfree(temp_cvt); - } - - list_for_each_entry_safe(temp_pin, pin_next, &hdmi->pin_list, head) { - for (i = 0; i < temp_pin->num_ports; i++) - temp_pin->ports[i].pin = NULL; - kfree(temp_pin->ports); - list_del(&temp_pin->head); - kfree(temp_pin); - } - + dev_err(&hdev->dev, "Failed to init DAI map with err: %d\n", + ret); return ret; } @@ -1782,7 +1766,7 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, * this is a new PCM device, create new pcm and * add to the pcm list */ - pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + pcm = devm_kzalloc(&hdev->dev, sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; pcm->pcm_id = device; @@ -1798,7 +1782,6 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, dev_err(&hdev->dev, "chmap control add failed with err: %d for pcm: %d\n", err, device); - kfree(pcm); return err; } } @@ -2075,42 +2058,8 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev) static int hdac_hdmi_dev_remove(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); - struct hdac_hdmi_pin *pin, *pin_next; - struct hdac_hdmi_cvt *cvt, *cvt_next; - struct hdac_hdmi_pcm *pcm, *pcm_next; - struct hdac_hdmi_port *port, *port_next; - int i; - snd_hdac_display_power(hdev->bus, hdev->addr, false); - list_for_each_entry_safe(pcm, pcm_next, &hdmi->pcm_list, head) { - pcm->cvt = NULL; - if (list_empty(&pcm->port_list)) - continue; - - list_for_each_entry_safe(port, port_next, - &pcm->port_list, head) - list_del(&port->head); - - list_del(&pcm->head); - kfree(pcm); - } - - list_for_each_entry_safe(cvt, cvt_next, &hdmi->cvt_list, head) { - list_del(&cvt->head); - kfree(cvt->name); - kfree(cvt); - } - - list_for_each_entry_safe(pin, pin_next, &hdmi->pin_list, head) { - for (i = 0; i < pin->num_ports; i++) - pin->ports[i].pin = NULL; - kfree(pin->ports); - list_del(&pin->head); - kfree(pin); - } - return 0; } -- 2.17.1

3 2

[alsa-devel] [PATCH v3] ASoC: wm8741: Add digital mute callback
by Sergej Sawazki 12 Feb '19

12 Feb '19

Signed-off-by: Sergej Sawazki <sergej(a)taudac.com> --- Changes in v3: * Removed debug message. sound/soc/codecs/wm8741.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 1fedf74..fdda83b 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -358,6 +358,15 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +int wm8741_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_component *component = codec_dai->component; + + snd_soc_component_update_bits(component, WM8741_VOLUME_CONTROL, + WM8741_SOFT_MASK, !!mute << WM8741_SOFT_SHIFT); + return 0; +} + #define WM8741_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ @@ -371,6 +380,7 @@ static const struct snd_soc_dai_ops wm8741_dai_ops = { .hw_params = wm8741_hw_params, .set_sysclk = wm8741_set_dai_sysclk, .set_fmt = wm8741_set_dai_fmt, + .digital_mute = wm8741_mute, }; static struct snd_soc_dai_driver wm8741_dai = { -- 2.7.4

3 2

[alsa-devel] [PATCH v2] ASoC: wm8741: Set OSR mode in hw_params()
by Sergej Sawazki 12 Feb '19

12 Feb '19

For correct operation of the digital filtering and other processing on the WM8741, the user must ensure the correct value of OSR[1:0] is set at all times.[1] Hence, depending the selected sampling rate, set the OSR (over- sampling rate) mode in hw_params(). References: [1] "WM8741 Data Sheet" Signed-off-by: Sergej Sawazki <sergej(a)taudac.com> --- Changes in v2: * Using update_bits() rather than write(). sound/soc/codecs/wm8741.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index fdda83b..a4b8c45 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -196,7 +196,7 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct wm8741_priv *wm8741 = snd_soc_component_get_drvdata(component); - unsigned int iface; + unsigned int iface, mode; int i; /* The set of sample rates that can be supported depends on the @@ -240,11 +240,21 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* oversampling rate */ + if (params_rate(params) > 96000) + mode = 0x40; + else if (params_rate(params) > 48000) + mode = 0x20; + else + mode = 0x00; + dev_dbg(component->dev, "wm8741_hw_params: bit size param = %d, rate param = %d", params_width(params), params_rate(params)); snd_soc_component_update_bits(component, WM8741_FORMAT_CONTROL, WM8741_IWL_MASK, iface); + snd_soc_component_update_bits(component, WM8741_MODE_CONTROL_1, WM8741_OSR_MASK, + mode); return 0; } -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: mediatek: btcvsd fix rx stream assign
by KaiChieh Chuang 12 Feb '19

12 Feb '19

fix tx/rx stream assign wrong direction Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index 4b613d5..1b8bcdaf 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -250,7 +250,7 @@ static int mtk_btcvsd_snd_rx_init(struct mtk_btcvsd_snd *bt) bt->rx->buf_size = BTCVSD_RX_BUF_SIZE; bt->rx->timeout = 0; bt->rx->rw_cnt = 0; - bt->tx->stream = SNDRV_PCM_STREAM_CAPTURE; + bt->rx->stream = SNDRV_PCM_STREAM_CAPTURE; return 0; } -- 1.9.1

2 1

[alsa-devel] [RFC v2] tlv320aic32x4: delay i2c access by 1 ms after hardware reset
by Peter Seiderer 12 Feb '19

12 Feb '19

As stated in 'TLV320AIC3254 Application Reference Guide' ([1]): 3.2 Device Startup Lockout Times After the TLV320AIC3254 initializes through hardware reset at power-up or software reset, the internal registers initialize to default values. This initialization takes place within 1ms after pulling the RESET signal high. During this initialization phase, no register-read or register-write operation should be performed on ADC or DAC coefficient buffers. Also, no block within the codec should be powered up during the initialization phase. [1] http://www.ti.com/lit/an/slaa408a/slaa408a.pdf Signed-off-by: Peter Seiderer <ps.report(a)gmx.net> --- Changes v1 -> v2: - call mdelay only in case the reset gpio was toggled (suggested by Mark Brown) Note: This came up after the electronic departement took a deeper look at the i2c-bus/vcc/reset pins with a logic analyzer and the documentation because of a vcc/reset timing problem (now fixed on custom board). Not 100% sure if the 1 ms is needed (as restricted to ADC/DAC coefficent and codec block power up). --- sound/soc/codecs/tlv320aic32x4.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e1bfba62fc08..96f1526cb258 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -970,6 +970,7 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); + mdelay(1); } snd_soc_component_write(component, AIC32X4_RESET, 0x01); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: Use __printf markup to silence compiler
by Mathieu Malaterre 12 Feb '19

12 Feb '19

Silence warnings (triggered at W=1) by adding relevant __printf attributes. sound/soc/soc-dapm.c:149:2: warning: function 'pop_dbg' might be a candidate for 'gnu_printf' format attribute [-Wsuggest-attribute=format] Signed-off-by: Mathieu Malaterre <malat(a)debian.org> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..92a2e7dab5fe 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -133,6 +133,7 @@ static void pop_wait(u32 pop_time) schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time)); } +__printf(3, 4) static void pop_dbg(struct device *dev, u32 pop_time, const char *fmt, ...) { va_list args; -- 2.19.2

2 1

[alsa-devel] [PATCH 0/2][RFC] ASoC: soc-core: modern style dai_link control
by Kuninori Morimoto 12 Feb '19

12 Feb '19

Hi Mark These are [RFC] patches which adds new rule for modern style dai_link. Many sound are using CPU component as Platform component, and then it has below settings on legacy style dai_link dai_link->platform_of_node = dai_link->cpu_of_node But, this duplicated component will be just ignored because it is already selected when CPU component binding. We want to allow "no Platform" on modern style dai_link. And, current legacy style dai_link allows no Platform settings, then, soc-core will select snd-soc-dummy platform for it automatically. In modern style, we need to have CPU/Codec/Platform as array/pointer style. Thus, this kind of "implicit" selection will be just un-understandable. These 2 patches adds new rule for modern style, and have no effect to legacy style. But, I can't test for *all* board/situation. Thus, I added [RFC] for these. Kuninori Morimoto (2): ASoC: soc-core: allow no Platform on dai_link ASoC: soc-core: no more implicit snd-soc-dummy platform on modern dai_link include/sound/soc.h | 2 +- sound/soc/generic/simple-card-utils.c | 4 ++ sound/soc/soc-core.c | 80 ++++++++++++++++++++++------------- 3 files changed, 56 insertions(+), 30 deletions(-) -- 2.7.4

3 11

[alsa-devel] [PATCH] pcm: dshare: Fix overflow when slave_hw_ptr rolls over boundary
by Brendan Shanks 12 Feb '19

12 Feb '19

In snd_pcm_dshare_sync_area() when 'slave_hw_ptr' rolls over 'slave_boundary', the wrong variable is checked ('dshare->slave_hw_ptr' vs the local 'slave_hw_ptr'). In some cases, this results in 'slave_hw_ptr' not rolling over correctly. 'slave_size' and 'size' are then much too large, and the for loop blocks for several minutes copying samples. This was likely only triggered on 32-bit systems, since the PCM boundary is computed based on LONG_MAX and is much larger on 64-bit systems. This same change was made to pcm_dmix in commit 6c7f60f7a982fdba828e4530a9d7aa0aa2b704ae ("Fix boundary overlap”) from June 2005. Signed-off-by: Brendan Shanks <brendan.shanks(a)teradek.com> diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 2bb735fe..f135b5df 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -121,7 +121,7 @@ static void snd_pcm_dshare_sync_area(snd_pcm_t *pcm) */ slave_hw_ptr -= slave_hw_ptr % dshare->slave_period_size; slave_hw_ptr += dshare->slave_buffer_size; - if (dshare->slave_hw_ptr > dshare->slave_boundary) + if (slave_hw_ptr >= dshare->slave_boundary) slave_hw_ptr -= dshare->slave_boundary; if (slave_hw_ptr < dshare->slave_appl_ptr) slave_size = slave_hw_ptr + (dshare->slave_boundary - dshare->slave_appl_ptr); -- 2.17.1

2 1

[alsa-devel] [sound:for-linus 23/27] (.init.text+0x134): multiple definition of `plat_irq_setup'; arch/sh/kernel/cpu/sh2/setup-sh7619.o:(.init.text+0x30): first defined here
by kbuild test robot 12 Feb '19

12 Feb '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 00a399cad1a063e7665f06b6497a807db20441fd commit: c97617a81a7616d49bc3700959e08c6c6f447093 [23/27] ALSA: hda/ca0132 - Fix build error without CONFIG_PCI config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 8.2.0-11) 8.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c97617a81a7616d49bc3700959e08c6c6f447093 # save the attached .config to linux build tree GCC_VERSION=8.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sh4-linux-gnu-ld: arch/sh/boards/of-generic.o: in function `plat_irq_setup': >> (.init.text+0x134): multiple definition of `plat_irq_setup'; arch/sh/kernel/cpu/sh2/setup-sh7619.o:(.init.text+0x30): first defined here sh4-linux-gnu-ld: arch/sh/boards/of-generic.o: in function `arch_init_clk_ops': (.init.text+0x118): multiple definition of `arch_init_clk_ops'; arch/sh/kernel/cpu/sh2/clock-sh7619.o:(.init.text+0x0): first defined here --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [sound:for-linus 23/27] (.init.text+0x118): multiple definition of `arch_init_clk_ops'; arch/sh/kernel/cpu/sh2/clock-sh7619.o:(.init.text+0x0): first defined here
by kbuild test robot 12 Feb '19

12 Feb '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 00a399cad1a063e7665f06b6497a807db20441fd commit: c97617a81a7616d49bc3700959e08c6c6f447093 [23/27] ALSA: hda/ca0132 - Fix build error without CONFIG_PCI config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 8.2.0-11) 8.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c97617a81a7616d49bc3700959e08c6c6f447093 # save the attached .config to linux build tree GCC_VERSION=8.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sh4-linux-gnu-ld: arch/sh/boards/of-generic.o: in function `plat_irq_setup': (.init.text+0x134): multiple definition of `plat_irq_setup'; arch/sh/kernel/cpu/sh2/setup-sh7619.o:(.init.text+0x30): first defined here sh4-linux-gnu-ld: arch/sh/boards/of-generic.o: in function `arch_init_clk_ops': >> (.init.text+0x118): multiple definition of `arch_init_clk_ops'; arch/sh/kernel/cpu/sh2/clock-sh7619.o:(.init.text+0x0): first defined here --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [RFC v1] tlv320aic32x4: delay i2c access by 1 ms after hardware reset
by Peter Seiderer 11 Feb '19

11 Feb '19

As stated in 'TLV320AIC3254 Application Reference Guide' ([1]): 3.2 Device Startup Lockout Times After the TLV320AIC3254 initializes through hardware reset at power-up or software reset, the internal registers initialize to default values. This initialization takes place within 1ms after pulling the RESET signal high. During this initialization phase, no register-read or register-write operation should be performed on ADC or DAC coefficient buffers. Also, no block within the codec should be powered up during the initialization phase. [1] http://www.ti.com/lit/an/slaa408a/slaa408a.pdf Signed-off-by: Peter Seiderer <ps.report(a)gmx.net> --- Note: This came up after the electronic departement took a deeper look at the i2c-bus/vcc/reset pins with a logic analyzer and the documentation because of a vcc/reset timing problem (now fixed on custom board). Not 100% sure if the 1 ms is needed (as restricted to ADC/DAC coefficent and codec block power up). --- sound/soc/codecs/tlv320aic32x4.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e1bfba62fc08..b3beb0ca87c3 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -972,6 +972,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) gpio_set_value(aic32x4->rstn_gpio, 1); } + mdelay(1); + snd_soc_component_write(component, AIC32X4_RESET, 0x01); if (aic32x4->setup) -- 2.20.1

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[alsa-devel] pcm_dshare calculates size incorrectly on buffer rollover
by Brendan Shanks 11 Feb '19

11 Feb '19

I'm working on an embedded system based on an Ambarella H1 SoC (32-bit ARM Cortex A9). Audio playback through ALSA (and GStreamer) works fine when playing to the raw hw device. When playing through dshare (my normal configuration), playback stops after exactly 7 hours 16 minutes, for about 3 minutes. During the 3 minutes, the playback thread consumes an entire CPU core. After the 3 minutes, GStreamer reports an xrun, recovers from it, and playback goes back to normal. After some debugging, the problem seems to be in snd_pcm_dshare_sync_area(). When dshare->appl_ptr rolls over to 0, 'size' becomes huge, 3036676576. 'slave_size' is also much bigger than it should be, 1258291680. This is what 'size' is set to when the for loop starts, and I believe the for loop then spends ~3 minutes copying a huge amount of samples. This also explains the 7h16m time, it's linked to the PCM boundary which is 1258291200. At 48 kHz, 1258291200 samples takes 7h16m54s. I'm not sure what the fix should be though. Is this really a bug in dshare, or are bad values being set somewhere else? GStreamer? Or maybe the period/buffer size (480/9600) is causing problems? Any advice is appreciated, my modified snd_pcm_dshare_sync_area() and the output are below. Brendan Shanks $ more /proc/asound/card0/pcm1p/sub0/hw_params access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 4 rate: 48000 (48000/1) period_size: 480 buffer_size: 9600 $ more /proc/asound/card0/pcm1p/sub0/sw_params tstamp_mode: ENABLE period_step: 1 avail_min: 480 start_threshold: 1 stop_threshold: 1258291200 silence_threshold: 0 silence_size: 1258291200 boundary: 1258291200 --- output from running 'gst-launch-1.0 -e audiotestsrc ! alsasink' with prints included as below snd_pcm_dshare_sync_area size 3036676576, dshare->appl_ptr 0 dshare->last_appl_ptr 1258290720 slave_hw_ptr1 1258282080 slave_period_size 480 slave_hw_ptr2 1258282080 slave_buffer_size 9600 slave_hw_ptr3 1258291680 slave_boundary 1258291200 slave_hw_ptr4 1258291680 slave_appl_ptr 0 slave_size1 1258291680 size 1258291680 appl_ptr 9120 size 1258291680 boundary 1258291200 last_appl_ptr 0 slave_appl_ptr 0 dshare->slave_appl_ptr 480 exiting snd_pcm_dshare_sync_area size 3036676576, dshare->appl_ptr 0 dshare->last_appl_ptr 1258290720 slave_hw_ptr1 8965920 slave_period_size 480 slave_hw_ptr2 8965920 slave_buffer_size 9600 slave_hw_ptr3 8975520 slave_boundary 1258291200 slave_hw_ptr4 8975520 slave_appl_ptr 8975040 slave_size1 480 size 480 appl_ptr 9120 size 480 boundary 1258291200 last_appl_ptr 0 slave_appl_ptr 8640 dshare->slave_appl_ptr 8975520 exiting --- snd_pcm_dshare_sync_area() from alsa-lib-1.1.6 with prints added: static void snd_pcm_dshare_sync_area(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; snd_pcm_uframes_t slave_hw_ptr, slave_appl_ptr, slave_size; snd_pcm_uframes_t appl_ptr, size; const snd_pcm_channel_area_t *src_areas, *dst_areas; int print = 0; /* calculate the size to transfer */ size = dshare->appl_ptr - dshare->last_appl_ptr; if (! size) return; if (size > 9600) { printf("snd_pcm_dshare_sync_area size %lu, dshare->appl_ptr %lu dshare->last_appl_ptr %lu\n", size, dshare->appl_ptr, dshare->last_appl_ptr); print = 1; } slave_hw_ptr = dshare->slave_hw_ptr; if (print) printf("slave_hw_ptr1 %lu slave_period_size %lu", slave_hw_ptr, dshare->slave_period_size); /* don't write on the last active period - this area may be cleared * by the driver during write operation... */ slave_hw_ptr -= slave_hw_ptr % dshare->slave_period_size; if (print) printf("slave_hw_ptr2 %lu slave_buffer_size %lu", slave_hw_ptr, dshare->slave_buffer_size); slave_hw_ptr += dshare->slave_buffer_size; if (print) printf("slave_hw_ptr3 %lu slave_boundary %lu", slave_hw_ptr, dshare->slave_boundary); if (dshare->slave_hw_ptr > dshare->slave_boundary) slave_hw_ptr -= dshare->slave_boundary; if (print) printf("slave_hw_ptr4 %lu slave_appl_ptr %lu", slave_hw_ptr, dshare->slave_appl_ptr); if (slave_hw_ptr < dshare->slave_appl_ptr) slave_size = slave_hw_ptr + (dshare->slave_boundary - dshare->slave_appl_ptr); else slave_size = slave_hw_ptr - dshare->slave_appl_ptr; if (print) printf("slave_size1 %lu", slave_size); if (slave_size < size) size = slave_size; if (! size) return; if (print) printf("size %lu", size); /* add sample areas here */ src_areas = snd_pcm_mmap_areas(pcm); dst_areas = snd_pcm_mmap_areas(dshare->spcm); appl_ptr = dshare->last_appl_ptr % pcm->buffer_size; if (print) printf("appl_ptr %lu size %lu boundary %lu", appl_ptr, size, pcm->boundary); dshare->last_appl_ptr += size; dshare->last_appl_ptr %= pcm->boundary; if (print) printf("last_appl_ptr %lu", dshare->last_appl_ptr); slave_appl_ptr = dshare->slave_appl_ptr % dshare->slave_buffer_size; if (print) printf("slave_appl_ptr %lu", slave_appl_ptr); dshare->slave_appl_ptr += size; dshare->slave_appl_ptr %= dshare->slave_boundary; if (print) printf("dshare->slave_appl_ptr %lu", dshare->slave_appl_ptr); for (;;) { snd_pcm_uframes_t transfer = size; if (appl_ptr + transfer > pcm->buffer_size) transfer = pcm->buffer_size - appl_ptr; if (slave_appl_ptr + transfer > dshare->slave_buffer_size) transfer = dshare->slave_buffer_size - slave_appl_ptr; share_areas(dshare, src_areas, dst_areas, appl_ptr, slave_appl_ptr, transfer); size -= transfer; if (! size) break; slave_appl_ptr += transfer; slave_appl_ptr %= dshare->slave_buffer_size; appl_ptr += transfer; appl_ptr %= pcm->buffer_size; } if (print) printf("exiting");

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[alsa-devel] [PATCH 21/22] ARM: dts: exynos5422-odroidxu3: Add support for secondary DAI
by Sylwester Nawrocki 11 Feb '19

11 Feb '19

This patch extends DAPM routing and adds secondary CPU DAI entry to support the secondary audio PCM interface on Odroid XU3. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi b/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi index e84544b220b9..51a843bd65ed 100644 --- a/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi +++ b/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi @@ -24,7 +24,9 @@ "Headphone Jack", "MICBIAS", "IN1", "Headphone Jack", "Speakers", "SPKL", - "Speakers", "SPKR"; + "Speakers", "SPKR", + "I2S Playback", "Mixer DAI TX", + "HiFi Playback", "Mixer DAI TX"; assigned-clocks = <&clock CLK_MOUT_EPLL>, <&clock CLK_MOUT_MAU_EPLL>, @@ -51,7 +53,7 @@ <196608000>; cpu { - sound-dai = <&i2s0 0>; + sound-dai = <&i2s0 0>, <&i2s0 1>; }; codec { sound-dai = <&hdmi>, <&max98090>; -- 2.20.1

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[alsa-devel] [PATCH 1/1] seq: arecordmidi: Add num-events option
by pmalani＠chromium.org 11 Feb '19

11 Feb '19

From: Prashant Malani <pmalani(a)chromium.org> Add a command line option to automatically exit after recording a fixed number of MIDI events. This allows a program using arecordmidi to expect a MIDI file to be written automatically when the specified number of events have been received, instead of having to send a SIGINT or SIGTERM programmatically. It also avoids the need to have the arecordmidi process running in the background, and then constantly stat the output file to check if any bytes have been written to it (this makes for less predictable and longer-running tests). This functionality finds use in Chrome OS functional testing, since having to send SIGTERM/SIGINT programmatically and then wait for the output file adds unpredictability and delay to the tests. The addition of this command-line option should (hopefully) not break any existing usage. Signed-off-by: Prashant Malani <pmalani(a)chromium.org> diff --git a/seq/aplaymidi/arecordmidi.c b/seq/aplaymidi/arecordmidi.c index 19dbb7d..ca25bfa 100644 --- a/seq/aplaymidi/arecordmidi.c +++ b/seq/aplaymidi/arecordmidi.c @@ -690,7 +690,8 @@ static void help(const char *argv0) " -t,--ticks=ticks resolution in ticks per beat or frame\n" " -s,--split-channels create a track for each channel\n" " -m,--metronome=client:port play a metronome signal\n" - " -i,--timesig=nn:dd time signature\n", + " -i,--timesig=nn:dd time signature\n" + " -n,--num-events=events fixed number of events to record, then exit\n", argv0); } @@ -706,7 +707,7 @@ static void sighandler(int sig) int main(int argc, char *argv[]) { - static const char short_options[] = "hVlp:b:f:t:sdm:i:"; + static const char short_options[] = "hVlp:b:f:t:sdm:i:n:"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, @@ -719,6 +720,7 @@ int main(int argc, char *argv[]) {"dump", 0, NULL, 'd'}, {"metronome", 1, NULL, 'm'}, {"timesig", 1, NULL, 'i'}, + {"num-events", 1, NULL, 'n'}, { } }; @@ -727,6 +729,9 @@ int main(int argc, char *argv[]) struct pollfd *pfds; int npfds; int c, err; + /* If |num_events| isn't specified, leave it at 0. */ + int num_events = 0; + int events_received = 0; init_seq(); @@ -775,6 +780,11 @@ int main(int argc, char *argv[]) case 'i': time_signature(optarg); break; + case 'n': + num_events = atoi(optarg); + if (num_events <= 0) + fatal("num_events must be greater than 0"); + break; default: help(argv[0]); return 1; @@ -864,13 +874,20 @@ int main(int argc, char *argv[]) err = snd_seq_event_input(seq, &event); if (err < 0) break; - if (event) + if (event) { record_event(event); + events_received++; + } } while (err > 0); if (stop) break; + if (num_events && (events_received == num_events)) + break; } + if (num_events && events_received < num_events) + fputs("Warning: Received signal before num_events\n", stdout); + finish_tracks(); write_file(); -- 2.20.1.495.gaa96b0ce6b-goog

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Re: [alsa-devel] Driver for MOTU devices
by Takashi Sakamoto 10 Feb '19

10 Feb '19

Hi, On Fri, 8 Feb 2019, Peter Sjöberg wrote: > Hello! > > I'm contacting you as the developer of the kernel driver for MOTU firewire devices. > > First I want to thank you for your time and effort to develop the driver - I have the Audio Express which was unusable in Linux for a long time so this was a > really nice surprise! > > I want to ask you a question about the code, if it's the wrong place to ask please redirect me. > In the file amdtp-motu.c line 139 the code thas converts 24 to 32 bits reads: > *dst = (byte[0] << 24) | (byte[1] << 16) | byte[2]; > ..however I am puzzled by the conversion of the third byte here. In my mind it should be: > *dst = (byte[0] << 24) | (byte[1] << 16) | (byte[2] << 8); > > The conversion in the other direction is written like this (line 167-169): > byte[0] = (*src >> 24) & 0xff; > byte[1] = (*src >> 16) & 0xff; > byte[2] = (*src >> 8) & 0xff; > > I just want to help out so please inform me if I'm wrong and why :) Oops. It's a bug since v4.12 or later, indeed. Unfortunately, I'm on vacation. I'll post the patch to fix it in next week after testing it with MOTU models I own. Thanks for your report ;) By the way, I guess that you see many kernel messages with 'unsolicited response' from firewire-core driver, like. fireire_core 0000:03:00.0: unsolicited response (source ffc0, tlabel 1f) As long as I know, AudioExpress often transfers invalid response code for any asynchronous transaction[1]. This brings failures of read/write transaction and ALSA MOTU driver sometimes fails to handle the device. For your information. [1] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/soun… Regards Takashi Sakamoto

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[alsa-devel] [PATCH] ASoC: codecs: jz4725b: Remove unnecessary const qualifier
by Nathan Chancellor 09 Feb '19

09 Feb '19

Clang warns: sound/soc/codecs/jz4725b.c:177:14: warning: duplicate 'const' declaration specifier [-Wduplicate-decl-specifier] static const SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, ^ include/sound/soc.h:356:2: note: expanded from macro 'SOC_VALUE_ENUM_SINGLE_DECL' SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) ^ include/sound/soc.h:353:2: note: expanded from macro 'SOC_VALUE_ENUM_DOUBLE_DECL' const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ ^ As it points out, SOC_VALUE_ENUM_DOUBLE_DECL has the const attribute in its definition so remove it here. Fixes: e9d97b05a80f ("ASoC: codecs: Add jz4725b-codec driver") Link: https://github.com/ClangBuiltLinux/linux/issues/354 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/soc/codecs/jz4725b.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index e3dba92f30bd..5cc8c7ca24c6 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -174,12 +174,12 @@ static const char * const jz4725b_codec_adc_src_texts[] = { "Mic 1", "Mic 2", "Line In", "Mixer", }; static const unsigned int jz4725b_codec_adc_src_values[] = { 0, 1, 2, 3, }; -static const SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, - JZ4725B_CODEC_REG_CR3, - REG_CR3_INSEL_OFFSET, - REG_CR3_INSEL_MASK, - jz4725b_codec_adc_src_texts, - jz4725b_codec_adc_src_values); +static SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, + JZ4725B_CODEC_REG_CR3, + REG_CR3_INSEL_OFFSET, + REG_CR3_INSEL_MASK, + jz4725b_codec_adc_src_texts, + jz4725b_codec_adc_src_values); static const struct snd_kcontrol_new jz4725b_codec_adc_src_ctrl = SOC_DAPM_ENUM("Route", jz4725b_codec_adc_src_enum); -- 2.20.1

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Re: [alsa-devel] [PATCH] ALSA: pcm: Revert capture stream behavior change in blocking mode
by Ricardo Biehl Pasquali 08 Feb '19

08 Feb '19

Takashi Iwai wrote: > e.g. one thread goes to a write loop in blocking mode while another > thread controls the start/stop of the stream manually. You meant a read loop. Agreed. What about adding a comment like: /* * If size < start_threshold, wait indefinitely. Another * thread may start capture */ So no one will bring this idea again. pasquali

2 1

[alsa-devel] ASoC: regulator notifier registration should be managed
by Guennadi Liakhovetski 08 Feb '19

08 Feb '19

From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Regulator notifiers, that were registered during codec driver probing, must be unregistered during driver release, or device managed versions have to be used. This patch fixes codec drivers, that weren't explicitly unregistering notifiers and simplifies those, that did that manually. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/codecs/max9860.c | 3 ++- sound/soc/codecs/pcm512x.c | 5 +++-- sound/soc/codecs/tlv320aic31xx.c | 16 +++------------- sound/soc/codecs/tlv320aic3x.c | 25 ++++--------------------- sound/soc/codecs/wm8770.c | 18 +++--------------- sound/soc/codecs/wm8962.c | 9 +++------ sound/soc/codecs/wm8995.c | 29 +++++++---------------------- sound/soc/codecs/wm8996.c | 9 +++------ 8 files changed, 28 insertions(+), 86 deletions(-) diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index de3d44e9199b..8be636fe6552 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -615,7 +615,8 @@ static int max9860_probe(struct i2c_client *i2c) max9860->dvddio_nb.notifier_call = max9860_dvddio_event; - ret = regulator_register_notifier(max9860->dvddio, &max9860->dvddio_nb); + ret = devm_regulator_register_notifier(max9860->dvddio, + &max9860->dvddio_nb); if (ret) dev_err(dev, "Failed to register DVDDIO notifier: %d\n", ret); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4cc24a5d5c31..18f414be849f 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1520,8 +1520,9 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) pcm512x->supply_nb[2].notifier_call = pcm512x_regulator_event_2; for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) { - ret = regulator_register_notifier(pcm512x->supplies[i].consumer, - &pcm512x->supply_nb[i]); + ret = devm_regulator_register_notifier( + pcm512x->supplies[i].consumer, + &pcm512x->supply_nb[i]); if (ret != 0) { dev_err(dev, "Failed to register regulator notifier: %d\n", diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 608ad49ad978..7a4a607cca3c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1274,8 +1274,9 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) aic31xx->disable_nb[i].nb.notifier_call = aic31xx_regulator_event; aic31xx->disable_nb[i].aic31xx = aic31xx; - ret = regulator_register_notifier(aic31xx->supplies[i].consumer, - &aic31xx->disable_nb[i].nb); + ret = devm_regulator_register_notifier( + aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); if (ret) { dev_err(component->dev, "Failed to request regulator notifier: %d\n", @@ -1298,19 +1299,8 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) return 0; } -static void aic31xx_codec_remove(struct snd_soc_component *component) -{ - struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); - int i; - - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) - regulator_unregister_notifier(aic31xx->supplies[i].consumer, - &aic31xx->disable_nb[i].nb); -} - static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, - .remove = aic31xx_codec_remove, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6a271e6e6b8f..6dcc3ace9d70 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1605,13 +1605,14 @@ static int aic3x_probe(struct snd_soc_component *component) for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event; aic3x->disable_nb[i].aic3x = aic3x; - ret = regulator_register_notifier(aic3x->supplies[i].consumer, - &aic3x->disable_nb[i].nb); + ret = devm_regulator_register_notifier( + aic3x->supplies[i].consumer, + &aic3x->disable_nb[i].nb); if (ret) { dev_err(component->dev, "Failed to request regulator notifier: %d\n", ret); - goto err_notif; + return ret; } } @@ -1669,29 +1670,11 @@ static int aic3x_probe(struct snd_soc_component *component) aic3x_add_widgets(component); return 0; - -err_notif: - while (i--) - regulator_unregister_notifier(aic3x->supplies[i].consumer, - &aic3x->disable_nb[i].nb); - return ret; -} - -static void aic3x_remove(struct snd_soc_component *component) -{ - struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); - int i; - - list_del(&aic3x->list); - for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) - regulator_unregister_notifier(aic3x->supplies[i].consumer, - &aic3x->disable_nb[i].nb); } static const struct snd_soc_component_driver soc_component_dev_aic3x = { .set_bias_level = aic3x_set_bias_level, .probe = aic3x_probe, - .remove = aic3x_remove, .controls = aic3x_snd_controls, .num_controls = ARRAY_SIZE(aic3x_snd_controls), .dapm_widgets = aic3x_dapm_widgets, diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 806245c70f8b..37467c512597 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -666,8 +666,9 @@ static int wm8770_spi_probe(struct spi_device *spi) /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8770->supplies); i++) { - ret = regulator_register_notifier(wm8770->supplies[i].consumer, - &wm8770->disable_nb[i]); + ret = devm_regulator_register_notifier( + wm8770->supplies[i].consumer, + &wm8770->disable_nb[i]); if (ret) { dev_err(&spi->dev, "Failed to register regulator notifier: %d\n", @@ -687,25 +688,12 @@ static int wm8770_spi_probe(struct spi_device *spi) return ret; } -static int wm8770_spi_remove(struct spi_device *spi) -{ - struct wm8770_priv *wm8770 = spi_get_drvdata(spi); - int i; - - for (i = 0; i < ARRAY_SIZE(wm8770->supplies); ++i) - regulator_unregister_notifier(wm8770->supplies[i].consumer, - &wm8770->disable_nb[i]); - - return 0; -} - static struct spi_driver wm8770_spi_driver = { .driver = { .name = "wm8770", .of_match_table = wm8770_of_match, }, .probe = wm8770_spi_probe, - .remove = wm8770_spi_remove }; module_spi_driver(wm8770_spi_driver); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index efd8910b1ff7..467ed78dd2df 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3424,8 +3424,9 @@ static int wm8962_probe(struct snd_soc_component *component) /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) { - ret = regulator_register_notifier(wm8962->supplies[i].consumer, - &wm8962->disable_nb[i]); + ret = devm_regulator_register_notifier( + wm8962->supplies[i].consumer, + &wm8962->disable_nb[i]); if (ret != 0) { dev_err(component->dev, "Failed to register regulator notifier: %d\n", @@ -3467,15 +3468,11 @@ static int wm8962_probe(struct snd_soc_component *component) static void wm8962_remove(struct snd_soc_component *component) { struct wm8962_priv *wm8962 = snd_soc_component_get_drvdata(component); - int i; cancel_delayed_work_sync(&wm8962->mic_work); wm8962_free_gpio(component); wm8962_free_beep(component); - for (i = 0; i < ARRAY_SIZE(wm8962->supplies); i++) - regulator_unregister_notifier(wm8962->supplies[i].consumer, - &wm8962->disable_nb[i]); } static const struct snd_soc_component_driver soc_component_dev_wm8962 = { diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 68c99fe37097..fe2156a1220a 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1995,20 +1995,6 @@ static int wm8995_set_bias_level(struct snd_soc_component *component, return 0; } -static void wm8995_remove(struct snd_soc_component *component) -{ - struct wm8995_priv *wm8995; - int i; - - wm8995 = snd_soc_component_get_drvdata(component); - - for (i = 0; i < ARRAY_SIZE(wm8995->supplies); ++i) - regulator_unregister_notifier(wm8995->supplies[i].consumer, - &wm8995->disable_nb[i]); - - regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); -} - static int wm8995_probe(struct snd_soc_component *component) { struct wm8995_priv *wm8995; @@ -2021,8 +2007,9 @@ static int wm8995_probe(struct snd_soc_component *component) for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) wm8995->supplies[i].supply = wm8995_supply_names[i]; - ret = regulator_bulk_get(component->dev, ARRAY_SIZE(wm8995->supplies), - wm8995->supplies); + ret = devm_regulator_bulk_get(component->dev, + ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); if (ret) { dev_err(component->dev, "Failed to request supplies: %d\n", ret); return ret; @@ -2039,8 +2026,9 @@ static int wm8995_probe(struct snd_soc_component *component) /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) { - ret = regulator_register_notifier(wm8995->supplies[i].consumer, - &wm8995->disable_nb[i]); + ret = devm_regulator_register_notifier( + wm8995->supplies[i].consumer, + &wm8995->disable_nb[i]); if (ret) { dev_err(component->dev, "Failed to register regulator notifier: %d\n", @@ -2052,7 +2040,7 @@ static int wm8995_probe(struct snd_soc_component *component) wm8995->supplies); if (ret) { dev_err(component->dev, "Failed to enable supplies: %d\n", ret); - goto err_reg_get; + retturn ret; } ret = snd_soc_component_read32(component, WM8995_SOFTWARE_RESET); @@ -2099,8 +2087,6 @@ static int wm8995_probe(struct snd_soc_component *component) err_reg_enable: regulator_bulk_disable(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); -err_reg_get: - regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); return ret; } @@ -2188,7 +2174,6 @@ static struct snd_soc_dai_driver wm8995_dai[] = { static const struct snd_soc_component_driver soc_component_dev_wm8995 = { .probe = wm8995_probe, - .remove = wm8995_remove, .set_bias_level = wm8995_set_bias_level, .controls = wm8995_snd_controls, .num_controls = ARRAY_SIZE(wm8995_snd_controls), diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 91711f8958c5..ab04ea18c312 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2801,8 +2801,9 @@ static int wm8996_i2c_probe(struct i2c_client *i2c, /* This should really be moved into the regulator core */ for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { - ret = regulator_register_notifier(wm8996->supplies[i].consumer, - &wm8996->disable_nb[i]); + ret = devm_regulator_register_notifier( + wm8996->supplies[i].consumer, + &wm8996->disable_nb[i]); if (ret != 0) { dev_err(&i2c->dev, "Failed to register regulator notifier: %d\n", @@ -3071,16 +3072,12 @@ static int wm8996_i2c_probe(struct i2c_client *i2c, static int wm8996_i2c_remove(struct i2c_client *client) { struct wm8996_priv *wm8996 = i2c_get_clientdata(client); - int i; wm8996_free_gpio(wm8996); if (wm8996->pdata.ldo_ena > 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); gpio_free(wm8996->pdata.ldo_ena); } - for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) - regulator_unregister_notifier(wm8996->supplies[i].consumer, - &wm8996->disable_nb[i]); return 0; } -- 2.17.1

2 4

[alsa-devel] Applied "ASoC: msm8916-wcd-analog: add missing license information" to the asoc tree
by Mark Brown 08 Feb '19

08 Feb '19

The patch ASoC: msm8916-wcd-analog: add missing license information has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3f22e31da833e576fcfecce1f4e7ed958477d1ec Mon Sep 17 00:00:00 2001 From: Johan Hovold <johan(a)kernel.org> Date: Fri, 8 Feb 2019 16:16:55 +0100 Subject: [PATCH] ASoC: msm8916-wcd-analog: add missing license information Add the missing license and copyright information which never made it into the analog driver when the original driver was split in two as part of the review process. Link: https://lkml.kernel.org/r/1465582725-30183-3-git-send-email-srinivas.kandag… Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Johan Hovold <johan(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index b7cf7cce95fe..368b6c09474b 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1,3 +1,6 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2016, The Linux Foundation. All rights reserved. + #include <linux/module.h> #include <linux/err.h> #include <linux/kernel.h> -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: msm8916-wcd-digital: convert license header to SPDX" to the asoc tree
by Mark Brown 08 Feb '19

08 Feb '19

The patch ASoC: msm8916-wcd-digital: convert license header to SPDX has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3ebc584ce7d1cc73151682c77ad2bea9318ddf98 Mon Sep 17 00:00:00 2001 From: Johan Hovold <johan(a)kernel.org> Date: Fri, 8 Feb 2019 16:16:56 +0100 Subject: [PATCH] ASoC: msm8916-wcd-digital: convert license header to SPDX Convert the GPLv2-only license header to SPDX. Signed-off-by: Johan Hovold <johan(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/msm8916-wcd-digital.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 423bfebabed4..a63961861e55 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -1,14 +1,5 @@ -/* Copyright (c) 2016, The Linux Foundation. All rights reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 and - * only version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - */ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2016, The Linux Foundation. All rights reserved. #include <linux/module.h> #include <linux/err.h> -- 2.20.1

1 0

[alsa-devel] Applied "gpiolib: acpi: Introduce ACPI_GPIO_QUIRK_ONLY_GPIOIO" to the asoc tree
by Mark Brown 08 Feb '19

08 Feb '19

The patch gpiolib: acpi: Introduce ACPI_GPIO_QUIRK_ONLY_GPIOIO has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4d1f7a6eabd45639d9de22a8a004f3c208d13c1a Mon Sep 17 00:00:00 2001 From: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Date: Wed, 6 Feb 2019 22:49:46 +0200 Subject: [PATCH] gpiolib: acpi: Introduce ACPI_GPIO_QUIRK_ONLY_GPIOIO New quirk enforces search for GPIO based on its type, i.e. iterate over GpioIo resources only. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Acked-by: Mika Westerberg <mika.westerberg(a)linux.intel.com> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Tested-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/gpio/gpiolib-acpi.c | 15 ++++-- include/linux/acpi.h | 7 +++ sound/soc/intel/boards/bytcr_rt5651.c | 74 ++++----------------------- 3 files changed, 27 insertions(+), 69 deletions(-) diff --git a/drivers/gpio/gpiolib-acpi.c b/drivers/gpio/gpiolib-acpi.c index 259cf6ab969b..4d291b75cb9f 100644 --- a/drivers/gpio/gpiolib-acpi.c +++ b/drivers/gpio/gpiolib-acpi.c @@ -530,17 +530,24 @@ static int acpi_populate_gpio_lookup(struct acpi_resource *ares, void *data) if (ares->type != ACPI_RESOURCE_TYPE_GPIO) return 1; - if (lookup->n++ == lookup->index && !lookup->desc) { + if (!lookup->desc) { const struct acpi_resource_gpio *agpio = &ares->data.gpio; - int pin_index = lookup->pin_index; + bool gpioint = agpio->connection_type == ACPI_RESOURCE_GPIO_TYPE_INT; + int pin_index; + if (lookup->info.quirks & ACPI_GPIO_QUIRK_ONLY_GPIOIO && gpioint) + lookup->index++; + + if (lookup->n++ != lookup->index) + return 1; + + pin_index = lookup->pin_index; if (pin_index >= agpio->pin_table_length) return 1; lookup->desc = acpi_get_gpiod(agpio->resource_source.string_ptr, agpio->pin_table[pin_index]); - lookup->info.gpioint = - agpio->connection_type == ACPI_RESOURCE_GPIO_TYPE_INT; + lookup->info.gpioint = gpioint; /* * Polarity and triggering are only specified for GpioInt diff --git a/include/linux/acpi.h b/include/linux/acpi.h index 87715f20b69a..03b4c4f225d0 100644 --- a/include/linux/acpi.h +++ b/include/linux/acpi.h @@ -1014,6 +1014,13 @@ struct acpi_gpio_mapping { /* Ignore IoRestriction field */ #define ACPI_GPIO_QUIRK_NO_IO_RESTRICTION BIT(0) +/* + * When ACPI GPIO mapping table is in use the index parameter inside it + * refers to the GPIO resource in _CRS method. That index has no + * distinction of actual type of the resource. When consumer wants to + * get GpioIo type explicitly, this quirk may be used. + */ +#define ACPI_GPIO_QUIRK_ONLY_GPIOIO BIT(1) unsigned int quirks; }; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index c3b7732929cc..b0a4d297176e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -844,74 +844,18 @@ static const struct x86_cpu_id cherrytrail_cpu_ids[] = { {} }; -static const struct acpi_gpio_params first_gpio = { 0, 0, false }; -static const struct acpi_gpio_params second_gpio = { 1, 0, false }; +static const struct acpi_gpio_params ext_amp_enable_gpios = { 0, 0, false }; -static const struct acpi_gpio_mapping byt_rt5651_amp_en_first[] = { - { "ext-amp-enable-gpios", &first_gpio, 1 }, - { }, -}; - -static const struct acpi_gpio_mapping byt_rt5651_amp_en_second[] = { - { "ext-amp-enable-gpios", &second_gpio, 1 }, +static const struct acpi_gpio_mapping cht_rt5651_gpios[] = { + /* + * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, + * other boards may have I2cSerialBusV2, GpioInt, GpioIo instead. + * We want the GpioIo one for the ext-amp-enable-gpio. + */ + { "ext-amp-enable-gpios", &ext_amp_enable_gpios, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { }, }; -/* - * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other - * boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the - * GpioIo one for the ext-amp-enable-gpio and both count for the index in - * acpi_gpio_params index. So we have 2 different mappings and the code - * below figures out which one to use. - */ -struct byt_rt5651_acpi_resource_data { - int gpio_count; - int gpio_int_idx; -}; - -static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg) -{ - struct byt_rt5651_acpi_resource_data *data = arg; - - if (ares->type != ACPI_RESOURCE_TYPE_GPIO) - return 0; - - if (ares->data.gpio.connection_type == ACPI_RESOURCE_GPIO_TYPE_INT) - data->gpio_int_idx = data->gpio_count; - - data->gpio_count++; - return 0; -} - -static void snd_byt_rt5651_mc_pick_amp_en_gpio_mapping(struct device *codec) -{ - struct byt_rt5651_acpi_resource_data data = { 0, -1 }; - LIST_HEAD(resources); - int ret; - - ret = acpi_dev_get_resources(ACPI_COMPANION(codec), &resources, - snd_byt_rt5651_acpi_resource, &data); - if (ret < 0) { - dev_warn(codec, "Failed to get ACPI resources, not adding external amplifier GPIO mapping\n"); - return; - } - - /* All info we need is gathered during the walk */ - acpi_dev_free_resource_list(&resources); - - switch (data.gpio_int_idx) { - case 0: - byt_rt5651_gpios = byt_rt5651_amp_en_second; - break; - case 1: - byt_rt5651_gpios = byt_rt5651_amp_en_first; - break; - default: - dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n", - data.gpio_int_idx); - } -} - struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ u64 mclock_value; /* usually 25MHz (0x17d7940), ignored */ @@ -1038,7 +982,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* Cherry Trail devices use an external amplifier enable gpio */ if (x86_match_cpu(cherrytrail_cpu_ids) && !byt_rt5651_gpios) - snd_byt_rt5651_mc_pick_amp_en_gpio_mapping(codec_dev); + byt_rt5651_gpios = cht_rt5651_gpios; if (byt_rt5651_gpios) { devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); -- 2.20.1

1 0

[alsa-devel] [PATCH v3] gpiolib: acpi: Introduce ACPI_GPIO_QUIRK_ONLY_GPIOIO
by Andy Shevchenko 08 Feb '19

08 Feb '19

New quirk enforces search for GPIO based on its type, i.e. iterate over GpioIo resources only. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Acked-by: Mika Westerberg <mika.westerberg(a)linux.intel.com> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Tested-by: Hans de Goede <hdegoede(a)redhat.com> --- drivers/gpio/gpiolib-acpi.c | 15 ++++-- include/linux/acpi.h | 7 +++ sound/soc/intel/boards/bytcr_rt5651.c | 74 ++++----------------------- 3 files changed, 27 insertions(+), 69 deletions(-) diff --git a/drivers/gpio/gpiolib-acpi.c b/drivers/gpio/gpiolib-acpi.c index 15077c0797b9..2513593f9800 100644 --- a/drivers/gpio/gpiolib-acpi.c +++ b/drivers/gpio/gpiolib-acpi.c @@ -533,17 +533,24 @@ static int acpi_populate_gpio_lookup(struct acpi_resource *ares, void *data) if (ares->type != ACPI_RESOURCE_TYPE_GPIO) return 1; - if (lookup->n++ == lookup->index && !lookup->desc) { + if (!lookup->desc) { const struct acpi_resource_gpio *agpio = &ares->data.gpio; - int pin_index = lookup->pin_index; + bool gpioint = agpio->connection_type == ACPI_RESOURCE_GPIO_TYPE_INT; + int pin_index; + if (lookup->info.quirks & ACPI_GPIO_QUIRK_ONLY_GPIOIO && gpioint) + lookup->index++; + + if (lookup->n++ != lookup->index) + return 1; + + pin_index = lookup->pin_index; if (pin_index >= agpio->pin_table_length) return 1; lookup->desc = acpi_get_gpiod(agpio->resource_source.string_ptr, agpio->pin_table[pin_index]); - lookup->info.gpioint = - agpio->connection_type == ACPI_RESOURCE_GPIO_TYPE_INT; + lookup->info.gpioint = gpioint; /* * Polarity and triggering are only specified for GpioInt diff --git a/include/linux/acpi.h b/include/linux/acpi.h index 13f5cb2c4763..7a3dffe8f971 100644 --- a/include/linux/acpi.h +++ b/include/linux/acpi.h @@ -1014,6 +1014,13 @@ struct acpi_gpio_mapping { /* Ignore IoRestriction field */ #define ACPI_GPIO_QUIRK_NO_IO_RESTRICTION BIT(0) +/* + * When ACPI GPIO mapping table is in use the index parameter inside it + * refers to the GPIO resource in _CRS method. That index has no + * distinction of actual type of the resource. When consumer wants to + * get GpioIo type explicitly, this quirk may be used. + */ +#define ACPI_GPIO_QUIRK_ONLY_GPIOIO BIT(1) unsigned int quirks; }; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index c3b7732929cc..b0a4d297176e 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -844,74 +844,18 @@ static const struct x86_cpu_id cherrytrail_cpu_ids[] = { {} }; -static const struct acpi_gpio_params first_gpio = { 0, 0, false }; -static const struct acpi_gpio_params second_gpio = { 1, 0, false }; +static const struct acpi_gpio_params ext_amp_enable_gpios = { 0, 0, false }; -static const struct acpi_gpio_mapping byt_rt5651_amp_en_first[] = { - { "ext-amp-enable-gpios", &first_gpio, 1 }, - { }, -}; - -static const struct acpi_gpio_mapping byt_rt5651_amp_en_second[] = { - { "ext-amp-enable-gpios", &second_gpio, 1 }, +static const struct acpi_gpio_mapping cht_rt5651_gpios[] = { + /* + * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, + * other boards may have I2cSerialBusV2, GpioInt, GpioIo instead. + * We want the GpioIo one for the ext-amp-enable-gpio. + */ + { "ext-amp-enable-gpios", &ext_amp_enable_gpios, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { }, }; -/* - * Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other - * boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the - * GpioIo one for the ext-amp-enable-gpio and both count for the index in - * acpi_gpio_params index. So we have 2 different mappings and the code - * below figures out which one to use. - */ -struct byt_rt5651_acpi_resource_data { - int gpio_count; - int gpio_int_idx; -}; - -static int snd_byt_rt5651_acpi_resource(struct acpi_resource *ares, void *arg) -{ - struct byt_rt5651_acpi_resource_data *data = arg; - - if (ares->type != ACPI_RESOURCE_TYPE_GPIO) - return 0; - - if (ares->data.gpio.connection_type == ACPI_RESOURCE_GPIO_TYPE_INT) - data->gpio_int_idx = data->gpio_count; - - data->gpio_count++; - return 0; -} - -static void snd_byt_rt5651_mc_pick_amp_en_gpio_mapping(struct device *codec) -{ - struct byt_rt5651_acpi_resource_data data = { 0, -1 }; - LIST_HEAD(resources); - int ret; - - ret = acpi_dev_get_resources(ACPI_COMPANION(codec), &resources, - snd_byt_rt5651_acpi_resource, &data); - if (ret < 0) { - dev_warn(codec, "Failed to get ACPI resources, not adding external amplifier GPIO mapping\n"); - return; - } - - /* All info we need is gathered during the walk */ - acpi_dev_free_resource_list(&resources); - - switch (data.gpio_int_idx) { - case 0: - byt_rt5651_gpios = byt_rt5651_amp_en_second; - break; - case 1: - byt_rt5651_gpios = byt_rt5651_amp_en_first; - break; - default: - dev_warn(codec, "Unknown GpioInt index %d, not adding external amplifier GPIO mapping\n", - data.gpio_int_idx); - } -} - struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ u64 aif_value; /* 1: AIF1, 2: AIF2 */ u64 mclock_value; /* usually 25MHz (0x17d7940), ignored */ @@ -1038,7 +982,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* Cherry Trail devices use an external amplifier enable gpio */ if (x86_match_cpu(cherrytrail_cpu_ids) && !byt_rt5651_gpios) - snd_byt_rt5651_mc_pick_amp_en_gpio_mapping(codec_dev); + byt_rt5651_gpios = cht_rt5651_gpios; if (byt_rt5651_gpios) { devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios); -- 2.20.1

3 3

[alsa-devel] [PATCH 0/3] ASoC: msm8916-wcd: license and documentation fixes
by Johan Hovold 08 Feb '19

08 Feb '19

This series adds the missing license information to the msm8916-wcd-analog driver and converts it's digital counterpart to SPDX. Included is also a fix of the binding example for the analog driver. Johan Johan Hovold (3): ASoC: msm8916-wcd-analog: add missing license information ASoC: msm8916-wcd-digital: convert license header to SPDX dt-bindings: sound: msm8916-wcd-analog: fix example regulator names .../bindings/sound/qcom,msm8916-wcd-analog.txt | 7 ++++--- sound/soc/codecs/msm8916-wcd-analog.c | 5 +++++ sound/soc/codecs/msm8916-wcd-digital.c | 13 +++---------- 3 files changed, 12 insertions(+), 13 deletions(-) -- 2.20.1

2 7

[alsa-devel] [PATCH v6 0/3] ALSA: pcm: anonymous dup implementation
by Jaroslav Kysela 08 Feb '19

08 Feb '19

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup The filtering might be refined. It depends on the real requirements. Perhaps, we may create more ioctl groups. Any comments are more than welcome. v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) v3 of the patches: - group integer declarations in snd_pcm_anonymous_dup() - replaced substream->pcm with pcm in snd_pcm_anonymous_dup() - added SNDRV_PCM_PERM_RW check for read/write/readv/writev syscalls v4 of the patches: - more simple restriction control (only two modes - full/buffer) - the tinyalsa implementation follows this change v5 of the patches: - merge pcm_..._mmap_allowed fcns to the snd_pcm_mmap_... fcns v6: - add proper open_mutex protection for snd_pcm_open_file() Cc: Phil Burk <philburk(a)google.com> Cc: Zach Riggle <riggle(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (3): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: merge pcm_..._mmap_allowed fcns to the snd_pcm_mmap_... fcns ALSA: pcm: implement the mmap buffer mode for the anonymous dup include/sound/pcm.h | 10 +-- include/uapi/sound/asound.h | 6 +- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 ++-- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 154 +++++++++++++++++++++++++++++++++----------- 6 files changed, 133 insertions(+), 53 deletions(-) -- 2.13.6

3 5

[alsa-devel] [PATCH v2 0/2] ASoC: msm8916-wcd: license and documentation fixes
by Johan Hovold 08 Feb '19

08 Feb '19

This series adds the missing license information to the msm8916-wcd-analog driver and converts it's digital counterpart to SPDX. Johan v2 - drop the now applied binding doc update - switch to c++ comments also for the copyright headers for consistency with the ASoC style Johan Hovold (2): ASoC: msm8916-wcd-analog: add missing license information ASoC: msm8916-wcd-digital: convert license header to SPDX sound/soc/codecs/msm8916-wcd-analog.c | 3 +++ sound/soc/codecs/msm8916-wcd-digital.c | 13 ++----------- 2 files changed, 5 insertions(+), 11 deletions(-) -- 2.20.1

1 2

[alsa-devel] [GIT PULL] ASoC updates for v5.1
by Mark Brown 08 Feb '19

08 Feb '19

The following changes since commit f17b5f06cb92ef2250513a1e154c47b78df07d40: Linux 5.0-rc4 (2019-01-27 15:18:05 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.1 for you to fetch changes up to a50e32694fbcdbf55875095258b9398e2eabd71f: ASoC: codecs: jz4725b: Use C++ style comments in header (2019-02-07 14:38:49 +0000) ---------------------------------------------------------------- ASoC: Updates for v5.1 Lots and lots of new drivers so far, a highlight being the MediaTek BTCVSD which is a driver for a Bluetooth radio chip - the first such driver we've had upstream. Hopefully we will soon also see a baseband with an upstream driver! - Support for only powering up channels that are actively being used. - Quite a few improvements to simplify the generic card drivers, especially the merge of the SCU cards into the main generic drivers. - Lots of fixes for probing on Intel systems, trying to rationalize things to look more standard from a framework point of view. - New drivers for Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341, Google ChromeOS embedded controllers, Ingenic JZ4725B, MediaTek BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328, Spreadtrum DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM formatters. ---------------------------------------------------------------- Adam Thomson (2): ASoC: da7219: MCLK should be enabled before DAI clocks ASoC: da7219: Add recalc_rate function to return DAI clock rate Agrawal, Akshu (1): ASoC: ADAU7002: Add optional delay before start of capture Alexander Shiyan (2): ASoC: cs4341: Add driver for CS4341 DAC ASoC: cs4341: Add DT bindings documentation for CS4341 DAC Alison Wang (1): ASoC: sgtl5000: Allow SCLK pad drive strength to be changed Amadeusz Sławiński (3): ASoC: topology: Reduce number of dereferences when accessing dobj ASoC: topology: Remove widgets from dobj list ASoC: topology: Fix memory leak from soc_tplg_denum_create_texts Anders Roxell (1): ASoC: cs4341: fix waring unused-function Axel Lin (1): ASoC: msm8916-wcd-digital: Select REGMAP_MMIO to fix build error Baolin Wang (2): dt-bindings: ASoC: Add Spreadtrum DMA platform documentation ASoC: sprd: Add Spreadtrum audio DMA platfrom driver Bard liao (7): ASoC: rt5645: remove unused mux define ASoC: rt5670: remove unused mux/mixer define ASoC: rt5645: store eq kcontrol byte in __be ASoC: rl6437a: use __be32 for a __be32 buf ASoC: rt5682: add default pdata for i2s mode ASoC: topology: fix memory leak in soc_tplg_dapm_widget_create ASoC: topology: unload physical dai link in remove Charles Keepax (6): ASoC: core: Allow soc_find_component lookups to match parent of_node ASoC: arizona: Add channel numbers to AIFs ASoC: dapm: Only power up active channels from a DAI ASoC: dapm: Check for NULL widget in dapm_update_dai_unlocked ASoC: compress: Clarify the intent of current compressed ops handling ASoC: compress: Add helper functions for component trigger/set_params Cheng-Yi Chiang (4): ASoC: qcom: Kconfig: select max98927 for sdm845 ASoC: sdm845: Set DAI format for dmic codec ASoC: Documentation: Add google, cros-ec-codec ASoC: cros_ec_codec: Add codec driver for Cros EC Clément Péron (1): ASoC: ak4118: fix missing header Colin Ian King (3): ASoC: Intel: make const arrays static, reduces object code size ASoC: mediatek: btcvsd: fix spelling mistake "offest" -> "offset" ASoC: codecs: jz4725b: fix spelling mistake "Deemphatize" -> "Deemphasize" Cosmin Samoila (2): ASoC: micfil: Add bindings for MICFIL DAI ASoC: Add MICFIL SoC Digital Audio Interface driver. Curtis Malainey (1): ASoC: soc-core: clear platform pointers on error Dan Carpenter (1): ASoC: wcd9335: remove some unnecessary NULL checks Daniel Baluta (2): ASoC: ak4458: Add support for AK4497 ASoC: dt-bindings: Document support for ak4497 Daniel Mack (1): ASoC: eliminate left-over from Raumfeld machine driver removal Danny Milosavljevic (8): ASoC: sun4i-codec: Add MIC2 Pre-Amplifier, Mic2 input ASoC: sun4i-codec: Add Mic Playback Volume ASoC: sun4i-codec: Add sun7i_codec_controls, sun7i_codec_codec ASoC: sun4i-codec: Add Mic1 Boost Volume, Mic2 Boost Volume ASoC: sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls ASoC: sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch ASoC: sun4i-codec: Add FM Playback Volume, FM Left, FM Right, FM Playback Switch ASoC: sun4i-codec: Add Line Playback Volume, Line Boost Volume, Line Right, Line Left, Line Playback Switch Dimitris Papavasiliou (2): ASoC: pcm512x: Implement the set_bclk_ratio interface ASoC: pcm512x: Fix clocking calculations when not using the PLL Fabio Estevam (1): ASoC: MAINTAINERS: fsl: Change Fabio's email address Guennadi Liakhovetski (1): ASoC: topology: fix oops/use-after-free case with dai driver Gustavo A. R. Silva (1): ASoC: xlnx: fix error handling in xlnx_formatter_pcm_probe Hans de Goede (17): ASoC: es8316: Add jack-detect support ASoC: es8316: Add DAC mono mix switch mixer control ASoC: Intel: bytcht_es8316: Sort includes alphabetically ASoC: Intel: bytcht_es8316: Minor refactoring ASoC: Intel: bytcht_es8316: Add support for SSP0 (BYTCR) ASoC: Intel: bytcht_es8316: Add jack-detect support ASoC: Intel: bytcht_es8316: Add external speaker mux support ASoC: Intel: bytcht_es8316: Add input-map support ASoC: Intel: bytcht_es8316: Set card long_name based on quirks ASoC: Intel: Add ACPI match table entry for ES8316 codec on BYTCR platform ASoC: Intel: common: Add quirk for PoV P1006W tablet ASoC: rt5651: Add ACPI ID 10EC5640 ASoC: rt5651: Add support for jack detect using an external GPIO ASoC: Intel: bytcr_rt5651: Revert "Fix DMIC map headsetmic mapping" ASoC: Intel: bytcr_rt5651: Add quirks module parameter ASoC: Intel: bytcr_rt5651: Add support for jack-detect using an external GPIO ASoC: Intel: bytcr_rt5651: Add quirk for PoV TAB-P1006W-232 (v1.0) tablet KaiChieh Chuang (2): ASoC: mediatek: add documents for btcvsd driver ASoC: mediatek: add btcvsd driver Katsuhiro Suzuki (2): ASoC: rockchip: support ACODEC for rk3328 ASoC: rockchip: add workaround for silence of rk3288 ACODEC Kuninori Morimoto (31): ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() ASoC: audio-graph-card: add asoc_graph_card_get_conversion() ASoC: audio-graph-scu-card: remove audio-graph-scu-card on Doc ASoC: audio-graph-scu-card: remove audio-graph-scu-card ASoC: simple-scu-card: remove simple-scu-card on Doc ASoC: simple-scu-card: remove simple-scu-card ASoC: audio-graph-card: add 1 CPU : 1 Codec support again ASoC: audio-graph-card: add link_info ASoC: audio-graph-card: cleanup DAI link loop method - step1 ASoC: audio-graph-card: cleanup DAI link loop method - step2 ASoC: audio-graph-card: reduce naming prefix ASoC: simple-card: add asoc_simple_card_get_conversion() ASoC: simple-card: add 1 CPU : 1 Codec support again ASoC: simple-card: add link_info ASoC: simple-card: cleanup DAI link loop method - step1 ASoC: simple-card: cleanup DAI link loop method - step2 ASoC: simple-card: reduce naming prefix ASoC: rsnd: update BSDSR/BSDISR handling ASoC: soc.h: add explanation of legacy/modern style of dai_link ASoC: soc-core: add .num_platform for dai_link ASoC: soc-core: add new snd_soc_flush_all_delayed_work() ASoC: soc-core: merge card resources cleanup method ASoC: soc-core: reduce if/else nest on soc_probe_link_dais ASoC: soc-core: add soc_cleanup_component() ASoC: soc-core: use for_each_link_codecs() for dai_link codecs ASoC: simple-card: rename to asoc_simple_card_canonicalize_platform() ASoC: Fixup build error for mt6358 ASoC: soc-core: use for_each_link_codecs() for dai_link codecs V2 ASoC: rsnd: fixup MIX kctrl registration ASoC: rsnd: synchronize connection check for simple-card/audio-graph ASoC: rsnd: fixup TDM Split mode check for CTU Marco Felsch (1): ASoC: ssm2602: switch to SPDX identifier Mark Brown (7): Merge branch 'asoc-4.21' into HEAD Merge branch 'asoc-4.22' into asoc-5.0 Merge tag 'asoc-fix-v5.0-rc2' into asoc-5.1 ASoC: core: Fix multi-CODEC setups ASoC: wcd9335: Fix missing slimbus dependency Merge tag 'v5.0-rc4' into asoc-5.1 Merge branch 'asoc-5.0' into asoc-5.1 for dapm table Maruthi Srinivas Bayyavarapu (7): dt-bindings: ASoC: xlnx, audio-formatter: Document audio formatter bindings ASoC: xlnx: add pcm formatter platform driver ASoC: xlnx: enable audio formatter driver build dt-bindings: ASoC: xlnx, spdif: Document spdif bindings ASoC: xlnx: add SPDIF audio driver ASoC: xlnx: enable SPDIF driver build ASoC: xlnx: parse AES audio parameters Michał Mirosław (3): ASoC: wm8904: make the driver visible in Kconfig ASoC: wm8904: save model id directly in of_device_id.data ASoC: wm8904: enable MCLK in STANDBY Paul Cercueil (7): ASoC: codecs: Add jz4725b-codec driver dt-bindings: sound: Document jz4740-codec bindings dt-bindings: sound: Document jz4725b-codec bindings ASoC: codecs: jz4740: Use SPDX license notifier ASoC: codecs: jz4740: Add support for devicetree ASoC: codecs: Kconfig: Show knob, and depend on MIPS || COMPILE_TEST ASoC: codecs: jz4725b: Use C++ style comments in header Peter Ujfalusi (3): ASoC: ti: davinci-mcasp: No need for IS_MODULE/BUILTIN check for pcm driver ASoC: davinci-mcasp: Document GPIO support ASoC: ti: davinci-mcasp: Add support for GPIO mode of the pins Philipp Zabel (1): ASoC: ssm2602: Fix ADC powerup sequencing Pierre-Louis Bossart (45): ASoC: Intel: Skylake: remove useless cast ASoC: Intel: Skylake: simplify boolean tests ASoC: Intel: Haswell: remove unneeded semicolon ASoC: Intel: Haswell: assign booleans to true/false ASoC: Intel: Baytrail: remove unneeded variable ASoC: Intel: Baytrail: simplify boolean test ASoC: Intel: Atom: simplify boolean tests ASoC: Intel: boards: use snd_mask_set_format in all machine drivers ASoC: dmic: declare trigger function as static ASoC: max98090: remove unused constant variables ASoC: es8316: remove unused constant variables ASoC: codecs: fix kernel doc descriptions ASoC: max98383: fix boolean assignments to true/false ASoC: cs4271: fix boolean assignments ASoC: rt274: fix boolean tests ASoc: rt286: fix boolean tests ASoC: rt5640: fix boolean assignments ASoC: max98927: fix boolean assignments ASoC: rt5651: fix boolean assignments ASoC: nau8824: fix boolean assignment ASoC: tscs42xx.c: fix boolean test ASoC: mt6351: remove unneeded variable ASoC: da7219: fix endianness issues ASoC: da7219: use logical AND ASoC: rt298: fix boolean tests ASoC: soc-acpi: add static inline fallbacks when CONFIG_ACPI=n ASoC: add helper to change platform name for all dailinks ASoC: Intel: haswell: platform name fixup support ASoC: Intel: broadwell: platform name fixup support ASoC: Intel: bdw-rt5677: platform name fixup support ASoC: Intel: bytcr_rt5640: platform name fixup support ASoC: Intel: bytcr_rt5651: platform name fixup support ASoC: Intel: bytcht_da7213: platform name fixup support ASoC: Intel: bytcht_es8316: platform name fixup support ASoC: Intel: cht_bsw_max98090_ti: platform name fixup support ASoC: Intel: cht_bsw_nau8824: platform name fixup support ASoC: Intel: cht_bsw_rt5645: platform name fixup support ASoC: Intel: cht_bsw_rt5672: platform name fixup support ASoC: Intel: bxt_da7219_max98357a: platform name fixup support ASoC: Intel: bxt_rt298: platform name fixup support ASoC: Intel: glk_rt5682_max98357a: platform name fixup support ASoC: Intel: cht_bsw_rt5672: remove useless test ASoC: dapm: fix use-after-free issue with dailink sname ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables ASoC: dapm: harden use of lookup tables Ranjani Sridharan (2): ASoC: topology: add SND_SOC_DOBJ_GRAPH type for dapm routes ASoC: topology: modify dapm route loading routine and add dapm route unloading Russell King (1): ASoC: hdmi-codec: fix oops on re-probe Shengjiu Wang (1): ASoC: fsl_sai: Remove expensive print in irq handler Shuming Fan (1): ASoC: rt5682: Correct the setting while select ASRC clk for AD/DA filter Shunli Wang (4): ASoC: mediatek: mt8183: add platform driver ASoC: mediatek: mt8183: add audio afe document ASoC: mediatek: mt6358: add codec driver ASoC: mediatek: mt6358: add codec document Srinivas Kandagatla (9): ASoC: msm8916-wcd-digital: remove few unused variables ASoC: dt-bindings: update wcd9335 bindings. ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: apq8096: add slim support Stefan Agner (6): ASoC: imx-sgtl5000: put of nodes if finding codec fails ASoC: imx-sgtl5000: lower log level for potential probe deferral cases ASoC: imx-sgtl5000: don't print EPROBE_DEFER as error ASoC: fsl_spdif: don't print EPROBE_DEFER as error ASoC: imx-spdif: don't print EPROBE_DEFER as error ASoC: soc-core: remove error due to probe deferral Stephan Gerhold (3): ASoC: Intel: sst: Simplify is_byt_cr() ASoC: Intel: sst: Fallback to BYT-CR if IRQ 5 is missing ASoC: Intel: bytcr_rt5640: Add quirks for ASUS MeMO Pad 7 (ME176C) Wei Yongjun (1): ASoC: Intel: bytcht_es8316: use correct drvdata in snd_byt_cht_es8316_mc_remove() Yizhuo (2): ASoC: rt274: Variable "buf" in function rt274_jack_detect() could be uninitialized ASoC: rt5651: Variable "ret" in function rt5651_i2c_probe() could be uninitialized YueHaibing (1): ASoC: rockchip: fix platform_no_drv_owner.cocci warnings Zhiwei Jiang (1): ASoC: dapm: Add warnings for widget overwrite when adding route b-ak (1): ASoC: tlv320aic32x4: SND_SOC_DAPM_MICBIAS is deprecated kbuild test robot (1): ASoC: wcd9335: fix semicolon.cocci warnings wen yang (1): ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe Documentation/devicetree/bindings/sound/ak4458.txt | 2 +- .../bindings/sound/audio-graph-scu-card.txt | 123 - Documentation/devicetree/bindings/sound/cs4341.txt | 22 + .../bindings/sound/davinci-mcasp-audio.txt | 17 + .../devicetree/bindings/sound/fsl,micfil.txt | 32 + .../bindings/sound/google,cros-ec-codec.txt | 26 + .../bindings/sound/ingenic,jz4725b-codec.txt | 20 + .../bindings/sound/ingenic,jz4740-codec.txt | 20 + Documentation/devicetree/bindings/sound/mt6358.txt | 18 + .../devicetree/bindings/sound/mt8183-afe-pcm.txt | 36 + .../devicetree/bindings/sound/mtk-btcvsd-snd.txt | 24 + .../devicetree/bindings/sound/qcom,wcd9335.txt | 10 +- .../bindings/sound/rockchip,rk3328-codec.txt | 23 + .../devicetree/bindings/sound/sgtl5000.txt | 9 + .../devicetree/bindings/sound/simple-scu-card.txt | 94 - .../devicetree/bindings/sound/sprd-pcm.txt | 23 + .../bindings/sound/xlnx,audio-formatter.txt | 29 + .../devicetree/bindings/sound/xlnx,spdif.txt | 28 + MAINTAINERS | 12 +- include/sound/simple_card_utils.h | 4 +- include/sound/soc-acpi.h | 28 +- include/sound/soc-dapm.h | 27 +- include/sound/soc-topology.h | 8 +- include/sound/soc.h | 70 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 62 +- sound/soc/codecs/Makefile | 12 + sound/soc/codecs/adau7002.c | 45 + sound/soc/codecs/ak4118.c | 1 + sound/soc/codecs/ak4458.c | 79 +- sound/soc/codecs/cros_ec_codec.c | 441 ++ sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs4341.c | 346 ++ sound/soc/codecs/cs47l24.c | 52 +- sound/soc/codecs/da7219-aad.c | 2 +- sound/soc/codecs/da7219.c | 85 +- sound/soc/codecs/da7219.h | 1 + sound/soc/codecs/dmic.c | 4 +- sound/soc/codecs/es8316.c | 200 +- sound/soc/codecs/es8316.h | 7 + sound/soc/codecs/hdmi-codec.c | 4 +- sound/soc/codecs/jz4725b.c | 598 +++ sound/soc/codecs/jz4740.c | 26 +- sound/soc/codecs/max98090.c | 15 - sound/soc/codecs/max98373.c | 6 +- sound/soc/codecs/max98927.c | 8 +- sound/soc/codecs/msm8916-wcd-digital.c | 10 - sound/soc/codecs/mt6351.c | 4 +- sound/soc/codecs/mt6358.c | 2336 +++++++++ sound/soc/codecs/mt6358.h | 2314 +++++++++ sound/soc/codecs/nau8824.c | 2 +- sound/soc/codecs/nau8825.c | 1 + sound/soc/codecs/pcm512x.c | 32 +- sound/soc/codecs/rk3328_codec.c | 519 ++ sound/soc/codecs/rk3328_codec.h | 210 + sound/soc/codecs/rl6347a.c | 4 +- sound/soc/codecs/rt274.c | 19 +- sound/soc/codecs/rt286.c | 8 +- sound/soc/codecs/rt298.c | 12 +- sound/soc/codecs/rt5514.c | 1 + sound/soc/codecs/rt5640.c | 6 +- sound/soc/codecs/rt5645.c | 66 +- sound/soc/codecs/rt5651.c | 67 +- sound/soc/codecs/rt5651.h | 1 + sound/soc/codecs/rt5670.c | 54 - sound/soc/codecs/rt5677.c | 8 +- sound/soc/codecs/rt5682.c | 10 + sound/soc/codecs/sgtl5000.c | 19 +- sound/soc/codecs/ssm2602.c | 71 +- sound/soc/codecs/tlv320aic32x4.c | 30 +- sound/soc/codecs/tlv320aic32x4.h | 1 + sound/soc/codecs/tscs42xx.c | 2 +- sound/soc/codecs/wcd-clsh-v2.c | 576 +++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5244 ++++++++++++++++++++ sound/soc/codecs/wcd9335.h | 640 +++ sound/soc/codecs/wm5102.c | 64 +- sound/soc/codecs/wm5110.c | 80 +- sound/soc/codecs/wm8904.c | 21 +- sound/soc/codecs/wm8997.c | 60 +- sound/soc/codecs/wm8998.c | 60 +- sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 1 + sound/soc/fsl/fsl_micfil.c | 826 +++ sound/soc/fsl/fsl_micfil.h | 283 ++ sound/soc/fsl/fsl_sai.c | 8 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 11 +- sound/soc/fsl/imx-spdif.c | 4 +- sound/soc/generic/Kconfig | 17 - sound/soc/generic/Makefile | 4 - sound/soc/generic/audio-graph-card.c | 481 +- sound/soc/generic/audio-graph-scu-card.c | 501 -- sound/soc/generic/simple-card-utils.c | 25 +- sound/soc/generic/simple-card.c | 472 +- sound/soc/generic/simple-scu-card.c | 474 -- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 45 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 2 +- sound/soc/intel/atom/sst/sst_loader.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 3 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 14 + sound/soc/intel/boards/broadwell.c | 16 + sound/soc/intel/boards/bxt_da7219_max98357a.c | 13 + sound/soc/intel/boards/bxt_rt298.c | 13 + sound/soc/intel/boards/bytcht_da7213.c | 8 + sound/soc/intel/boards/bytcht_es8316.c | 336 +- sound/soc/intel/boards/bytcr_rt5640.c | 23 +- sound/soc/intel/boards/bytcr_rt5651.c | 114 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 12 + sound/soc/intel/boards/cht_bsw_nau8824.c | 12 + sound/soc/intel/boards/cht_bsw_rt5645.c | 9 + sound/soc/intel/boards/cht_bsw_rt5672.c | 30 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 23 +- sound/soc/intel/boards/haswell.c | 15 + sound/soc/intel/boards/kbl_da7219_max98927.c | 4 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 44 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 4 +- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 2 +- sound/soc/mediatek/Kconfig | 19 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/Makefile | 2 + sound/soc/mediatek/common/mtk-btcvsd.c | 1364 +++++ sound/soc/mediatek/mt8183/Makefile | 13 + sound/soc/mediatek/mt8183/mt8183-afe-clk.c | 611 +++ sound/soc/mediatek/mt8183/mt8183-afe-clk.h | 38 + sound/soc/mediatek/mt8183/mt8183-afe-common.h | 108 + sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 1237 +++++ sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 501 ++ sound/soc/mediatek/mt8183/mt8183-dai-hostless.c | 118 + sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 1040 ++++ sound/soc/mediatek/mt8183/mt8183-dai-pcm.c | 318 ++ sound/soc/mediatek/mt8183/mt8183-dai-tdm.c | 639 +++ sound/soc/mediatek/mt8183/mt8183-interconnection.h | 33 + sound/soc/mediatek/mt8183/mt8183-reg.h | 1666 +++++++ sound/soc/pxa/Makefile | 1 - sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/apq8096.c | 71 +- sound/soc/qcom/sdm845.c | 2 + sound/soc/sh/rcar/core.c | 48 +- sound/soc/sh/rcar/src.c | 125 +- sound/soc/soc-compress.c | 177 +- sound/soc/soc-core.c | 267 +- sound/soc/soc-dapm.c | 244 +- sound/soc/soc-pcm.c | 4 + sound/soc/soc-topology.c | 171 +- sound/soc/sprd/Kconfig | 6 + sound/soc/sprd/Makefile | 4 + sound/soc/sprd/sprd-pcm-dma.c | 562 +++ sound/soc/sprd/sprd-pcm-dma.h | 15 + sound/soc/sunxi/sun4i-codec.c | 149 +- sound/soc/ti/davinci-mcasp.c | 175 +- sound/soc/xilinx/Kconfig | 14 + sound/soc/xilinx/Makefile | 4 + sound/soc/xilinx/xlnx_formatter_pcm.c | 708 +++ sound/soc/xilinx/xlnx_spdif.c | 339 ++ 163 files changed, 27411 insertions(+), 2566 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt create mode 100644 Documentation/devicetree/bindings/sound/cs4341.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.txt create mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/mt6358.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt delete mode 100644 Documentation/devicetree/bindings/sound/simple-scu-card.txt create mode 100644 Documentation/devicetree/bindings/sound/sprd-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt create mode 100644 Documentation/devicetree/bindings/sound/xlnx,spdif.txt create mode 100644 sound/soc/codecs/cros_ec_codec.c create mode 100644 sound/soc/codecs/cs4341.c create mode 100644 sound/soc/codecs/jz4725b.c create mode 100644 sound/soc/codecs/mt6358.c create mode 100644 sound/soc/codecs/mt6358.h create mode 100644 sound/soc/codecs/rk3328_codec.c create mode 100644 sound/soc/codecs/rk3328_codec.h create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c create mode 100644 sound/soc/codecs/wcd9335.h create mode 100644 sound/soc/fsl/fsl_micfil.c create mode 100644 sound/soc/fsl/fsl_micfil.h delete mode 100644 sound/soc/generic/audio-graph-scu-card.c delete mode 100644 sound/soc/generic/simple-scu-card.c create mode 100644 sound/soc/mediatek/common/mtk-btcvsd.c create mode 100644 sound/soc/mediatek/mt8183/Makefile create mode 100644 sound/soc/mediatek/mt8183/mt8183-afe-clk.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-afe-clk.h create mode 100644 sound/soc/mediatek/mt8183/mt8183-afe-common.h create mode 100644 sound/soc/mediatek/mt8183/mt8183-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-dai-adda.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-dai-hostless.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-interconnection.h create mode 100644 sound/soc/mediatek/mt8183/mt8183-reg.h create mode 100644 sound/soc/sprd/Kconfig create mode 100644 sound/soc/sprd/Makefile create mode 100644 sound/soc/sprd/sprd-pcm-dma.c create mode 100644 sound/soc/sprd/sprd-pcm-dma.h create mode 100644 sound/soc/xilinx/xlnx_formatter_pcm.c create mode 100644 sound/soc/xilinx/xlnx_spdif.c

2 3

[alsa-devel] [GIT PULL] ASoC fixes for v5.0
by Mark Brown 08 Feb '19

08 Feb '19

The following changes since commit 4cb79ef9c6c4413427cd70afbb1f3bc01e9b7abf: ASoC: amd: Fix potential NULL pointer dereference (2019-01-15 19:06:23 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.0-rc5 for you to fetch changes up to 860b454c2c0cbda6892954f5cdbbb48931b3c8db: ASoC: samsung: Prevent clk_get_rate() calls in atomic context (2019-02-07 14:34:53 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.0 A selection of driver specific fixes here, along with a few core fixes: - A fixup for some MFD devices that were broken by the previous fixes for deferred probe. - A fix for potential out of bounds array accesses when ordering DAPM power/up down sequences. - Avoid use after free issue when unloading and reloading drivers using topologies. ---------------------------------------------------------------- Charles Keepax (1): ASoC: core: Allow soc_find_component lookups to match parent of_node Fabio Estevam (1): ASoC: MAINTAINERS: fsl: Change Fabio's email address Guennadi Liakhovetski (1): ASoC: topology: fix oops/use-after-free case with dai driver Jiada Wang (1): ASoC: rsnd: ssiu: correct shift bit for ssiu9 Kuninori Morimoto (2): ASoC: rsnd: fixup MIX kctrl registration ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check Pierre-Louis Bossart (1): ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables Russell King (1): ASoC: hdmi-codec: fix oops on re-probe Shuming Fan (1): ASoC: rt5682: Correct the setting while select ASRC clk for AD/DA filter Sylwester Nawrocki (1): ASoC: samsung: Prevent clk_get_rate() calls in atomic context MAINTAINERS | 4 ++-- sound/soc/codecs/hdmi-codec.c | 4 ++-- sound/soc/codecs/rt5682.c | 2 ++ sound/soc/samsung/i2s.c | 10 +++++----- sound/soc/sh/rcar/core.c | 8 ++++---- sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/sh/rcar/ssiu.c | 2 +- sound/soc/soc-core.c | 9 +++++++-- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++ sound/soc/soc-topology.c | 5 +++++ 10 files changed, 53 insertions(+), 17 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: doc: Fix typos in dpcm.rst
by Sylwester Nawrocki 08 Feb '19

08 Feb '19

This patch fixes a few typos in the DPCM documentation. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Documentation/sound/soc/dpcm.rst | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/Documentation/sound/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst index f6845b2278ea..77f67ded53de 100644 --- a/Documentation/sound/soc/dpcm.rst +++ b/Documentation/sound/soc/dpcm.rst @@ -13,7 +13,7 @@ drivers that expose several ALSA PCMs and can route to multiple DAIs. The DPCM runtime routing is determined by the ALSA mixer settings in the same way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM graph representing the DSP internal audio paths and uses the mixer settings to -determine the patch used by each ALSA PCM. +determine the path used by each ALSA PCM. DPCM re-uses all the existing component codec, platform and DAI drivers without any modifications. @@ -101,7 +101,7 @@ The audio driver processes this as follows :- 4. Machine driver or audio HAL enables the speaker path. -5. DPCM runs the PCM ops for startup(), hw_params(), prepapre() and +5. DPCM runs the PCM ops for startup(), hw_params(), prepare() and trigger(start) for DAI1 Speakers since the path is enabled. In this example, the machine driver or userspace audio HAL can alter the routing @@ -221,7 +221,7 @@ like a BT phone call :- This allows the host CPU to sleep while the DSP, MODEM DAI and the BT DAI are still in operation. -A BE DAI link can also set the codec to a dummy device if the code is a device +A BE DAI link can also set the codec to a dummy device if the codec is a device that is managed externally. Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the @@ -249,7 +249,7 @@ configuration. struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - /* The DSP will covert the FE rate to 48k, stereo */ + /* The DSP will convert the FE rate to 48k, stereo */ rate->min = rate->max = 48000; channels->min = channels->max = 2; @@ -386,5 +386,3 @@ This means creating a new FE that is connected with a virtual path to both DAI links. The DAI links will be started when the FE PCM is started and stopped when the FE PCM is stopped. Note that the FE PCM cannot read or write data in this configuration. - - -- 2.20.1

2 1

[alsa-devel] [PATCH v3 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by James Schulman 08 Feb '19

08 Feb '19

Add driver support for Cirrus Logic CS35L36 boosted speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs35l36.h | 43 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l36.c | 1958 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l36.h | 446 ++++++++++ 5 files changed, 2454 insertions(+) create mode 100644 include/sound/cs35l36.h create mode 100644 sound/soc/codecs/cs35l36.c create mode 100644 sound/soc/codecs/cs35l36.h diff --git a/include/sound/cs35l36.h b/include/sound/cs35l36.h new file mode 100644 index 0000000..8f8049d --- /dev/null +++ b/include/sound/cs35l36.h @@ -0,0 +1,43 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * linux/sound/cs35l36.h -- Platform data for CS35L36 + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H +#define __CS35L36_H + +struct cs35l36_vpbr_cfg { + bool is_present; + bool vpbr_en; + int vpbr_thld; + int vpbr_atk_rate; + int vpbr_atk_vol; + int vpbr_max_attn; + int vpbr_wait; + int vpbr_rel_rate; + int vpbr_mute_en; +}; + +struct cs35l36_platform_data { + bool multi_amp_mode; + bool dcm_mode; + bool amp_pcm_inv; + bool imon_pol_inv; + bool vmon_pol_inv; + int boost_ind; + int bst_vctl; + int bst_vctl_sel; + int bst_ipk; + bool extern_boost; + int temp_warn_thld; + int irq_drv_sel; + int irq_gpio_sel; + struct cs35l36_vpbr_cfg vpbr_config; +}; + +#endif /* __CS35L36_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a15710c..3946222 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,6 +55,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C select SND_SOC_CS35L35 if I2C + select SND_SOC_CS35L36 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -483,6 +484,10 @@ config SND_SOC_CS35L35 tristate "Cirrus Logic CS35L35 CODEC" depends on I2C +config SND_SOC_CS35L36 + tristate "Cirrus Logic CS35L36 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d7a597..427b180 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -47,6 +47,7 @@ snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs35l35-objs := cs35l35.o +snd-soc-cs35l36-objs := cs35l36.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -318,6 +319,7 @@ obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o +obj-$(CONFIG_SND_SOC_CS35L36) += snd-soc-cs35l36.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c new file mode 100644 index 0000000..4f880a6 --- /dev/null +++ b/sound/soc/codecs/cs35l36.c @@ -0,0 +1,1958 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// cs35l36.c -- CS35L36 ALSA SoC audio driver +// +// Copyright 2018 Cirrus Logic, Inc. +// +// Author: James Schulman <james.schulman(a)cirrus.com> + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l36.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l36.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L36_VALID_PDATA 0x80000000 + +static const char * const cs35l36_supplies[] = { + "VA", + "VP", +}; + +struct cs35l36_private { + struct device *dev; + struct cs35l36_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int clksrc; + int chip_version; + int rev_id; + int ldm_mode_sel; + struct gpio_desc *reset_gpio; +}; + +struct cs35l36_pll_config { + int freq; + int clk_cfg; + int fll_igain; +}; + +static const struct cs35l36_pll_config cs35l36_pll_sysclk[] = { + {32768, 0x00, 0x05}, + {8000, 0x01, 0x03}, + {11025, 0x02, 0x03}, + {12000, 0x03, 0x03}, + {16000, 0x04, 0x04}, + {22050, 0x05, 0x04}, + {24000, 0x06, 0x04}, + {32000, 0x07, 0x05}, + {44100, 0x08, 0x05}, + {48000, 0x09, 0x05}, + {88200, 0x0A, 0x06}, + {96000, 0x0B, 0x06}, + {128000, 0x0C, 0x07}, + {176400, 0x0D, 0x07}, + {192000, 0x0E, 0x07}, + {256000, 0x0F, 0x08}, + {352800, 0x10, 0x08}, + {384000, 0x11, 0x08}, + {512000, 0x12, 0x09}, + {705600, 0x13, 0x09}, + {750000, 0x14, 0x09}, + {768000, 0x15, 0x09}, + {1000000, 0x16, 0x0A}, + {1024000, 0x17, 0x0A}, + {1200000, 0x18, 0x0A}, + {1411200, 0x19, 0x0A}, + {1500000, 0x1A, 0x0A}, + {1536000, 0x1B, 0x0A}, + {2000000, 0x1C, 0x0A}, + {2048000, 0x1D, 0x0A}, + {2400000, 0x1E, 0x0A}, + {2822400, 0x1F, 0x0A}, + {3000000, 0x20, 0x0A}, + {3072000, 0x21, 0x0A}, + {3200000, 0x22, 0x0A}, + {4000000, 0x23, 0x0A}, + {4096000, 0x24, 0x0A}, + {4800000, 0x25, 0x0A}, + {5644800, 0x26, 0x0A}, + {6000000, 0x27, 0x0A}, + {6144000, 0x28, 0x0A}, + {6250000, 0x29, 0x08}, + {6400000, 0x2A, 0x0A}, + {6500000, 0x2B, 0x08}, + {6750000, 0x2C, 0x09}, + {7526400, 0x2D, 0x0A}, + {8000000, 0x2E, 0x0A}, + {8192000, 0x2F, 0x0A}, + {9600000, 0x30, 0x0A}, + {11289600, 0x31, 0x0A}, + {12000000, 0x32, 0x0A}, + {12288000, 0x33, 0x0A}, + {12500000, 0x34, 0x08}, + {12800000, 0x35, 0x0A}, + {13000000, 0x36, 0x0A}, + {13500000, 0x37, 0x0A}, + {19200000, 0x38, 0x0A}, + {22579200, 0x39, 0x0A}, + {24000000, 0x3A, 0x0A}, + {24576000, 0x3B, 0x0A}, + {25000000, 0x3C, 0x0A}, + {25600000, 0x3D, 0x0A}, + {26000000, 0x3E, 0x0A}, + {27000000, 0x3F, 0x0A}, +}; + +struct reg_default cs35l36_reg[] = { + {CS35L36_TESTKEY_CTRL, 0x00000000}, + {CS35L36_USERKEY_CTL, 0x00000000}, + {CS35L36_OTP_CTRL1, 0x00002460}, + {CS35L36_OTP_CTRL2, 0x00000000}, + {CS35L36_OTP_CTRL3, 0x00000000}, + {CS35L36_OTP_CTRL4, 0x00000000}, + {CS35L36_OTP_CTRL5, 0x00000000}, + {CS35L36_PAC_CTL1, 0x00000004}, + {CS35L36_PAC_CTL2, 0x00000000}, + {CS35L36_PAC_CTL3, 0x00000000}, + {CS35L36_PWR_CTRL1, 0x00000000}, + {CS35L36_PWR_CTRL2, 0x00003321}, + {CS35L36_PWR_CTRL3, 0x01000010}, + {CS35L36_CTRL_OVRRIDE, 0x00000002}, + {CS35L36_AMP_OUT_MUTE, 0x00000000}, + {CS35L36_OTP_TRIM_STATUS, 0x00000000}, + {CS35L36_DISCH_FILT, 0x00000000}, + {CS35L36_PROTECT_REL_ERR, 0x00000000}, + {CS35L36_PAD_INTERFACE, 0x00000038}, + {CS35L36_PLL_CLK_CTRL, 0x00000010}, + {CS35L36_GLOBAL_CLK_CTRL, 0x00000003}, + {CS35L36_ADC_CLK_CTRL, 0x00000000}, + {CS35L36_SWIRE_CLK_CTRL, 0x00000000}, + {CS35L36_SP_SCLK_CLK_CTRL, 0x00000000}, + {CS35L36_MDSYNC_EN, 0x00000000}, + {CS35L36_MDSYNC_TX_ID, 0x00000000}, + {CS35L36_MDSYNC_PWR_CTRL, 0x00000000}, + {CS35L36_MDSYNC_DATA_TX, 0x00000000}, + {CS35L36_MDSYNC_TX_STATUS, 0x00000002}, + {CS35L36_MDSYNC_RX_STATUS, 0x00000000}, + {CS35L36_MDSYNC_ERR_STATUS, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL1, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL2, 0x00000001}, + {CS35L36_BSTCVRT_PEAK_CUR, 0x0000004A}, + {CS35L36_BSTCVRT_SFT_RAMP, 0x00000003}, + {CS35L36_BSTCVRT_COEFF, 0x00002424}, + {CS35L36_BSTCVRT_SLOPE_LBST, 0x00005800}, + {CS35L36_BSTCVRT_SW_FREQ, 0x00010000}, + {CS35L36_BSTCVRT_DCM_CTRL, 0x00002001}, + {CS35L36_BSTCVRT_DCM_MODE_FORCE, 0x00000000}, + {CS35L36_BSTCVRT_OVERVOLT_CTRL, 0x00000130}, + {CS35L36_VPI_LIMIT_MODE, 0x00000000}, + {CS35L36_VPI_LIMIT_MINMAX, 0x00003000}, + {CS35L36_VPI_VP_THLD, 0x00101010}, + {CS35L36_VPI_TRACK_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_MODE_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_STEPS, 0x00000000}, + {CS35L36_VI_SPKMON_FILT, 0x00000003}, + {CS35L36_VI_SPKMON_GAIN, 0x00000909}, + {CS35L36_VI_SPKMON_IP_SEL, 0x00000000}, + {CS35L36_DTEMP_WARN_THLD, 0x00000002}, + {CS35L36_DTEMP_STATUS, 0x00000000}, + {CS35L36_VPVBST_FS_SEL, 0x00000001}, + {CS35L36_VPVBST_VP_CTRL, 0x000001C0}, + {CS35L36_VPVBST_VBST_CTRL, 0x000001C0}, + {CS35L36_ASP_TX_PIN_CTRL, 0x00000028}, + {CS35L36_ASP_RATE_CTRL, 0x00090000}, + {CS35L36_ASP_FORMAT, 0x00000002}, + {CS35L36_ASP_FRAME_CTRL, 0x00180018}, + {CS35L36_ASP_TX1_TX2_SLOT, 0x00010000}, + {CS35L36_ASP_TX3_TX4_SLOT, 0x00030002}, + {CS35L36_ASP_TX5_TX6_SLOT, 0x00050004}, + {CS35L36_ASP_TX7_TX8_SLOT, 0x00070006}, + {CS35L36_ASP_RX1_SLOT, 0x00000000}, + {CS35L36_ASP_RX_TX_EN, 0x00000000}, + {CS35L36_ASP_RX1_SEL, 0x00000008}, + {CS35L36_ASP_TX1_SEL, 0x00000018}, + {CS35L36_ASP_TX2_SEL, 0x00000019}, + {CS35L36_ASP_TX3_SEL, 0x00000028}, + {CS35L36_ASP_TX4_SEL, 0x00000029}, + {CS35L36_ASP_TX5_SEL, 0x00000020}, + {CS35L36_ASP_TX6_SEL, 0x00000000}, + {CS35L36_SWIRE_P1_TX1_SEL, 0x00000018}, + {CS35L36_SWIRE_P1_TX2_SEL, 0x00000019}, + {CS35L36_SWIRE_P2_TX1_SEL, 0x00000028}, + {CS35L36_SWIRE_P2_TX2_SEL, 0x00000029}, + {CS35L36_SWIRE_P2_TX3_SEL, 0x00000020}, + {CS35L36_SWIRE_DP1_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP2_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP3_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_PCM_RX_DATA, 0x00000000}, + {CS35L36_SWIRE_FS_SEL, 0x00000001}, + {CS35L36_AMP_DIG_VOL_CTRL, 0x00008000}, + {CS35L36_VPBR_CFG, 0x02AA1905}, + {CS35L36_VBBR_CFG, 0x02AA1905}, + {CS35L36_VPBR_STATUS, 0x00000000}, + {CS35L36_VBBR_STATUS, 0x00000000}, + {CS35L36_OVERTEMP_CFG, 0x00000001}, + {CS35L36_AMP_ERR_VOL, 0x00000000}, + {CS35L36_CLASSH_CFG, 0x000B0405}, + {CS35L36_CLASSH_FET_DRV_CFG, 0x00000111}, + {CS35L36_NG_CFG, 0x00000033}, + {CS35L36_AMP_GAIN_CTRL, 0x00000273}, + {CS35L36_PWM_MOD_IO_CTRL, 0x00000000}, + {CS35L36_PWM_MOD_STATUS, 0x00000000}, + {CS35L36_DAC_MSM_CFG, 0x00000000}, + {CS35L36_AMP_SLOPE_CTRL, 0x00000B00}, + {CS35L36_AMP_PDM_VOLUME, 0x00000000}, + {CS35L36_AMP_PDM_RATE_CTRL, 0x00000000}, + {CS35L36_PDM_CH_SEL, 0x00000000}, + {CS35L36_AMP_NG_CTRL, 0x0000212F}, + {CS35L36_PDM_HIGHFILT_CTRL, 0x00000000}, + {CS35L36_PAC_INT0_CTRL, 0x00000001}, + {CS35L36_PAC_INT1_CTRL, 0x00000001}, + {CS35L36_PAC_INT2_CTRL, 0x00000001}, + {CS35L36_PAC_INT3_CTRL, 0x00000001}, + {CS35L36_PAC_INT4_CTRL, 0x00000001}, + {CS35L36_PAC_INT5_CTRL, 0x00000001}, + {CS35L36_PAC_INT6_CTRL, 0x00000001}, + {CS35L36_PAC_INT7_CTRL, 0x00000001}, +}; + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_OTP_MEM30: + case CS35L36_OTP_CTRL1: + case CS35L36_OTP_CTRL2: + case CS35L36_OTP_CTRL3: + case CS35L36_OTP_CTRL4: + case CS35L36_OTP_CTRL5: + case CS35L36_PAC_CTL1: + case CS35L36_PAC_CTL2: + case CS35L36_PAC_CTL3: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_PWR_CTRL1: + case CS35L36_PWR_CTRL2: + case CS35L36_PWR_CTRL3: + case CS35L36_CTRL_OVRRIDE: + case CS35L36_AMP_OUT_MUTE: + case CS35L36_OTP_TRIM_STATUS: + case CS35L36_DISCH_FILT: + case CS35L36_PROTECT_REL_ERR: + case CS35L36_PAD_INTERFACE: + case CS35L36_PLL_CLK_CTRL: + case CS35L36_GLOBAL_CLK_CTRL: + case CS35L36_ADC_CLK_CTRL: + case CS35L36_SWIRE_CLK_CTRL: + case CS35L36_SP_SCLK_CLK_CTRL: + case CS35L36_TST_FS_MON0: + case CS35L36_MDSYNC_EN: + case CS35L36_MDSYNC_TX_ID: + case CS35L36_MDSYNC_PWR_CTRL: + case CS35L36_MDSYNC_DATA_TX: + case CS35L36_MDSYNC_TX_STATUS: + case CS35L36_MDSYNC_RX_STATUS: + case CS35L36_MDSYNC_ERR_STATUS: + case CS35L36_BSTCVRT_VCTRL1: + case CS35L36_BSTCVRT_VCTRL2: + case CS35L36_BSTCVRT_PEAK_CUR: + case CS35L36_BSTCVRT_SFT_RAMP: + case CS35L36_BSTCVRT_COEFF: + case CS35L36_BSTCVRT_SLOPE_LBST: + case CS35L36_BSTCVRT_SW_FREQ: + case CS35L36_BSTCVRT_DCM_CTRL: + case CS35L36_BSTCVRT_DCM_MODE_FORCE: + case CS35L36_BSTCVRT_OVERVOLT_CTRL: + case CS35L36_BST_TST_MANUAL: + case CS35L36_BST_ANA2_TEST: + case CS35L36_VPI_LIMIT_MODE: + case CS35L36_VPI_LIMIT_MINMAX: + case CS35L36_VPI_VP_THLD: + case CS35L36_VPI_TRACK_CTRL: + case CS35L36_VPI_TRIG_MODE_CTRL: + case CS35L36_VPI_TRIG_STEPS: + case CS35L36_VI_SPKMON_FILT: + case CS35L36_VI_SPKMON_GAIN: + case CS35L36_VI_SPKMON_IP_SEL: + case CS35L36_DTEMP_WARN_THLD: + case CS35L36_DTEMP_STATUS: + case CS35L36_VPVBST_FS_SEL: + case CS35L36_VPVBST_VP_CTRL: + case CS35L36_VPVBST_VBST_CTRL: + case CS35L36_ASP_TX_PIN_CTRL: + case CS35L36_ASP_RATE_CTRL: + case CS35L36_ASP_FORMAT: + case CS35L36_ASP_FRAME_CTRL: + case CS35L36_ASP_TX1_TX2_SLOT: + case CS35L36_ASP_TX3_TX4_SLOT: + case CS35L36_ASP_TX5_TX6_SLOT: + case CS35L36_ASP_TX7_TX8_SLOT: + case CS35L36_ASP_RX1_SLOT: + case CS35L36_ASP_RX_TX_EN: + case CS35L36_ASP_RX1_SEL: + case CS35L36_ASP_TX1_SEL: + case CS35L36_ASP_TX2_SEL: + case CS35L36_ASP_TX3_SEL: + case CS35L36_ASP_TX4_SEL: + case CS35L36_ASP_TX5_SEL: + case CS35L36_ASP_TX6_SEL: + case CS35L36_SWIRE_P1_TX1_SEL: + case CS35L36_SWIRE_P1_TX2_SEL: + case CS35L36_SWIRE_P2_TX1_SEL: + case CS35L36_SWIRE_P2_TX2_SEL: + case CS35L36_SWIRE_P2_TX3_SEL: + case CS35L36_SWIRE_DP1_FIFO_CFG: + case CS35L36_SWIRE_DP2_FIFO_CFG: + case CS35L36_SWIRE_DP3_FIFO_CFG: + case CS35L36_SWIRE_PCM_RX_DATA: + case CS35L36_SWIRE_FS_SEL: + case CS35L36_AMP_DIG_VOL_CTRL: + case CS35L36_VPBR_CFG: + case CS35L36_VBBR_CFG: + case CS35L36_VPBR_STATUS: + case CS35L36_VBBR_STATUS: + case CS35L36_OVERTEMP_CFG: + case CS35L36_AMP_ERR_VOL: + case CS35L36_CLASSH_CFG: + case CS35L36_CLASSH_FET_DRV_CFG: + case CS35L36_NG_CFG: + case CS35L36_AMP_GAIN_CTRL: + case CS35L36_PWM_MOD_IO_CTRL: + case CS35L36_PWM_MOD_STATUS: + case CS35L36_DAC_MSM_CFG: + case CS35L36_AMP_SLOPE_CTRL: + case CS35L36_AMP_PDM_VOLUME: + case CS35L36_AMP_PDM_RATE_CTRL: + case CS35L36_PDM_CH_SEL: + case CS35L36_AMP_NG_CTRL: + case CS35L36_PDM_HIGHFILT_CTRL: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + case CS35L36_PAC_INT0_CTRL: + case CS35L36_PAC_INT1_CTRL: + case CS35L36_PAC_INT2_CTRL: + case CS35L36_PAC_INT3_CTRL: + case CS35L36_PAC_INT4_CTRL: + case CS35L36_PAC_INT5_CTRL: + case CS35L36_PAC_INT6_CTRL: + case CS35L36_PAC_INT7_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +bool cs35l36_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_TST_FS_MON0: + return true; + default: + return false; + } +} + +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); + +static const char * const cs35l36_pcm_sftramp_text[] = { + "Off", ".5ms", "1ms", "2ms", "4ms", "8ms", "15ms", "30ms"}; + +static SOC_ENUM_SINGLE_DECL(pcm_sft_ramp, CS35L36_AMP_DIG_VOL_CTRL, 0, + cs35l36_pcm_sftramp_text); + +static int cs35l36_ldm_sel_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = cs35l36->ldm_mode_sel; + + return 0; +} + +static int cs35l36_ldm_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int val = (ucontrol->value.integer.value[0]) ? CS35L36_NG_AMP_EN_MASK : + 0; + + cs35l36->ldm_mode_sel = val; + + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_AMP_EN_MASK, val); + + return 0; +} + +static const struct snd_kcontrol_new cs35l36_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, + 3, 0x4D0, 0x390, dig_vol_tlv), + SOC_SINGLE_TLV("Analog PCM Volume", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, + amp_gain_tlv), + SOC_ENUM("PCM Soft Ramp", pcm_sft_ramp), + SOC_SINGLE("Amp Gain Zero-Cross Switch", CS35L36_AMP_GAIN_CTRL, + CS35L36_AMP_ZC_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Enter Ramp Switch", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_ENTER_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Exit Ramp Switch", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_EXIT_SHIFT, 1, 0), + SOC_SINGLE_BOOL_EXT("LDM Select Switch", 0, cs35l36_ldm_sel_get, + cs35l36_ldm_sel_put), +}; + +static int cs35l36_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + u32 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 1 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); + + if (WARN_ON_ONCE(reg & CS35L36_PLL_UNLOCK_MASK)) + dev_crit(cs35l36->dev, "PLL Unlocked\n"); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_PCM); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 0 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_ZERO); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 1 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 0 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_boost_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_EN << + CS35L36_BST_EN_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_DIS_VP << + CS35L36_BST_EN_SHIFT); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const char * const cs35l36_chan_text[] = { + "RX1", + "RX2", +}; + +static SOC_ENUM_SINGLE_DECL(chansel_enum, CS35L36_ASP_RX1_SLOT, 0, + cs35l36_chan_text); + +static const struct snd_kcontrol_new cs35l36_chan_mux = + SOC_DAPM_ENUM("Input Mux", chansel_enum); + +static const struct snd_kcontrol_new amp_enable_ctrl = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_SHIFT, 1, 1); + +static const struct snd_kcontrol_new boost_ctrl = + SOC_DAPM_SINGLE_VIRT("Switch", 1); + +static const char * const asp_tx_src_text[] = { + "Zero Fill", "ASPRX1", "VMON", "IMON", "ERRVOL", "VPMON", "VBSTMON" +}; + +static const unsigned int asp_tx_src_values[] = { + 0x00, 0x08, 0x18, 0x19, 0x20, 0x28, 0x29 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx1_src_enum, CS35L36_ASP_TX1_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx1_src = + SOC_DAPM_ENUM("ASPTX1SRC", asp_tx1_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx2_src_enum, CS35L36_ASP_TX2_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx2_src = + SOC_DAPM_ENUM("ASPTX2SRC", asp_tx2_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx3_src_enum, CS35L36_ASP_TX3_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx3_src = + SOC_DAPM_ENUM("ASPTX3SRC", asp_tx3_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx4_src_enum, CS35L36_ASP_TX4_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx4_src = + SOC_DAPM_ENUM("ASPTX4SRC", asp_tx4_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx5_src_enum, CS35L36_ASP_TX5_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx5_src = + SOC_DAPM_ENUM("ASPTX5SRC", asp_tx5_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx6_src_enum, CS35L36_ASP_TX6_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx6_src = + SOC_DAPM_ENUM("ASPTX6SRC", asp_tx6_src_enum); + +static const struct snd_soc_dapm_widget cs35l36_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Channel Mux", SND_SOC_NOPM, 0, 0, &cs35l36_chan_mux), + SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, CS35L36_ASP_RX_TX_EN, 16, 0), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L36_PWR_CTRL2, 0, 0, NULL, 0, + cs35l36_main_amp_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 1, &amp_enable_ctrl), + SND_SOC_DAPM_MIXER("CLASS H", CS35L36_PWR_CTRL3, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH_E("BOOST Enable", SND_SOC_NOPM, 0, 0, &boost_ctrl, + cs35l36_boost_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_AIF_OUT("ASPTX1", NULL, 0, CS35L36_ASP_RX_TX_EN, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX2", NULL, 1, CS35L36_ASP_RX_TX_EN, 1, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 2, CS35L36_ASP_RX_TX_EN, 2, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 3, CS35L36_ASP_RX_TX_EN, 3, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX5", NULL, 4, CS35L36_ASP_RX_TX_EN, 4, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX6", NULL, 5, CS35L36_ASP_RX_TX_EN, 5, 0), + + SND_SOC_DAPM_MUX("ASPTX1SRC", SND_SOC_NOPM, 0, 0, &asp_tx1_src), + SND_SOC_DAPM_MUX("ASPTX2SRC", SND_SOC_NOPM, 0, 0, &asp_tx2_src), + SND_SOC_DAPM_MUX("ASPTX3SRC", SND_SOC_NOPM, 0, 0, &asp_tx3_src), + SND_SOC_DAPM_MUX("ASPTX4SRC", SND_SOC_NOPM, 0, 0, &asp_tx4_src), + SND_SOC_DAPM_MUX("ASPTX5SRC", SND_SOC_NOPM, 0, 0, &asp_tx5_src), + SND_SOC_DAPM_MUX("ASPTX6SRC", SND_SOC_NOPM, 0, 0, &asp_tx6_src), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L36_PWR_CTRL2, 12, 0), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L36_PWR_CTRL2, 13, 0), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L36_PWR_CTRL2, 8, 0), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L36_PWR_CTRL2, 9, 0), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("VSENSE"), +}; + +static const struct snd_soc_dapm_route cs35l36_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "VSENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + + {"ASPTX1SRC", "IMON", "IMON ADC"}, + {"ASPTX1SRC", "VMON", "VMON ADC"}, + {"ASPTX1SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX1SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX2SRC", "IMON", "IMON ADC"}, + {"ASPTX2SRC", "VMON", "VMON ADC"}, + {"ASPTX2SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX2SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX3SRC", "IMON", "IMON ADC"}, + {"ASPTX3SRC", "VMON", "VMON ADC"}, + {"ASPTX3SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX3SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX4SRC", "IMON", "IMON ADC"}, + {"ASPTX4SRC", "VMON", "VMON ADC"}, + {"ASPTX4SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX4SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX5SRC", "IMON", "IMON ADC"}, + {"ASPTX5SRC", "VMON", "VMON ADC"}, + {"ASPTX5SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX5SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX6SRC", "IMON", "IMON ADC"}, + {"ASPTX6SRC", "VMON", "VMON ADC"}, + {"ASPTX6SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX6SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX1", NULL, "ASPTX1SRC"}, + {"ASPTX2", NULL, "ASPTX2SRC"}, + {"ASPTX3", NULL, "ASPTX3SRC"}, + {"ASPTX4", NULL, "ASPTX4SRC"}, + {"ASPTX5", NULL, "ASPTX5SRC"}, + {"ASPTX6", NULL, "ASPTX6SRC"}, + + {"AMP Capture", NULL, "ASPTX1"}, + {"AMP Capture", NULL, "ASPTX2"}, + {"AMP Capture", NULL, "ASPTX3"}, + {"AMP Capture", NULL, "ASPTX4"}, + {"AMP Capture", NULL, "ASPTX5"}, + {"AMP Capture", NULL, "ASPTX6"}, + + {"AMP Enable", "Switch", "AMP Playback"}, + {"SDIN", NULL, "AMP Enable"}, + {"Channel Mux", "RX1", "SDIN"}, + {"Channel Mux", "RX2", "SDIN"}, + {"BOOST Enable", "Switch", "Channel Mux"}, + {"CLASS H", NULL, "BOOST Enable"}, + {"Main AMP", NULL, "Channel Mux"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, + unsigned int fmt) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component_dai->component); + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode, clk_frc; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + slave_mode = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_MSTR_MASK, + slave_mode << CS35L36_SCLK_MSTR_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_MSTR_MASK, + slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + clk_frc = 1; + break; + case SND_SOC_DAIFMT_GATED: + clk_frc = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_FRC_MASK, clk_frc << + CS35L36_SCLK_FRC_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_FRC_MASK, clk_frc << + CS35L36_LRCLK_FRC_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + asp_fmt = 0; + break; + case SND_SOC_DAIFMT_I2S: + asp_fmt = 2; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + lrclk_fmt = 1; + sclk_fmt = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + lrclk_fmt = 0; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + lrclk_fmt = 1; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + lrclk_fmt = 0; + sclk_fmt = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_INV_MASK, + lrclk_fmt << CS35L36_LRCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_INV_MASK, + sclk_fmt << CS35L36_SCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FORMAT, + CS35L36_ASP_FMT_MASK, asp_fmt); + + return 0; +} + +struct cs35l36_global_fs_config { + int rate; + int fs_cfg; +}; + +static const struct cs35l36_global_fs_config cs35l36_fs_rates[] = { + {12000, 0x01}, + {24000, 0x02}, + {48000, 0x03}, + {96000, 0x04}, + {192000, 0x05}, + {384000, 0x06}, + {11025, 0x09}, + {22050, 0x0A}, + {44100, 0x0B}, + {88200, 0x0C}, + {176400, 0x0D}, + {8000, 0x11}, + {16000, 0x12}, + {32000, 0x13}, +}; + +static int cs35l36_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(dai->component); + unsigned int asp_width, global_fs = params_rate(params); + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_fs_rates); i++) { + if (global_fs == cs35l36_fs_rates[i].rate) + regmap_update_bits(cs35l36->regmap, + CS35L36_GLOBAL_CLK_CTRL, + CS35L36_GLOBAL_FS_MASK, + cs35l36_fs_rates[i].fs_cfg << + CS35L36_GLOBAL_FS_SHIFT); + } + + switch (params_width(params)) { + case 16: + asp_width = CS35L36_ASP_WIDTH_16; + break; + case 24: + asp_width = CS35L36_ASP_WIDTH_24; + break; + case 32: + asp_width = CS35L36_ASP_WIDTH_32; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_RX_WIDTH_MASK, + asp_width << CS35L36_ASP_RX_WIDTH_SHIFT); + } else { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_TX_WIDTH_MASK, + asp_width << CS35L36_ASP_TX_WIDTH_SHIFT); + } + + return 0; +} + +static int cs35l36_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int fs1, fs2; + + if (freq > CS35L36_FS_NOM_6MHZ) { + fs1 = CS35L36_FS1_DEFAULT_VAL; + fs2 = CS35L36_FS2_DEFAULT_VAL; + } else { + fs1 = 3 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + fs2 = 5 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + } + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_TST_FS_MON0, + CS35L36_FS1_WINDOW_MASK | CS35L36_FS2_WINDOW_MASK, + fs1 | (fs2 << CS35L36_FS2_WINDOW_SHIFT)); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + return 0; +} + +static const struct cs35l36_pll_config *cs35l36_get_clk_config( + struct cs35l36_private *cs35l36, int freq) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_pll_sysclk); i++) { + if (cs35l36_pll_sysclk[i].freq == freq) + return &cs35l36_pll_sysclk[i]; + } + + return NULL; +} + +static const unsigned int cs35l36_src_rates[] = { + 8000, 12000, 11025, 16000, 22050, 24000, 32000, + 44100, 48000, 88200, 96000, 176400, 192000, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { + .count = ARRAY_SIZE(cs35l36_src_rates), + .list = cs35l36_src_rates, +}; + +static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l36_ops = { + .startup = cs35l36_pcm_startup, + .set_fmt = cs35l36_set_dai_fmt, + .hw_params = cs35l36_pcm_hw_params, + .set_sysclk = cs35l36_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l36_dai[] = { + { + .name = "cs35l36-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_RX_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_TX_FORMATS, + }, + .ops = &cs35l36_ops, + .symmetric_rates = 1, + }, +}; + +static int cs35l36_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + const struct cs35l36_pll_config *clk_cfg; + int prev_clksrc; + bool pdm_switch; + + prev_clksrc = cs35l36->clksrc; + + switch (clk_id) { + case 0: + cs35l36->clksrc = CS35L36_PLLSRC_SCLK; + break; + case 1: + cs35l36->clksrc = CS35L36_PLLSRC_LRCLK; + break; + case 2: + cs35l36->clksrc = CS35L36_PLLSRC_PDMCLK; + break; + case 3: + cs35l36->clksrc = CS35L36_PLLSRC_SELF; + break; + case 4: + cs35l36->clksrc = CS35L36_PLLSRC_MCLK; + break; + default: + return -EINVAL; + } + + clk_cfg = cs35l36_get_clk_config(cs35l36, freq); + if (clk_cfg == NULL) { + dev_err(component->dev, "Invalid CLK Config Freq: %d\n", freq); + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 1 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_REFCLK_FREQ_MASK, + clk_cfg->clk_cfg << CS35L36_REFCLK_FREQ_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 0 << CS35L36_PLL_REFCLK_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_CLK_SEL_MASK, + cs35l36->clksrc); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 0 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 1 << CS35L36_PLL_REFCLK_EN_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_write(cs35l36->regmap, CS35L36_DCO_CTRL, 0x00036DA8); + regmap_write(cs35l36->regmap, CS35L36_MISC_CTRL, 0x0100EE0E); + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_IGAIN_MASK, + CS35L36_PLL_IGAIN << + CS35L36_PLL_IGAIN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_FFL_IGAIN_MASK, + clk_cfg->fll_igain); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->clksrc == CS35L36_PLLSRC_PDMCLK) { + pdm_switch = cs35l36->ldm_mode_sel && + (prev_clksrc != CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 1 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } else { + pdm_switch = cs35l36->ldm_mode_sel && + (prev_clksrc == CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 0 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + + return 0; +} + +static int cs35l36_boost_inductor(struct cs35l36_private *cs35l36, int inductor) +{ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K1_MASK, 0x3C); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K2_MASK, + 0x3C << CS35L36_BSTCVRT_K2_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SW_FREQ, + CS35L36_BSTCVRT_CCMFREQ_MASK, 0x00); + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x75 << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x00); + break; + case 1200: /* 1.2 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x6B << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x01); + break; + default: + dev_err(cs35l36->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_component_probe(struct snd_soc_component *component) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + if ((cs35l36->rev_id == CS35L36_REV_A0) && cs35l36->pdata.dcm_mode) { + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_DCM_CTRL, + CS35L36_DCM_AUTO_MASK, + CS35L36_DCM_AUTO_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_MASK, + 0 << CS35L36_BST_MAN_IPKCOMP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_EN_MASK, + CS35L36_BST_MAN_IPKCOMP_EN_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->pdata.amp_pcm_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_DIG_VOL_CTRL, + CS35L36_AMP_PCM_INV_MASK, + CS35L36_AMP_PCM_INV_MASK); + + if (cs35l36->pdata.multi_amp_mode) + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_ASP_TX_HIZ_MASK, + CS35L36_ASP_TX_HIZ_MASK); + + if (cs35l36->pdata.imon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_IMON_POL_MASK, 0); + + if (cs35l36->pdata.vmon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_VMON_POL_MASK, 0); + + if (cs35l36->pdata.bst_vctl) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + cs35l36->pdata.bst_vctl); + + if (cs35l36->pdata.bst_vctl_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L35_BSTCVRT_CTL_SEL_MASK, + cs35l36->pdata.bst_vctl_sel); + + if (cs35l36->pdata.bst_ipk) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_PEAK_CUR, + CS35L36_BST_IPK_MASK, + cs35l36->pdata.bst_ipk); + + if (cs35l36->pdata.boost_ind) { + ret = cs35l36_boost_inductor(cs35l36, cs35l36->pdata.boost_ind); + if (ret < 0) { + dev_err(cs35l36->dev, + "Boost inductor config failed(%d)\n", ret); + return ret; + } + } + + if (cs35l36->pdata.temp_warn_thld) + regmap_update_bits(cs35l36->regmap, CS35L36_DTEMP_WARN_THLD, + CS35L36_TEMP_THLD_MASK, + cs35l36->pdata.temp_warn_thld); + + if (cs35l36->pdata.irq_drv_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_DRV_SEL_MASK, + cs35l36->pdata.irq_drv_sel << + CS35L36_INT_DRV_SEL_SHIFT); + + if (cs35l36->pdata.irq_gpio_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_GPIO_SEL_MASK, + cs35l36->pdata.irq_gpio_sel << + CS35L36_INT_GPIO_SEL_SHIFT); + + /* + * Rev B0 has 2 versions + * L36 is 10V + * L37 is 12V + * If L36 we need to clamp some values for safety + * after probe has setup dt values. We want to make + * sure we dont miss any values set in probe + */ + if (cs35l36->chip_version == CS35L36_10V_L36) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_OVERVOLT_CTRL, + CS35L36_BST_OVP_THLD_MASK, + CS35L36_BST_OVP_THLD_11V); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_ANA2_TEST, + CS35L36_BST_OVP_TRIM_MASK, + CS35L36_BST_OVP_TRIM_11V << + CS35L36_BST_OVP_TRIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L36_BST_CTRL_LIM_MASK, + 1 << CS35L36_BST_CTRL_LIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + CS35L36_BST_CTRL_10V_CLAMP); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + /* + * RevA and B require the disabling of + * SYNC_GLOBAL_OVR when GLOBAL_EN = 0. + * Just turn it off from default + */ + regmap_update_bits(cs35l36->regmap, CS35L36_CTRL_OVRRIDE, + CS35L36_SYNC_GLOBAL_OVR_MASK, + 0 << CS35L36_SYNC_GLOBAL_OVR_SHIFT); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_cs35l36 = { + .probe = &cs35l36_component_probe, + .set_sysclk = cs35l36_component_set_sysclk, + .dapm_widgets = cs35l36_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l36_dapm_widgets), + .dapm_routes = cs35l36_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l36_audio_map), + .controls = cs35l36_aud_controls, + .num_controls = ARRAY_SIZE(cs35l36_aud_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct regmap_config cs35l36_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = CS35L36_PAC_PMEM_WORD1023, + .reg_defaults = cs35l36_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l36_reg), + .precious_reg = cs35l36_precious_reg, + .volatile_reg = cs35l36_volatile_reg, + .readable_reg = cs35l36_readable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l36_irq(int irq, void *data) +{ + struct cs35l36_private *cs35l36 = data; + unsigned int status[4]; + unsigned int masks[4]; + int ret = IRQ_NONE; + + /* ack the irq by reading all status registers */ + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_STATUS, status, + ARRAY_SIZE(status)); + + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_MASK, masks, + ARRAY_SIZE(masks)); + + /* Check to see if unmasked bits are active */ + if (!(status[0] & ~masks[0]) && !(status[1] & ~masks[1]) && + !(status[2] & ~masks[2]) && !(status[3] & ~masks[3])) { + return IRQ_NONE; + } + + /* + * The following interrupts require a + * protection release cycle to get the + * speaker out of Safe-Mode. + */ + if (status[2] & CS35L36_AMP_SHORT_ERR) { + dev_crit(cs35l36->dev, "Amp short error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, + CS35L36_AMP_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT3_STATUS, + CS35L36_AMP_SHORT_ERR, + CS35L36_AMP_SHORT_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_WARN) { + dev_crit(cs35l36->dev, "Over temperature warning\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, + CS35L36_TEMP_WARN_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_WARN, CS35L36_TEMP_WARN); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_ERR) { + dev_crit(cs35l36->dev, "Over temperature error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_ERR, CS35L36_TEMP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_OVP_ERR) { + dev_crit(cs35l36->dev, "VBST Over Voltage error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_OVP_ERR, CS35L36_BST_OVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_DCM_UVP_ERR) { + dev_crit(cs35l36->dev, "DCM VBST Under Voltage Error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, + CS35L36_BST_UVP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_DCM_UVP_ERR, + CS35L36_BST_DCM_UVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_SHORT_ERR) { + dev_crit(cs35l36->dev, "LBST SHORT error!\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, + CS35L36_BST_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_SHORT_ERR, + CS35L36_BST_SHORT_ERR); + ret = IRQ_HANDLED; + } + + return ret; +} + +static int cs35l36_handle_of_data(struct i2c_client *i2c_client, + struct cs35l36_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct cs35l36_vpbr_cfg *vpbr_config = &pdata->vpbr_config; + struct device_node *vpbr_node; + unsigned int val; + int ret; + + if (!np) + return 0; + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val); + if (!ret) { + if (val < 2550 || val > 12000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + pdata->bst_vctl = (((val - 2550) / 100) + 1) << 1; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-ctl-millivolt'"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "cirrus,boost-ctl-select", &val); + if (!ret) + pdata->bst_vctl_sel = val | CS35L36_VALID_PDATA; + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val); + if (!ret) { + if (val < 1600 || val > 4500) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val); + return -EINVAL; + } + + pdata->bst_ipk = (val - 1600) / 50; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-peak-milliamp'"); + return -EINVAL; + } + + pdata->multi_amp_mode = of_property_read_bool(np, + "cirrus,multi-amp-mode"); + + pdata->dcm_mode = of_property_read_bool(np, + "cirrus,dcm-mode-enable"); + + pdata->amp_pcm_inv = of_property_read_bool(np, + "cirrus,amp-pcm-inv"); + + pdata->imon_pol_inv = of_property_read_bool(np, + "cirrus,imon-pol-inv"); + + pdata->vmon_pol_inv = of_property_read_bool(np, + "cirrus,vmon-pol-inv"); + + if (of_property_read_u32(np, "cirrus,temp-warn-threshold", &val) >= 0) + pdata->temp_warn_thld = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + if (of_property_read_u32(np, "cirrus,irq-drive-select", &val) >= 0) + pdata->irq_drv_sel = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,irq-gpio-select", &val) >= 0) + pdata->irq_gpio_sel = val | CS35L36_VALID_PDATA; + + /* VPBR Config */ + vpbr_node = of_get_child_by_name(np, "cirrus,vpbr-config"); + vpbr_config->is_present = vpbr_node ? true : false; + if (vpbr_config->is_present) { + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-en", + &val) >= 0) + vpbr_config->vpbr_en = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-thld", + &val) >= 0) + vpbr_config->vpbr_thld = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-rate", + &val) >= 0) + vpbr_config->vpbr_atk_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-vol", + &val) >= 0) + vpbr_config->vpbr_atk_vol = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-max-attn", + &val) >= 0) + vpbr_config->vpbr_max_attn = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-wait", + &val) >= 0) + vpbr_config->vpbr_wait = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-rel-rate", + &val) >= 0) + vpbr_config->vpbr_rel_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-mute-en", + &val) >= 0) + vpbr_config->vpbr_mute_en = val; + } + of_node_put(vpbr_node); + + return 0; +} + +static int cs35l36_pac(struct cs35l36_private *cs35l36) +{ + int ret, count; + unsigned int val; + + if (cs35l36->rev_id != CS35L36_REV_B0) + return 0; + + /* + * Magic code for internal PAC + */ + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + usleep_range(9500, 10500); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_RESET); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS); + regmap_write(cs35l36->regmap, CS35L36_PAC_PMEM_WORD0, + CS35L36_B0_PAC_PATCH); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS_CLR); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK); + + usleep_range(9500, 10500); + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", ret); + return ret; + } + + count = 0; + while (!(val & CS35L36_MCU_CONFIG_CLR)) { + usleep_range(100, 200); + count++; + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, + &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + if (count >= 100) + return -EINVAL; + } + + regmap_write(cs35l36->regmap, CS35L36_INT4_STATUS, + CS35L36_MCU_CONFIG_CLR); + regmap_update_bits(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK, 0); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + + return 0; +} + +static void cs35l36_apply_vpbr_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct cs35l36_vpbr_cfg *vpbr_config = &pdata->vpbr_config; + + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL3, + CS35L36_VPBR_EN_MASK, + vpbr_config->vpbr_en << + CS35L36_VPBR_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_THLD_MASK, + vpbr_config->vpbr_thld << + CS35L36_VPBR_THLD_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MAX_ATTN_MASK, + vpbr_config->vpbr_max_attn << + CS35L36_VPBR_MAX_ATTN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_VOL_MASK, + vpbr_config->vpbr_atk_vol << + CS35L36_VPBR_ATK_VOL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_RATE_MASK, + vpbr_config->vpbr_atk_rate << + CS35L36_VPBR_ATK_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_WAIT_MASK, + vpbr_config->vpbr_wait << + CS35L36_VPBR_WAIT_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_REL_RATE_MASK, + vpbr_config->vpbr_rel_rate << + CS35L36_VPBR_REL_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MUTE_EN_MASK, + vpbr_config->vpbr_mute_en << + CS35L36_VPBR_MUTE_EN_SHIFT); +} + +static const struct reg_sequence cs35l36_reva0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + /* Errata Writes */ + { CS35L36_OTP_CTRL1, 0x00002060 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { CS35L36_OTP_CTRL1, 0x00002460 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { 0x00002088, 0x012A1838 }, + { 0x00003014, 0x0100EE0E }, + { 0x00003008, 0x0008184A }, + { 0x00007418, 0x509001C8 }, + { 0x00007064, 0x0929A800 }, + { 0x00002D10, 0x0002C01C }, + { 0x0000410C, 0x00000A11 }, + { 0x00006E08, 0x8B19140C }, + { 0x00006454, 0x0300000A }, + { CS35L36_AMP_NG_CTRL, 0x000020EF }, + { 0x00007E34, 0x0000000E }, + { 0x0000410C, 0x00000A11 }, + { 0x00007410, 0x20514B00 }, + /* PAC Config */ + { CS35L36_CTRL_OVRRIDE, 0x00000000 }, + { CS35L36_PAC_INT0_CTRL, 0x00860001 }, + { CS35L36_PAC_INT1_CTRL, 0x00860001 }, + { CS35L36_PAC_INT2_CTRL, 0x00860001 }, + { CS35L36_PAC_INT3_CTRL, 0x00860001 }, + { CS35L36_PAC_INT4_CTRL, 0x00860001 }, + { CS35L36_PAC_INT5_CTRL, 0x00860001 }, + { CS35L36_PAC_INT6_CTRL, 0x00860001 }, + { CS35L36_PAC_INT7_CTRL, 0x00860001 }, + { CS35L36_PAC_INT_FLUSH_CTRL, 0x000000FF }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_revb0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { 0x00007064, 0x0929A800 }, + { 0x00007850, 0x00002FA9 }, + { 0x00007854, 0x0003F1D5 }, + { 0x00007858, 0x0003F5E3 }, + { 0x0000785C, 0x00001137 }, + { 0x00007860, 0x0001A7A5 }, + { 0x00007864, 0x0002F16A }, + { 0x00007868, 0x00003E21 }, + { 0x00007848, 0x00000001 }, + { 0x00003854, 0x05180240 }, + { 0x00007418, 0x509001C8 }, + { 0x0000394C, 0x028764BD }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static int cs35l36_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l36_private *cs35l36; + struct device *dev = &i2c_client->dev; + struct cs35l36_platform_data *pdata = dev_get_platdata(dev); + struct irq_data *irq_d; + int ret, irq_pol, chip_irq_pol, i; + u32 reg_id, reg_revid, l37_id_reg; + + cs35l36 = devm_kzalloc(dev, sizeof(struct cs35l36_private), GFP_KERNEL); + if (!cs35l36) + return -ENOMEM; + + cs35l36->dev = dev; + + i2c_set_clientdata(i2c_client, cs35l36); + cs35l36->regmap = devm_regmap_init_i2c(i2c_client, &cs35l36_regmap); + if (IS_ERR(cs35l36->regmap)) { + ret = PTR_ERR(cs35l36->regmap); + dev_err(dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); + for (i = 0; i < ARRAY_SIZE(cs35l36_supplies); i++) + cs35l36->supplies[i].supply = cs35l36_supplies[i]; + + ret = devm_regulator_bulk_get(dev, cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request core supplies: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l36->pdata = *pdata; + } else { + pdata = devm_kzalloc(dev, sizeof(struct cs35l36_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + if (i2c_client->dev.of_node) { + ret = cs35l36_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + + cs35l36->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l36->num_supplies, cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs35l36->reset_gpio)) { + ret = PTR_ERR(cs35l36->reset_gpio); + cs35l36->reset_gpio = NULL; + if (ret == -EBUSY) { + dev_info(dev, "Reset line busy, assuming shared reset\n"); + } else { + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); + goto err_disable_regs; + } + } + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 1); + + usleep_range(2000, 2100); + + /* initialize amplifier */ + ret = regmap_read(cs35l36->regmap, CS35L36_SW_RESET, &reg_id); + if (ret < 0) { + dev_err(dev, "Get Device ID failed %d\n", ret); + goto err; + } + + if (reg_id != CS35L36_CHIP_ID) { + dev_err(dev, "Device ID (%X). Expected ID %X\n", reg_id, + CS35L36_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l36->regmap, CS35L36_REV_ID, &reg_revid); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed %d\n", ret); + goto err; + } + + cs35l36->rev_id = reg_revid >> 8; + + ret = regmap_read(cs35l36->regmap, CS35L36_OTP_MEM30, &l37_id_reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to read otp_id Register %d\n", + ret); + return ret; + } + + if ((l37_id_reg & CS35L36_OTP_REV_MASK) == CS35L36_OTP_REV_L37) + cs35l36->chip_version = CS35L36_12V_L37; + else + cs35l36->chip_version = CS35L36_10V_L36; + + switch (cs35l36->rev_id) { + case CS35L36_REV_A0: + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_reva0_errata_patch, + ARRAY_SIZE(cs35l36_reva0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0 errata patch %d\n", + ret); + goto err; + } + break; + case CS35L36_REV_B0: + ret = cs35l36_pac(cs35l36); + if (ret < 0) { + dev_err(dev, "Failed to Trim OTP %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_revb0_errata_patch, + ARRAY_SIZE(cs35l36_revb0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply B0 errata patch %d\n", + ret); + goto err; + } + break; + } + + if (pdata->vpbr_config.is_present) + cs35l36_apply_vpbr_config(cs35l36); + + irq_d = irq_get_irq_data(i2c_client->irq); + if (IS_ERR(irq_d)) { + dev_err(&i2c_client->dev, "Invalid IRQ: %d\n", i2c_client->irq); + ret = PTR_ERR(irq_d); + goto err; + } + + irq_pol = irqd_get_trigger_type(irq_d); + + switch (irq_pol) { + case IRQF_TRIGGER_FALLING: + case IRQF_TRIGGER_LOW: + chip_irq_pol = 0; + break; + case IRQF_TRIGGER_RISING: + case IRQF_TRIGGER_HIGH: + chip_irq_pol = 1; + break; + default: + dev_err(cs35l36->dev, "Invalid IRQ polarity: %d\n", irq_pol); + ret = -EINVAL; + goto err; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_POL_SEL_MASK, + chip_irq_pol << CS35L36_INT_POL_SEL_SHIFT); + + ret = devm_request_threaded_irq(dev, i2c_client->irq, NULL, cs35l36_irq, + IRQF_ONESHOT | irq_pol, "cs35l36", + cs35l36); + if (ret != 0) { + dev_err(dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_OUTPUT_EN_MASK, 1); + + /* Set interrupt masks for critical errors */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_DEFAULT); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_DEFAULT); + + dev_info(&i2c_client->dev, "Cirrus Logic CS35L%d, Revision: %02X\n", + cs35l36->chip_version, reg_revid >> 8); + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_cs35l36, + cs35l36_dai, + ARRAY_SIZE(cs35l36_dai)); + if (ret < 0) { + dev_err(dev, "%s: Register component failed %d\n", __func__, + ret); + goto err; + } + + return 0; + +err: + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + +err_disable_regs: + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + return ret; +} + +static int cs35l36_i2c_remove(struct i2c_client *client) +{ + struct cs35l36_private *cs35l36 = i2c_get_clientdata(client); + + /* Reset interrupt masks for device removal */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_RESET); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_RESET); + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + + return 0; +} +static const struct of_device_id cs35l36_of_match[] = { + {.compatible = "cirrus,cs35l36"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l36_of_match); + +static const struct i2c_device_id cs35l36_id[] = { + {"cs35l36", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l36_id); + +static struct i2c_driver cs35l36_i2c_driver = { + .driver = { + .name = "cs35l36", + .of_match_table = cs35l36_of_match, + }, + .id_table = cs35l36_id, + .probe = cs35l36_i2c_probe, + .remove = cs35l36_i2c_remove, +}; +module_i2c_driver(cs35l36_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L36 driver"); +MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l36.h b/sound/soc/codecs/cs35l36.h new file mode 100644 index 0000000..f6e38c6 --- /dev/null +++ b/sound/soc/codecs/cs35l36.h @@ -0,0 +1,446 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.h -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H__ +#define __CS35L36_H__ + +#include <linux/regmap.h> + +#define CS35L36_FIRSTREG 0x00000000 +#define CS35L36_LASTREG 0x00E037FC +#define CS35L36_SW_RESET 0x00000000 +#define CS35L36_SW_REV 0x00000004 +#define CS35L36_HW_REV 0x00000008 +#define CS35L36_TESTKEY_CTRL 0x00000020 +#define CS35L36_USERKEY_CTL 0x00000024 +#define CS35L36_OTP_MEM30 0x00000478 +#define CS35L36_OTP_CTRL1 0x00000500 +#define CS35L36_OTP_CTRL2 0x00000504 +#define CS35L36_OTP_CTRL3 0x00000508 +#define CS35L36_OTP_CTRL4 0x0000050C +#define CS35L36_OTP_CTRL5 0x00000510 +#define CS35L36_PAC_CTL1 0x00000C00 +#define CS35L36_PAC_CTL2 0x00000C04 +#define CS35L36_PAC_CTL3 0x00000C08 +#define CS35L36_DEVICE_ID 0x00002004 +#define CS35L36_FAB_ID 0x00002008 +#define CS35L36_REV_ID 0x0000200C +#define CS35L36_PWR_CTRL1 0x00002014 +#define CS35L36_PWR_CTRL2 0x00002018 +#define CS35L36_PWR_CTRL3 0x0000201C +#define CS35L36_CTRL_OVRRIDE 0x00002020 +#define CS35L36_AMP_OUT_MUTE 0x00002024 +#define CS35L36_OTP_TRIM_STATUS 0x00002028 +#define CS35L36_DISCH_FILT 0x0000202C +#define CS35L36_OSC_TRIM 0x00002030 +#define CS35L36_PROTECT_REL_ERR 0x00002034 +#define CS35L36_PAD_INTERFACE 0x00002400 +#define CS35L36_PLL_CLK_CTRL 0x00002C04 +#define CS35L36_GLOBAL_CLK_CTRL 0x00002C0C +#define CS35L36_ADC_CLK_CTRL 0x00002C10 +#define CS35L36_SWIRE_CLK_CTRL 0x00002C14 +#define CS35L36_SP_SCLK_CLK_CTRL 0x00002D00 +#define CS35L36_TST_FS_MON0 0x00002D10 +#define CS35L36_PLL_LOOP_PARAMS 0x00003008 +#define CS35L36_DCO_CTRL 0x00003010 +#define CS35L36_MISC_CTRL 0x00003014 +#define CS35L36_MDSYNC_EN 0x00003404 +#define CS35L36_MDSYNC_TX_ID 0x00003408 +#define CS35L36_MDSYNC_PWR_CTRL 0x0000340C +#define CS35L36_MDSYNC_DATA_TX 0x00003410 +#define CS35L36_MDSYNC_TX_STATUS 0x0000341C +#define CS35L36_MDSYNC_RX_STATUS 0x00003420 +#define CS35L36_MDSYNC_ERR_STATUS 0x00003424 +#define CS35L36_BSTCVRT_VCTRL1 0x00003800 +#define CS35L36_BSTCVRT_VCTRL2 0x00003804 +#define CS35L36_BSTCVRT_PEAK_CUR 0x00003808 +#define CS35L36_BSTCVRT_SFT_RAMP 0x0000380C +#define CS35L36_BSTCVRT_COEFF 0x00003810 +#define CS35L36_BSTCVRT_SLOPE_LBST 0x00003814 +#define CS35L36_BSTCVRT_SW_FREQ 0x00003818 +#define CS35L36_BSTCVRT_DCM_CTRL 0x0000381C +#define CS35L36_BSTCVRT_DCM_MODE_FORCE 0x00003820 +#define CS35L36_BSTCVRT_OVERVOLT_CTRL 0x00003830 +#define CS35L36_BST_TST_MANUAL 0x0000393C +#define CS35L36_BST_ANA2_TEST 0x0000394C +#define CS35L36_VPI_LIMIT_MODE 0x00003C04 +#define CS35L36_VPI_LIMIT_MINMAX 0x00003C08 +#define CS35L36_VPI_VP_THLD 0x00003C0C +#define CS35L36_VPI_TRACK_CTRL 0x00003C10 +#define CS35L36_VPI_TRIG_MODE_CTRL 0x00003C14 +#define CS35L36_VPI_TRIG_STEPS 0x00003C18 +#define CS35L36_VI_SPKMON_FILT 0x00004004 +#define CS35L36_VI_SPKMON_GAIN 0x00004008 +#define CS35L36_VI_SPKMON_IP_SEL 0x00004100 +#define CS35L36_DTEMP_WARN_THLD 0x00004220 +#define CS35L36_DTEMP_STATUS 0x00004300 +#define CS35L36_VPVBST_FS_SEL 0x00004400 +#define CS35L36_VPVBST_VP_CTRL 0x00004440 +#define CS35L36_VPVBST_VBST_CTRL 0x00004444 +#define CS35L36_ASP_TX_PIN_CTRL 0x00004800 +#define CS35L36_ASP_RATE_CTRL 0x00004804 +#define CS35L36_ASP_FORMAT 0x00004808 +#define CS35L36_ASP_FRAME_CTRL 0x00004818 +#define CS35L36_ASP_TX1_TX2_SLOT 0x0000481C +#define CS35L36_ASP_TX3_TX4_SLOT 0x00004820 +#define CS35L36_ASP_TX5_TX6_SLOT 0x00004824 +#define CS35L36_ASP_TX7_TX8_SLOT 0x00004828 +#define CS35L36_ASP_RX1_SLOT 0x0000482C +#define CS35L36_ASP_RX_TX_EN 0x0000483C +#define CS35L36_ASP_RX1_SEL 0x00004C00 +#define CS35L36_ASP_TX1_SEL 0x00004C20 +#define CS35L36_ASP_TX2_SEL 0x00004C24 +#define CS35L36_ASP_TX3_SEL 0x00004C28 +#define CS35L36_ASP_TX4_SEL 0x00004C2C +#define CS35L36_ASP_TX5_SEL 0x00004C30 +#define CS35L36_ASP_TX6_SEL 0x00004C34 +#define CS35L36_SWIRE_P1_TX1_SEL 0x00004C40 +#define CS35L36_SWIRE_P1_TX2_SEL 0x00004C44 +#define CS35L36_SWIRE_P2_TX1_SEL 0x00004C60 +#define CS35L36_SWIRE_P2_TX2_SEL 0x00004C64 +#define CS35L36_SWIRE_P2_TX3_SEL 0x00004C68 +#define CS35L36_SWIRE_DP1_FIFO_CFG 0x00005000 +#define CS35L36_SWIRE_DP2_FIFO_CFG 0x00005004 +#define CS35L36_SWIRE_DP3_FIFO_CFG 0x00005008 +#define CS35L36_SWIRE_PCM_RX_DATA 0x0000500C +#define CS35L36_SWIRE_FS_SEL 0x00005010 +#define CS35L36_SPARE_CP_BITS 0x00005C00 +#define CS35L36_AMP_DIG_VOL_CTRL 0x00006000 +#define CS35L36_VPBR_CFG 0x00006404 +#define CS35L36_VBBR_CFG 0x00006408 +#define CS35L36_VPBR_STATUS 0x0000640C +#define CS35L36_VBBR_STATUS 0x00006410 +#define CS35L36_OVERTEMP_CFG 0x00006414 +#define CS35L36_AMP_ERR_VOL 0x00006418 +#define CS35L36_CLASSH_CFG 0x00006800 +#define CS35L36_CLASSH_FET_DRV_CFG 0x00006804 +#define CS35L36_NG_CFG 0x00006808 +#define CS35L36_AMP_GAIN_CTRL 0x00006C04 +#define CS35L36_PWM_MOD_IO_CTRL 0x0000706C +#define CS35L36_PWM_MOD_STATUS 0x00007070 +#define CS35L36_DAC_MSM_CFG 0x00007400 +#define CS35L36_AMP_SLOPE_CTRL 0x00007410 +#define CS35L36_AMP_PDM_VOLUME 0x00007E04 +#define CS35L36_AMP_PDM_RATE_CTRL 0x00007E08 +#define CS35L36_PDM_CH_SEL 0x00007E10 +#define CS35L36_AMP_NG_CTRL 0x00007E14 +#define CS35L36_PDM_HIGHFILT_CTRL 0x00007E3C +#define CS35L36_INT1_STATUS 0x00D00000 +#define CS35L36_INT2_STATUS 0x00D00004 +#define CS35L36_INT3_STATUS 0x00D00008 +#define CS35L36_INT4_STATUS 0x00D0000C +#define CS35L36_INT1_RAW_STATUS 0x00D00020 +#define CS35L36_INT2_RAW_STATUS 0x00D00024 +#define CS35L36_INT3_RAW_STATUS 0x00D00028 +#define CS35L36_INT4_RAW_STATUS 0x00D0002C +#define CS35L36_INT1_MASK 0x00D00040 +#define CS35L36_INT2_MASK 0x00D00044 +#define CS35L36_INT3_MASK 0x00D00048 +#define CS35L36_INT4_MASK 0x00D0004C +#define CS35L36_INT1_EDGE_LVL_CTRL 0x00D00060 +#define CS35L36_INT3_EDGE_LVL_CTRL 0x00D00068 +#define CS35L36_PAC_INT_STATUS 0x00D00200 +#define CS35L36_PAC_INT_RAW_STATUS 0x00D00210 +#define CS35L36_PAC_INT_FLUSH_CTRL 0x00D00218 +#define CS35L36_PAC_INT0_CTRL 0x00D00220 +#define CS35L36_PAC_INT1_CTRL 0x00D00224 +#define CS35L36_PAC_INT2_CTRL 0x00D00228 +#define CS35L36_PAC_INT3_CTRL 0x00D0022C +#define CS35L36_PAC_INT4_CTRL 0x00D00230 +#define CS35L36_PAC_INT5_CTRL 0x00D00234 +#define CS35L36_PAC_INT6_CTRL 0x00D00238 +#define CS35L36_PAC_INT7_CTRL 0x00D0023C +#define CS35L36_PAC_PMEM_WORD0 0x00E02800 +#define CS35L36_PAC_PMEM_WORD1 0x00E02804 +#define CS35L36_PAC_PMEM_WORD1023 0x00E037FC + +#define CS35L36_INTPAC_REG_COUNT 25 +#define CS35L36_CHIP_ID 0x00035A36 + +#define CS35L36_INT_OUTPUT_EN_MASK 0x01 +#define CS35L36_INT_GPIO_SEL_MASK 0x02 +#define CS35L36_INT_GPIO_SEL_SHIFT 1 +#define CS35L36_INT_POL_SEL_MASK 0x04 +#define CS35L36_INT_POL_SEL_SHIFT 2 +#define CS35L36_INT_DRV_SEL_MASK 0x20 +#define CS35L36_INT_DRV_SEL_SHIFT 5 +#define CS35L36_IRQ_SRC_MASK 0x08 +#define CS35L36_IRQ_SRC_SHIFT 3 + +#define CS35L36_SCLK_MSTR_MASK 0x40 +#define CS35L36_SCLK_MSTR_SHIFT 6 +#define CS35L36_LRCLK_MSTR_MASK 0x01 +#define CS35L36_LRCLK_MSTR_SHIFT 0 +#define CS35L36_SCLK_INV_MASK 0x100 +#define CS35L36_SCLK_INV_SHIFT 8 +#define CS35L36_LRCLK_INV_MASK 0x04 +#define CS35L36_LRCLK_INV_SHIFT 2 +#define CS35L36_SCLK_FRC_MASK 0x80 +#define CS35L36_SCLK_FRC_SHIFT 7 +#define CS35L36_LRCLK_FRC_MASK 0x02 +#define CS35L36_LRCLK_FRC_SHIFT 1 + +#define CS35L36_PDM_MODE_MASK 0x01 +#define CS35L36_PDM_MODE_SHIFT 0 + +#define CS35L36_ASP_FMT_MASK 0x07 +#define CS35L36_ASP_FMT_SHIFT 0 + +#define CS35L36_ASP_RX_WIDTH_MASK 0xFF0000 +#define CS35L36_ASP_RX_WIDTH_SHIFT 16 +#define CS35L36_ASP_TX_WIDTH_MASK 0xFF +#define CS35L36_ASP_TX_WIDTH_SHIFT 0 +#define CS35L36_ASP_WIDTH_16 0x10 +#define CS35L36_ASP_WIDTH_24 0x18 +#define CS35L36_ASP_WIDTH_32 0x20 + +#define CS35L36_ASP_RX1_SLOT_MASK 0x3F +#define CS35L36_ASP_RX1_EN_MASK 0x00010000 +#define CS35L36_ASP_RX1_EN_SHIFT 16 + +#define CS35L36_ASP_TX1_SLOT_MASK 0x3F +#define CS35L36_ASP_TX2_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX2_SLOT_SHIFT 16 +#define CS35L36_ASP_TX3_SLOT_MASK 0x3F +#define CS35L36_ASP_TX4_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX4_SLOT_SHIFT 16 +#define CS35L36_ASP_TX5_SLOT_MASK 0x3F +#define CS35L36_ASP_TX6_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX6_SLOT_SHIFT 16 +#define CS35L36_ASP_TX7_SLOT_MASK 0x3F +#define CS35L36_ASP_TX8_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX8_SLOT_SHIFT 16 +#define CS35L36_ASP_TX_HIZ_MASK 0x200000 + +#define CS35L36_APS_TX_SEL_MASK 0x7F + +#define CS35L36_ASP_TX1_EN_MASK 0x01 +#define CS35L36_ASP_TX2_EN_MASK 0x02 +#define CS35L36_ASP_TX2_EN_SHIFT 1 +#define CS35L36_ASP_TX3_EN_MASK 0x04 +#define CS35L36_ASP_TX3_EN_SHIFT 2 +#define CS35L36_ASP_TX4_EN_MASK 0x08 +#define CS35L36_ASP_TX4_EN_SHIFT 3 +#define CS35L36_ASP_TX5_EN_MASK 0x10 +#define CS35L36_ASP_TX5_EN_SHIFT 4 +#define CS35L36_ASP_TX6_EN_MASK 0x20 +#define CS35L36_ASP_TX6_EN_SHIFT 5 +#define CS35L36_ASP_TX7_EN_MASK 0x40 +#define CS35L36_ASP_TX7_EN_SHIFT 6 +#define CS35L36_ASP_TX8_EN_MASK 0x80 +#define CS35L36_ASP_TX8_EN_SHIFT 7 + + +#define CS35L36_PLL_CLK_SEL_MASK 0x07 +#define CS35L36_PLL_CLK_SEL_SHIFT 0 +#define CS35L36_PLLSRC_SCLK 0 +#define CS35L36_PLLSRC_LRCLK 1 +#define CS35L36_PLLSRC_SELF 3 +#define CS35L36_PLLSRC_PDMCLK 4 +#define CS35L36_PLLSRC_MCLK 5 +#define CS35L36_PLLSRC_SWIRE 7 +#define CS35L36_REFCLK_FREQ_MASK 0x7E0 +#define CS35L36_REFCLK_FREQ_SHIFT 5 +#define CS35L36_PLL_OPENLOOP_MASK 0x800 +#define CS35L36_PLL_OPENLOOP_SHIFT 11 +#define CS35L36_PLL_REFCLK_EN_MASK 0x10 +#define CS35L36_PLL_REFCLK_EN_SHIFT 4 + + +#define CS35L36_GLOBAL_FS_MASK 0x1F +#define CS35L36_GLOBAL_FS_SHIFT 0 + +#define CS35L36_HPF_PCM_EN_MASK 0x800 +#define CS35L36_HPF_PCM_EN_SHIFT 15 +#define CS35L36_PCM_RX_SEL_MASK 0x7F +#define CS35L36_PCM_RX_SEL_SHIFT 0 + +#define CS35L36_PCM_RX_SEL_ZERO 0x00 +#define CS35L36_PCM_RX_SEL_PCM 0x08 +#define CS35L36_PCM_RX_SEL_SWIRE 0x10 +#define CS35L36_PCM_RX_SEL_DIAG 0x04 + +#define CS35L36_GLOBAL_EN_MASK 0x01 +#define CS35L36_GLOBAL_EN_SHIFT 0x00 + +#define CS35L36_AMP_PCM_INV_MASK 0x4000 +#define CS35L36_AMP_PCM_INV_SHIFT 14 + +#define CS35L36_AMP_VOL_PCM_MASK 0x3FF8 +#define CS35L36_AMP_VOL_PCM_SHIFT 3 +#define CS35L36_DIGITAL_MUTE 0x04CF + +#define CS35L36_AMP_RAMP_MASK 0x0007 +#define CS35L36_AMP_RAMP_SHIFT 0 + +#define CS35L36_AMP_MUTE_MASK 0x0010 +#define CS35L36_AMP_MUTE_SHIFT 4 + +#define CS35L36_GLOBAL_RESYNC_FS1_MASK 0x00000200 +#define CS35L36_GLOBAL_RESYNC_FS2_MASK 0x00000400 +#define CS35L36_SYNC_GLOBAL_OVR_MASK 0x00000002 +#define CS35L36_SYNC_GLOBAL_OVR_SHIFT 1 + +#define CS35L36_REFCLK_IN_MASK 0x00100000 +#define CS35L36_PLL_UNLOCK_MASK 0x00002000 + +#define CS35L36_ASP_RX_UDF_MASK 0x00000040 +#define CS35L36_ASP_RX_OVF_MASK 0x00000080 + +#define CS35L36_IMON_POL_MASK 0x02 +#define CS35L36_IMON_POL_SHIFT 1 + +#define CS35L36_VMON_POL_MASK 0x01 +#define CS35L36_VMON_POL_SHIFT 0 + +#define CS35L36_PDN_DONE 0x40 +#define CS35L36_PDN_DONE_SHIFT 6 +#define CS35L36_PUP_DONE 0x80 +#define CS35L36_PUP_DONE_SHIFT 7 +#define CS35L36_GLOBAL_EN_ASSRT 0x20 +#define CS35L36_PUP_DONE_IRQ_UNMASK 0x7F +#define CS35L36_PUP_DONE_IRQ_MASK 0xBF + +#define CS35L36_FS1_WINDOW_MASK 0x000007FF +#define CS35L36_FS2_WINDOW_MASK 0x00FFF800 +#define CS35L36_FS2_WINDOW_SHIFT 12 + +#define CS35L36_PLL_FFL_IGAIN_MASK 0x0F +#define CS35L36_PLL_IGAIN_MASK 0x3F0 +#define CS35L36_PLL_IGAIN_SHIFT 4 +#define CS35L36_PLL_IGAIN 0x04 + +#define CS35L36_BST_EN_MASK 0x30 +#define CS35L36_BST_EN 0x02 +#define CS35L36_BST_DIS_VP 0x01 +#define CS35L36_BST_DIS_EXTN 0x00 +#define CS35L36_BST_EN_SHIFT 4 +#define CS35L36_BST_MAN_IPKCOMP_MASK 0x200 +#define CS35L36_BST_MAN_IPKCOMP_SHIFT 9 + +#define CS35L36_BST_MAN_IPKCOMP_EN_MASK 0x100 +#define CS35L36_BST_MAN_IPKCOMP_EN_SHIFT 8 + +#define CS35L36_BST_IPK_MASK 0x7F +#define CS35L36_BST_OVP_THLD_MASK 0x3F +#define CS35L36_BST_OVP_THLD_11V 0x10 +#define CS35L36_BST_OVP_TRIM_MASK 0x00078000 +#define CS35L36_BST_OVP_TRIM_SHIFT 15 +#define CS35L36_BST_OVP_TRIM_11V 0x0C +#define CS35L36_BST_CTRL_LIM_MASK 0x04 +#define CS35L36_BST_CTRL_LIM_SHIFT 2 +#define CS35L36_BST_CTRL_10V_CLAMP 0x96 + +#define CS35L36_NG_AMP_EN_MASK 0x3F00 +#define CS35L36_NG_DELAY_MASK 0x70 +#define CS35L36_NG_DELAY_SHIFT 4 +#define CS35L36_AMP_ZC_SHIFT 10 +#define CS35L36_PDM_LDM_ENTER_SHIFT 3 +#define CS35L36_PDM_LDM_EXIT_SHIFT 4 + +#define CS35L36_BSTCVRT_K1_MASK 0xFF +#define CS35L36_BSTCVRT_K2_MASK 0xFF00 +#define CS35L36_BSTCVRT_K2_SHIFT 8 +#define CS35L36_BSTCVRT_SLOPE_MASK 0xFF00 +#define CS35L36_BSTCVRT_SLOPE_SHIFT 8 +#define CS35L36_BSTCVRT_CCMFREQ_MASK 0x0F +#define CS35L36_BSTCVRT_LBSTVAL_MASK 0x03 +#define CS35L35_BSTCVRT_CTL_MASK 0xFF +#define CS35L35_BSTCVRT_CTL_SEL_MASK 0x03 +#define CS35L36_DCM_AUTO_MASK 0x01 + +#define CS35L36_INT1_MASK_DEFAULT 0xF9BA7FFF +#define CS35L36_INT1_MASK_RESET 0xFFFFFFFF +#define CS35L36_INT3_MASK_DEFAULT 0xFFFFEFFF +#define CS35L36_INT3_MASK_RESET 0xFFFFFFFF + + +#define CS35L36_AMP_SHORT_ERR 0x1000 +#define CS35L36_BST_SHORT_ERR 0x40000 +#define CS35L36_TEMP_WARN 0x2000000 +#define CS35L36_TEMP_ERR 0x4000000 +#define CS35L36_BST_OVP_ERR 0x10000 +#define CS35L36_BST_DCM_UVP_ERR 0x20000 + +#define CS35L36_AMP_SHORT_ERR_RLS 0x02 +#define CS35L36_BST_SHORT_ERR_RLS 0x04 +#define CS35L36_BST_OVP_ERR_RLS 0x08 +#define CS35L36_BST_UVP_ERR_RLS 0x10 +#define CS35L36_TEMP_WARN_ERR_RLS 0x20 +#define CS35L36_TEMP_ERR_RLS 0x40 +#define CS35L36_TEMP_THLD_MASK 0x03 + +#define CS35L36_REV_B0 0xb0 +#define CS35L36_REV_A0 0xa0 +#define CS35L36_B0_PAC_PATCH 0x00DD0102 + +#define CS35L36_OTP_ECC_EN_MASK 0x400 +#define CS35L36_OTP_ECC_EN_SHIFT 10 +#define CS35L36_OTP_RUN_BOOT_MASK 0x01 +#define CS35L36_OTP_BOOT_DONE 0x2000000 +#define CS35L36_PAC_RESET_MASK 0x04 +#define CS35L36_PAC_RESET_SHIFT 2 +#define CS35L36_PAC_STALL_MASK 0x02 +#define CS35L36_PAC_STALL_SHIFT 1 +#define CS35L36_PAC_ENABLE_MASK 0x00000001 +#define CS35L36_PAC_MEM_ACCESS 0x01 +#define CS35L36_PAC_MEM_ACCESS_CLR 0 +#define CS35L36_SOFT_RESET 0x5AAA +#define CS35L36_MCU_BOOT_COMPLETE 0x02 +#define CS35L36_MCU_CONFIG_UNMASK 0x00FEFFFF +#define CS35L36_MCU_CONFIG_CLR 0x00010000 +#define CS35L36_MCU_CONFIG_MASK 0x00FFFFFF +#define CS35L36_GPIO_INT_SEL_MASK 0x0000003B +#define CS35L36_GPIO_INT_SEL_UNMASK 0x0000003A +#define CS35L36_PAC_RESET 0x00000000 +#define CS35L36_OTP_REV_MASK 0x00FF0000 +#define CS35L36_OTP_REV_L37 0x00CC0000 +#define CS35L36_12V_L37 37 +#define CS35L36_10V_L36 36 + +#define CS35L36_VPBR_EN_MASK 0x00001000 +#define CS35L36_VPBR_EN_SHIFT 12 + +#define CS35L36_VPBR_THLD_MASK 0x0000001F +#define CS35L36_VPBR_THLD_SHIFT 0 +#define CS35L36_VPBR_MAX_ATTN_MASK 0x00000F00 +#define CS35L36_VPBR_MAX_ATTN_SHIFT 8 +#define CS35L36_VPBR_ATK_VOL_MASK 0x0000F000 +#define CS35L36_VPBR_ATK_VOL_SHIFT 12 +#define CS35L36_VPBR_ATK_RATE_MASK 0x00070000 +#define CS35L36_VPBR_ATK_RATE_SHIFT 16 +#define CS35L36_VPBR_WAIT_MASK 0x00180000 +#define CS35L36_VPBR_WAIT_SHIFT 19 +#define CS35L36_VPBR_REL_RATE_MASK 0x00E00000 +#define CS35L36_VPBR_REL_RATE_SHIFT 21 +#define CS35L36_VPBR_MUTE_EN_MASK 0x01000000 +#define CS35L36_VPBR_MUTE_EN_SHIFT 24 + +#define CS35L36_OSC_FREQ_TRIM_MASK 0x070 +#define CS35L36_OSC_TRIM_DONE 0x08 + +#define CS35L36_FS1_DEFAULT_VAL 16 +#define CS35L36_FS2_DEFAULT_VAL 36 +#define CS35L36_FS_NOM_6MHZ 6000000 + +#define CS35L36_TEST_UNLOCK1 0x00005555 +#define CS35L36_TEST_UNLOCK2 0x0000AAAA +#define CS35L36_TEST_LOCK1 0x0000CCCC +#define CS35L36_TEST_LOCK2 0x00003333 + +#define CS35L36_PAC_PROG_MEM 512 + +#define CS35L36_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define CS35L36_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +extern const int cs35l36_a0_pac_patch[CS35L36_PAC_PROG_MEM]; + +#endif -- 1.9.1

2 1

[alsa-devel] Applied "ASoC: cs35l36: Add device tree documentation for CS35L36" to the asoc tree
by Mark Brown 08 Feb '19

08 Feb '19

The patch ASoC: cs35l36: Add device tree documentation for CS35L36 has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0d250bf24a68ec25b4aa2b07c200b7de4607b182 Mon Sep 17 00:00:00 2001 From: James Schulman <james.schulman(a)cirrus.com> Date: Thu, 7 Feb 2019 12:12:19 -0600 Subject: [PATCH] ASoC: cs35l36: Add device tree documentation for CS35L36 Add device tree documentation for Cirrus Logic CS35L36 speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/cs35l36.txt | 168 ++++++++++++++++++ 1 file changed, 168 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l36.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l36.txt b/Documentation/devicetree/bindings/sound/cs35l36.txt new file mode 100644 index 000000000000..912bd162b477 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l36.txt @@ -0,0 +1,168 @@ +CS35L36 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l36" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2550mV to 12000mV with + increments of 50mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1600mA and goes to a maximum of 4500mA with increments of + 50mA. + (Default) 4.50 Amps + + - cirrus,boost-ind-nanohenry : Inductor estimation LBST reference value. + Seeds the digital boost converter's inductor estimation block with the initial + inductance value to reference. + + 1000 = 1uH (Default) + 1200 = 1.2uH + +Optional properties: + - cirrus,multi-amp-mode : Boolean to determine if there are more than + one amplifier in the system. If more than one it is best to Hi-Z the ASP + port to prevent bus contention on the output signal + + - cirrus,boost-ctl-select : Boost conerter control source selection. + Selects the source of the BST_CTL target VBST voltage for the boost + converter to generate. + 0x00 - Control Port Value + 0x01 - Class H Tracking (Default) + 0x10 - MultiDevice Sync Value + + - cirrus,amp-pcm-inv : Boolean to determine Amplifier will invert incoming + PCM data + + - cirrus,imon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound IMON feedback data + + - cirrus,vmon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound VMON feedback data + + - cirrus,dcm-mode-enable : Boost converter automatic DCM Mode enable. + This enables the digital boost converter to operate in a low power + (Discontinuous Conduction) mode during low loading conditions. + + - cirrus,weak-fet-disable : Boolean : The strength of the output drivers is + reduced when operating in a Weak-FET Drive Mode and must not be used to drive + a large load. + + - cirrus,classh-wk-fet-delay : Weak-FET entry delay. Controls the delay + (in ms) before the Class H algorithm switches to the weak-FET voltage + (after the audio falls and remains below the value specified in WKFET_AMP_THLD). + + 0 = 0ms + 1 = 5ms + 2 = 10ms + 3 = 50ms + 4 = 100ms (Default) + 5 = 200ms + 6 = 500ms + 7 = 1000ms + + - cirrus,classh-weak-fet-thld-millivolt : Weak-FET amplifier drive threshold. + Configures the signal threshold at which the PWM output stage enters + weak-FET operation. The range is 50mV to 700mV in 50mV increments. + + - cirrus,temp-warn-threshold : Amplifier overtemperature warning threshold. + Configures the threshold at which the overtemperature warning condition occurs. + When the threshold is met, the overtemperature warning attenuation is applied + and the TEMP_WARN_EINT interrupt status bit is set. + If TEMP_WARN_MASK = 0, INTb is asserted. + + 0 = 105C + 1 = 115C + 2 = 125C (Default) + 3 = 135C + + - cirrus,irq-drive-select : Selects the driver type of the selected interrupt + output. + + 0 = Open-drain + 1 = Push-pull (Default) + + - cirrus,irq-gpio-select : Selects the pin to serve as the programmable + interrupt output. + + 0 = PDM_DATA / SWIRE_SD / INT (Default) + 1 = GPIO + +Optional properties for the "cirrus,vpbr-config" Sub-node + + - cirrus,vpbr-en : VBST brownout prevention enable. Configures whether the + VBST brownout prevention algorithm is enabled or disabled. + + 0 = VBST brownout prevention disabled (default) + 1 = VBST brownout prevention enabled + + See Section 7.31.1 VPBR Config for configuration options & further details + + - cirrus,vpbr-thld : Initial VPBR threshold. Configures the VP brownout + threshold voltage + + - cirrus,cirrus,vpbr-atk-rate : Attenuation attack step rate. Configures the + amount delay between consecutive volume attenuation steps when a brownout + condition is present and the VP brownout condition is in an attacking state. + + - cirrus,vpbr-atk-vol : VP brownout prevention step size. Configures the VP + brownout prevention attacking attenuation step size when operating in either + digital volume or analog gain modes. + + - cirrus,vpbr-max-attn : Maximum attenuation that the VP brownout prevention + can apply to the audio signal. + + - cirrus,vpbr-wait : Configures the delay time between a brownout condition + no longer being present and the VP brownout prevention entering an attenuation + release state. + + - cirrus,vpbr-rel-rate : Attenuation release step rate. Configures the delay + between consecutive volume attenuation release steps when a brownout condition + is not longer present and the VP brownout is in an attenuation release state. + + - cirrus,vpbr-mute-en : During the attack state, if the vpbr-max-attn value + is reached, the error condition still remains, and this bit is set, the audio + is muted. + +Example: + +cs35l36: cs35l36@40 { + compatible = "cirrus,cs35l36"; + reg = <0x40>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio0 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-ctl-millivolt = <10000>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ctl-select = <0x00>; + cirrus,weak-fet-delay = <0x04>; + cirrus,weak-fet-thld = <0x01>; + cirrus,temp-warn-threshold = <0x01>; + cirrus,multi-amp-mode; + cirrus,irq-drive-select = <0x01>; + cirrus,irq-gpio-select = <0x01>; + + cirrus,vpbr-config { + cirrus,vpbr-en = <0x00>; + cirrus,vpbr-thld = <0x05>; + cirrus,vpbr-atk-rate = <0x02>; + cirrus,vpbr-atk-vol = <0x01>; + cirrus,vpbr-max-attn = <0x09>; + cirrus,vpbr-wait = <0x01>; + cirrus,vpbr-rel-rate = <0x05>; + cirrus,vpbr-mute-en = <0x00>; + }; +}; -- 2.20.1

1 0

[alsa-devel] [PATCH 0/2] ASoC: stm32: i2s: add power management
by Olivier Moysan 08 Feb '19

08 Feb '19

Add support of system low power modes to STM32 I2S driver: Implement sleep PM suspend and resume callbacks, to restore STM32 I2S registers after low power modes. Add STM32 I2S peripheral and kernel clocks power management: - Enable/disable kernel clock on audio stream startup/shutdown. - Manage peripheral clock power through regmap services . Olivier Moysan (2): ASoC: stm32: i2s: add power management SoC: stm32: i2s: manage clock power sound/soc/stm/stm32_i2s.c | 77 +++++++++++++++++++++++++++-------------------- 1 file changed, 45 insertions(+), 32 deletions(-) -- 2.7.4

2 4

[alsa-devel] [PATCH RFC/untested] hda-intel: suspend HDA codecs and the PCI device in error cases
by Guennadi Liakhovetski 08 Feb '19

08 Feb '19

From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> If HDA probing on Intel platforms fails, all the devices, that have previously been powered up, have to be suspended again. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- I came across HDA not suspending when no codec driver has successfully probed while working on SOF. Then I decided to look how this is handled in the mainline Intel HDA code and I think something like this patch is needed, however, I don't have a set up to test it. I haven't looked at the tegra HDA implementation, possibly they need something similar. Please, comment. sound/pci/hda/hda_intel.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e784130..d3caa37 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2229,14 +2229,14 @@ static int azx_probe_continue(struct azx *chip) /* create codec instances */ err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); if (err < 0) - goto out_free; + goto out_power_down; #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { err = snd_hda_load_patch(&chip->bus, chip->fw->size, chip->fw->data); if (err < 0) - goto out_free; + goto out_power_down; #ifndef CONFIG_PM release_firmware(chip->fw); /* no longer needed */ chip->fw = NULL; @@ -2246,12 +2246,12 @@ static int azx_probe_continue(struct azx *chip) if ((probe_only[dev] & 1) == 0) { err = azx_codec_configure(chip); if (err < 0) - goto out_free; + goto out_power_down; } err = snd_card_register(chip->card); if (err < 0) - goto out_free; + goto out_power_down; setup_vga_switcheroo_runtime_pm(chip); @@ -2260,6 +2260,7 @@ static int azx_probe_continue(struct azx *chip) set_default_power_save(chip); +out_power_down: if (azx_has_pm_runtime(chip)) pm_runtime_put_autosuspend(&pci->dev); -- 1.9.3

2 5

[alsa-devel] Async ops SOC compress issue
by b_lkasam＠codeaurora.org 08 Feb '19

08 Feb '19

Hi ALSA team, Need inputs wrt one issue seen with soc-compress code. Please help check below issue seen in soc-compress file wrt handling multiple BEs with async ops ASYNC_DPCM_SND_SOC_HW_PARAMS set for given FE dailink/BE dailinks also. Issue : pcm_prepare failure when 2 BE's on same FE enabled during compress playback, seen only when both BE's have SOC_HW_PARAM defined as an async_op. Analysis: - With SOC_HW_PARAMS defined as async_op on both BE's, hw_params api for both of these BE's are scheduled for asynchronous execution. - It is observed that dpcm_be_hw_params_prepare is getting invoked twice, but for the same BE. Below are the issue logs taken for above scenario. 12-21 16:38:32.074 0 0 D SDM660 Compress1: connected new DPCM playback path SDM660 Compress1 -> INT4_MI2S_RX 12-21 16:38:32.075 0 0 D SDM660 Compress1: connected new DPCM playback path SDM660 Compress1 -> INT0_MI2S_RX 12-21 16:38:32.075 0 0 D INT0_MI2S_RX: ASoC: open playback BE INT0_MI2S_RX 12-21 16:38:32.076 0 0 D INT4_MI2S_RX: ASoC: open playback BE INT4_MI2S_RX 12-21 16:38:32.077 0 0 E soc_compr_set_params_fe: : async schedule hw_params for be INT0_MI2S_RX 12-21 16:38:32.077 0 0 E soc_compr_set_params_fe: : async schedule hw_params for be INT4_MI2S_RX 12-21 16:38:32.078 0 0 E dpcm_be_hw_params_prepare: : BE INT4_MI2S_RX -> dpcm_fe_dai_hw_params_be 12-21 16:38:32.079 0 0 E msm-dai-q6-mi2s soc: qcom,msm-dai-mi2s:qcom,msm-dai-q6-int-mi2s4: msm_dai_q6_mi2s_hw_params: dai id 11 dai_data->channels = 1 12-21 16:38:32.143 0 0 E dpcm_be_hw_params_prepare: : BE INT4_MI2S_RX -> dpcm_fe_dai_hw_params_be 12-21 16:38:32.173 0 0 E msm-dai-q6-mi2s soc: qcom,msm-dai-mi2s:qcom,msm-dai-q6-int-mi2s4: msm_dai_q6_mi2s_hw_params: ERROR dai id 11 dai_data->channels = 1 It is not clear why dpcm_be_hw_params_prepare is happening for same BE twice even though async schedule was done for both. - With async_synchronize call after scheduling of hw_param, the behavior is as expected. hw_param for both BE's are success. diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ad3cc68..b0a1995 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -535,6 +535,7 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, } } } + async_synchronize_full_domain(&async_domain); - Adding this code fixing the issue, but looks not correct way, please suggest for (i = 0; i < j; i++) { cstream->be = be_list[i]; dpcm_be_hw_params_prepare(cstream); With regards Laxminath Kasam

2 2

[alsa-devel] [PATCH] ALSA: usb-audio: Fix implicit fb endpoint setup by quirk
by manuel.rhdt＠gmail.com 07 Feb '19

07 Feb '19

The commit a60945fd08e4 ("ALSA: usb-audio: move implicit fb quirks to separate function") introduced an error in the handling of quirks for implicit feedback endpoints. This commit fixes this. If a quirk successfully sets up an implicit feedback endpoint, usb-audio no longer tries to find the implicit fb endpoint itself. Signed-off-by: Manuel Reinhardt <manuel.rhdt(a)gmail.com> --- sound/usb/pcm.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 382847154227..db114f3977e0 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -314,6 +314,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, return 0; } +/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk + * applies. Returns 1 if a quirk was found. + */ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, struct usb_device *dev, struct usb_interface_descriptor *altsd, @@ -384,7 +387,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; - return 0; + return 1; } static int set_sync_endpoint(struct snd_usb_substream *subs, @@ -423,6 +426,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, if (err < 0) return err; + /* endpoint set by quirk */ + if (err > 0) + return 0; + if (altsd->bNumEndpoints < 2) return 0; -- 2.20.1

3 2

[alsa-devel] [PATCH v3 2/2] ASoC: cs35l36: Add device tree documentation for CS35L36
by James Schulman 07 Feb '19

07 Feb '19

Add device tree documentation for Cirrus Logic CS35L36 speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- .../devicetree/bindings/sound/cs35l36.txt | 168 +++++++++++++++++++++ 1 file changed, 168 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l36.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l36.txt b/Documentation/devicetree/bindings/sound/cs35l36.txt new file mode 100644 index 0000000..912bd16 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l36.txt @@ -0,0 +1,168 @@ +CS35L36 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l36" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2550mV to 12000mV with + increments of 50mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1600mA and goes to a maximum of 4500mA with increments of + 50mA. + (Default) 4.50 Amps + + - cirrus,boost-ind-nanohenry : Inductor estimation LBST reference value. + Seeds the digital boost converter's inductor estimation block with the initial + inductance value to reference. + + 1000 = 1uH (Default) + 1200 = 1.2uH + +Optional properties: + - cirrus,multi-amp-mode : Boolean to determine if there are more than + one amplifier in the system. If more than one it is best to Hi-Z the ASP + port to prevent bus contention on the output signal + + - cirrus,boost-ctl-select : Boost conerter control source selection. + Selects the source of the BST_CTL target VBST voltage for the boost + converter to generate. + 0x00 - Control Port Value + 0x01 - Class H Tracking (Default) + 0x10 - MultiDevice Sync Value + + - cirrus,amp-pcm-inv : Boolean to determine Amplifier will invert incoming + PCM data + + - cirrus,imon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound IMON feedback data + + - cirrus,vmon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound VMON feedback data + + - cirrus,dcm-mode-enable : Boost converter automatic DCM Mode enable. + This enables the digital boost converter to operate in a low power + (Discontinuous Conduction) mode during low loading conditions. + + - cirrus,weak-fet-disable : Boolean : The strength of the output drivers is + reduced when operating in a Weak-FET Drive Mode and must not be used to drive + a large load. + + - cirrus,classh-wk-fet-delay : Weak-FET entry delay. Controls the delay + (in ms) before the Class H algorithm switches to the weak-FET voltage + (after the audio falls and remains below the value specified in WKFET_AMP_THLD). + + 0 = 0ms + 1 = 5ms + 2 = 10ms + 3 = 50ms + 4 = 100ms (Default) + 5 = 200ms + 6 = 500ms + 7 = 1000ms + + - cirrus,classh-weak-fet-thld-millivolt : Weak-FET amplifier drive threshold. + Configures the signal threshold at which the PWM output stage enters + weak-FET operation. The range is 50mV to 700mV in 50mV increments. + + - cirrus,temp-warn-threshold : Amplifier overtemperature warning threshold. + Configures the threshold at which the overtemperature warning condition occurs. + When the threshold is met, the overtemperature warning attenuation is applied + and the TEMP_WARN_EINT interrupt status bit is set. + If TEMP_WARN_MASK = 0, INTb is asserted. + + 0 = 105C + 1 = 115C + 2 = 125C (Default) + 3 = 135C + + - cirrus,irq-drive-select : Selects the driver type of the selected interrupt + output. + + 0 = Open-drain + 1 = Push-pull (Default) + + - cirrus,irq-gpio-select : Selects the pin to serve as the programmable + interrupt output. + + 0 = PDM_DATA / SWIRE_SD / INT (Default) + 1 = GPIO + +Optional properties for the "cirrus,vpbr-config" Sub-node + + - cirrus,vpbr-en : VBST brownout prevention enable. Configures whether the + VBST brownout prevention algorithm is enabled or disabled. + + 0 = VBST brownout prevention disabled (default) + 1 = VBST brownout prevention enabled + + See Section 7.31.1 VPBR Config for configuration options & further details + + - cirrus,vpbr-thld : Initial VPBR threshold. Configures the VP brownout + threshold voltage + + - cirrus,cirrus,vpbr-atk-rate : Attenuation attack step rate. Configures the + amount delay between consecutive volume attenuation steps when a brownout + condition is present and the VP brownout condition is in an attacking state. + + - cirrus,vpbr-atk-vol : VP brownout prevention step size. Configures the VP + brownout prevention attacking attenuation step size when operating in either + digital volume or analog gain modes. + + - cirrus,vpbr-max-attn : Maximum attenuation that the VP brownout prevention + can apply to the audio signal. + + - cirrus,vpbr-wait : Configures the delay time between a brownout condition + no longer being present and the VP brownout prevention entering an attenuation + release state. + + - cirrus,vpbr-rel-rate : Attenuation release step rate. Configures the delay + between consecutive volume attenuation release steps when a brownout condition + is not longer present and the VP brownout is in an attenuation release state. + + - cirrus,vpbr-mute-en : During the attack state, if the vpbr-max-attn value + is reached, the error condition still remains, and this bit is set, the audio + is muted. + +Example: + +cs35l36: cs35l36@40 { + compatible = "cirrus,cs35l36"; + reg = <0x40>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio0 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-ctl-millivolt = <10000>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ctl-select = <0x00>; + cirrus,weak-fet-delay = <0x04>; + cirrus,weak-fet-thld = <0x01>; + cirrus,temp-warn-threshold = <0x01>; + cirrus,multi-amp-mode; + cirrus,irq-drive-select = <0x01>; + cirrus,irq-gpio-select = <0x01>; + + cirrus,vpbr-config { + cirrus,vpbr-en = <0x00>; + cirrus,vpbr-thld = <0x05>; + cirrus,vpbr-atk-rate = <0x02>; + cirrus,vpbr-atk-vol = <0x01>; + cirrus,vpbr-max-attn = <0x09>; + cirrus,vpbr-wait = <0x01>; + cirrus,vpbr-rel-rate = <0x05>; + cirrus,vpbr-mute-en = <0x00>; + }; +}; -- 1.9.1

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by James Schulman 07 Feb '19

07 Feb '19

Add driver support for Cirrus Logic CS35L36 boosted speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs35l36.h | 43 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l36.c | 1960 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l36.h | 446 ++++++++++ 5 files changed, 2456 insertions(+) create mode 100644 include/sound/cs35l36.h create mode 100644 sound/soc/codecs/cs35l36.c create mode 100644 sound/soc/codecs/cs35l36.h diff --git a/include/sound/cs35l36.h b/include/sound/cs35l36.h new file mode 100644 index 0000000..f1cf121 --- /dev/null +++ b/include/sound/cs35l36.h @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * linux/sound/cs35l36.h -- Platform data for CS35L36 + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H +#define __CS35L36_H + +struct cs35l36_vpbr_cfg { + bool is_present; + bool vpbr_en; + int vpbr_thld; + int vpbr_atk_rate; + int vpbr_atk_vol; + int vpbr_max_attn; + int vpbr_wait; + int vpbr_rel_rate; + int vpbr_mute_en; +}; + +struct cs35l36_platform_data { + bool multi_amp_mode; + bool dcm_mode; + bool amp_pcm_inv; + bool imon_pol_inv; + bool vmon_pol_inv; + int boost_ind; + int bst_vctl; + int bst_vctl_sel; + int bst_ipk; + bool extern_boost; + int temp_warn_thld; + int irq_drv_sel; + int irq_gpio_sel; + struct cs35l36_vpbr_cfg vpbr_config; +}; + +#endif /* __CS35L36_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a15710c..3946222 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,6 +55,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C select SND_SOC_CS35L35 if I2C + select SND_SOC_CS35L36 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -483,6 +484,10 @@ config SND_SOC_CS35L35 tristate "Cirrus Logic CS35L35 CODEC" depends on I2C +config SND_SOC_CS35L36 + tristate "Cirrus Logic CS35L36 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d7a597..427b180 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -47,6 +47,7 @@ snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs35l35-objs := cs35l35.o +snd-soc-cs35l36-objs := cs35l36.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -318,6 +319,7 @@ obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o +obj-$(CONFIG_SND_SOC_CS35L36) += snd-soc-cs35l36.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c new file mode 100644 index 0000000..261bc0b --- /dev/null +++ b/sound/soc/codecs/cs35l36.c @@ -0,0 +1,1960 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * cs35l36.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l36.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l36.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L36_VALID_PDATA 0x80000000 + +static const char * const cs35l36_supplies[] = { + "VA", + "VP", +}; + +struct cs35l36_private { + struct device *dev; + struct cs35l36_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int clksrc; + int chip_version; + int rev_id; + int ldm_mode_sel; + struct gpio_desc *reset_gpio; +}; + +struct cs35l36_pll_config { + int freq; + int clk_cfg; + int fll_igain; +}; + +static const struct cs35l36_pll_config cs35l36_pll_sysclk[] = { + {32768, 0x00, 0x05}, + {8000, 0x01, 0x03}, + {11025, 0x02, 0x03}, + {12000, 0x03, 0x03}, + {16000, 0x04, 0x04}, + {22050, 0x05, 0x04}, + {24000, 0x06, 0x04}, + {32000, 0x07, 0x05}, + {44100, 0x08, 0x05}, + {48000, 0x09, 0x05}, + {88200, 0x0A, 0x06}, + {96000, 0x0B, 0x06}, + {128000, 0x0C, 0x07}, + {176400, 0x0D, 0x07}, + {192000, 0x0E, 0x07}, + {256000, 0x0F, 0x08}, + {352800, 0x10, 0x08}, + {384000, 0x11, 0x08}, + {512000, 0x12, 0x09}, + {705600, 0x13, 0x09}, + {750000, 0x14, 0x09}, + {768000, 0x15, 0x09}, + {1000000, 0x16, 0x0A}, + {1024000, 0x17, 0x0A}, + {1200000, 0x18, 0x0A}, + {1411200, 0x19, 0x0A}, + {1500000, 0x1A, 0x0A}, + {1536000, 0x1B, 0x0A}, + {2000000, 0x1C, 0x0A}, + {2048000, 0x1D, 0x0A}, + {2400000, 0x1E, 0x0A}, + {2822400, 0x1F, 0x0A}, + {3000000, 0x20, 0x0A}, + {3072000, 0x21, 0x0A}, + {3200000, 0x22, 0x0A}, + {4000000, 0x23, 0x0A}, + {4096000, 0x24, 0x0A}, + {4800000, 0x25, 0x0A}, + {5644800, 0x26, 0x0A}, + {6000000, 0x27, 0x0A}, + {6144000, 0x28, 0x0A}, + {6250000, 0x29, 0x08}, + {6400000, 0x2A, 0x0A}, + {6500000, 0x2B, 0x08}, + {6750000, 0x2C, 0x09}, + {7526400, 0x2D, 0x0A}, + {8000000, 0x2E, 0x0A}, + {8192000, 0x2F, 0x0A}, + {9600000, 0x30, 0x0A}, + {11289600, 0x31, 0x0A}, + {12000000, 0x32, 0x0A}, + {12288000, 0x33, 0x0A}, + {12500000, 0x34, 0x08}, + {12800000, 0x35, 0x0A}, + {13000000, 0x36, 0x0A}, + {13500000, 0x37, 0x0A}, + {19200000, 0x38, 0x0A}, + {22579200, 0x39, 0x0A}, + {24000000, 0x3A, 0x0A}, + {24576000, 0x3B, 0x0A}, + {25000000, 0x3C, 0x0A}, + {25600000, 0x3D, 0x0A}, + {26000000, 0x3E, 0x0A}, + {27000000, 0x3F, 0x0A}, +}; + +struct reg_default cs35l36_reg[] = { + {CS35L36_TESTKEY_CTRL, 0x00000000}, + {CS35L36_USERKEY_CTL, 0x00000000}, + {CS35L36_OTP_CTRL1, 0x00002460}, + {CS35L36_OTP_CTRL2, 0x00000000}, + {CS35L36_OTP_CTRL3, 0x00000000}, + {CS35L36_OTP_CTRL4, 0x00000000}, + {CS35L36_OTP_CTRL5, 0x00000000}, + {CS35L36_PAC_CTL1, 0x00000004}, + {CS35L36_PAC_CTL2, 0x00000000}, + {CS35L36_PAC_CTL3, 0x00000000}, + {CS35L36_PWR_CTRL1, 0x00000000}, + {CS35L36_PWR_CTRL2, 0x00003321}, + {CS35L36_PWR_CTRL3, 0x01000010}, + {CS35L36_CTRL_OVRRIDE, 0x00000002}, + {CS35L36_AMP_OUT_MUTE, 0x00000000}, + {CS35L36_OTP_TRIM_STATUS, 0x00000000}, + {CS35L36_DISCH_FILT, 0x00000000}, + {CS35L36_PROTECT_REL_ERR, 0x00000000}, + {CS35L36_PAD_INTERFACE, 0x00000038}, + {CS35L36_PLL_CLK_CTRL, 0x00000010}, + {CS35L36_GLOBAL_CLK_CTRL, 0x00000003}, + {CS35L36_ADC_CLK_CTRL, 0x00000000}, + {CS35L36_SWIRE_CLK_CTRL, 0x00000000}, + {CS35L36_SP_SCLK_CLK_CTRL, 0x00000000}, + {CS35L36_MDSYNC_EN, 0x00000000}, + {CS35L36_MDSYNC_TX_ID, 0x00000000}, + {CS35L36_MDSYNC_PWR_CTRL, 0x00000000}, + {CS35L36_MDSYNC_DATA_TX, 0x00000000}, + {CS35L36_MDSYNC_TX_STATUS, 0x00000002}, + {CS35L36_MDSYNC_RX_STATUS, 0x00000000}, + {CS35L36_MDSYNC_ERR_STATUS, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL1, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL2, 0x00000001}, + {CS35L36_BSTCVRT_PEAK_CUR, 0x0000004A}, + {CS35L36_BSTCVRT_SFT_RAMP, 0x00000003}, + {CS35L36_BSTCVRT_COEFF, 0x00002424}, + {CS35L36_BSTCVRT_SLOPE_LBST, 0x00005800}, + {CS35L36_BSTCVRT_SW_FREQ, 0x00010000}, + {CS35L36_BSTCVRT_DCM_CTRL, 0x00002001}, + {CS35L36_BSTCVRT_DCM_MODE_FORCE, 0x00000000}, + {CS35L36_BSTCVRT_OVERVOLT_CTRL, 0x00000130}, + {CS35L36_VPI_LIMIT_MODE, 0x00000000}, + {CS35L36_VPI_LIMIT_MINMAX, 0x00003000}, + {CS35L36_VPI_VP_THLD, 0x00101010}, + {CS35L36_VPI_TRACK_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_MODE_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_STEPS, 0x00000000}, + {CS35L36_VI_SPKMON_FILT, 0x00000003}, + {CS35L36_VI_SPKMON_GAIN, 0x00000909}, + {CS35L36_VI_SPKMON_IP_SEL, 0x00000000}, + {CS35L36_DTEMP_WARN_THLD, 0x00000002}, + {CS35L36_DTEMP_STATUS, 0x00000000}, + {CS35L36_VPVBST_FS_SEL, 0x00000001}, + {CS35L36_VPVBST_VP_CTRL, 0x000001C0}, + {CS35L36_VPVBST_VBST_CTRL, 0x000001C0}, + {CS35L36_ASP_TX_PIN_CTRL, 0x00000028}, + {CS35L36_ASP_RATE_CTRL, 0x00090000}, + {CS35L36_ASP_FORMAT, 0x00000002}, + {CS35L36_ASP_FRAME_CTRL, 0x00180018}, + {CS35L36_ASP_TX1_TX2_SLOT, 0x00010000}, + {CS35L36_ASP_TX3_TX4_SLOT, 0x00030002}, + {CS35L36_ASP_TX5_TX6_SLOT, 0x00050004}, + {CS35L36_ASP_TX7_TX8_SLOT, 0x00070006}, + {CS35L36_ASP_RX1_SLOT, 0x00000000}, + {CS35L36_ASP_RX_TX_EN, 0x00000000}, + {CS35L36_ASP_RX1_SEL, 0x00000008}, + {CS35L36_ASP_TX1_SEL, 0x00000018}, + {CS35L36_ASP_TX2_SEL, 0x00000019}, + {CS35L36_ASP_TX3_SEL, 0x00000028}, + {CS35L36_ASP_TX4_SEL, 0x00000029}, + {CS35L36_ASP_TX5_SEL, 0x00000020}, + {CS35L36_ASP_TX6_SEL, 0x00000000}, + {CS35L36_SWIRE_P1_TX1_SEL, 0x00000018}, + {CS35L36_SWIRE_P1_TX2_SEL, 0x00000019}, + {CS35L36_SWIRE_P2_TX1_SEL, 0x00000028}, + {CS35L36_SWIRE_P2_TX2_SEL, 0x00000029}, + {CS35L36_SWIRE_P2_TX3_SEL, 0x00000020}, + {CS35L36_SWIRE_DP1_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP2_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP3_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_PCM_RX_DATA, 0x00000000}, + {CS35L36_SWIRE_FS_SEL, 0x00000001}, + {CS35L36_AMP_DIG_VOL_CTRL, 0x00008000}, + {CS35L36_VPBR_CFG, 0x02AA1905}, + {CS35L36_VBBR_CFG, 0x02AA1905}, + {CS35L36_VPBR_STATUS, 0x00000000}, + {CS35L36_VBBR_STATUS, 0x00000000}, + {CS35L36_OVERTEMP_CFG, 0x00000001}, + {CS35L36_AMP_ERR_VOL, 0x00000000}, + {CS35L36_CLASSH_CFG, 0x000B0405}, + {CS35L36_CLASSH_FET_DRV_CFG, 0x00000111}, + {CS35L36_NG_CFG, 0x00000033}, + {CS35L36_AMP_GAIN_CTRL, 0x00000273}, + {CS35L36_PWM_MOD_IO_CTRL, 0x00000000}, + {CS35L36_PWM_MOD_STATUS, 0x00000000}, + {CS35L36_DAC_MSM_CFG, 0x00000000}, + {CS35L36_AMP_SLOPE_CTRL, 0x00000B00}, + {CS35L36_AMP_PDM_VOLUME, 0x00000000}, + {CS35L36_AMP_PDM_RATE_CTRL, 0x00000000}, + {CS35L36_PDM_CH_SEL, 0x00000000}, + {CS35L36_AMP_NG_CTRL, 0x0000212F}, + {CS35L36_PDM_HIGHFILT_CTRL, 0x00000000}, + {CS35L36_PAC_INT0_CTRL, 0x00000001}, + {CS35L36_PAC_INT1_CTRL, 0x00000001}, + {CS35L36_PAC_INT2_CTRL, 0x00000001}, + {CS35L36_PAC_INT3_CTRL, 0x00000001}, + {CS35L36_PAC_INT4_CTRL, 0x00000001}, + {CS35L36_PAC_INT5_CTRL, 0x00000001}, + {CS35L36_PAC_INT6_CTRL, 0x00000001}, + {CS35L36_PAC_INT7_CTRL, 0x00000001}, +}; + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_OTP_MEM30: + case CS35L36_OTP_CTRL1: + case CS35L36_OTP_CTRL2: + case CS35L36_OTP_CTRL3: + case CS35L36_OTP_CTRL4: + case CS35L36_OTP_CTRL5: + case CS35L36_PAC_CTL1: + case CS35L36_PAC_CTL2: + case CS35L36_PAC_CTL3: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_PWR_CTRL1: + case CS35L36_PWR_CTRL2: + case CS35L36_PWR_CTRL3: + case CS35L36_CTRL_OVRRIDE: + case CS35L36_AMP_OUT_MUTE: + case CS35L36_OTP_TRIM_STATUS: + case CS35L36_DISCH_FILT: + case CS35L36_PROTECT_REL_ERR: + case CS35L36_PAD_INTERFACE: + case CS35L36_PLL_CLK_CTRL: + case CS35L36_GLOBAL_CLK_CTRL: + case CS35L36_ADC_CLK_CTRL: + case CS35L36_SWIRE_CLK_CTRL: + case CS35L36_SP_SCLK_CLK_CTRL: + case CS35L36_TST_FS_MON0: + case CS35L36_MDSYNC_EN: + case CS35L36_MDSYNC_TX_ID: + case CS35L36_MDSYNC_PWR_CTRL: + case CS35L36_MDSYNC_DATA_TX: + case CS35L36_MDSYNC_TX_STATUS: + case CS35L36_MDSYNC_RX_STATUS: + case CS35L36_MDSYNC_ERR_STATUS: + case CS35L36_BSTCVRT_VCTRL1: + case CS35L36_BSTCVRT_VCTRL2: + case CS35L36_BSTCVRT_PEAK_CUR: + case CS35L36_BSTCVRT_SFT_RAMP: + case CS35L36_BSTCVRT_COEFF: + case CS35L36_BSTCVRT_SLOPE_LBST: + case CS35L36_BSTCVRT_SW_FREQ: + case CS35L36_BSTCVRT_DCM_CTRL: + case CS35L36_BSTCVRT_DCM_MODE_FORCE: + case CS35L36_BSTCVRT_OVERVOLT_CTRL: + case CS35L36_BST_TST_MANUAL: + case CS35L36_BST_ANA2_TEST: + case CS35L36_VPI_LIMIT_MODE: + case CS35L36_VPI_LIMIT_MINMAX: + case CS35L36_VPI_VP_THLD: + case CS35L36_VPI_TRACK_CTRL: + case CS35L36_VPI_TRIG_MODE_CTRL: + case CS35L36_VPI_TRIG_STEPS: + case CS35L36_VI_SPKMON_FILT: + case CS35L36_VI_SPKMON_GAIN: + case CS35L36_VI_SPKMON_IP_SEL: + case CS35L36_DTEMP_WARN_THLD: + case CS35L36_DTEMP_STATUS: + case CS35L36_VPVBST_FS_SEL: + case CS35L36_VPVBST_VP_CTRL: + case CS35L36_VPVBST_VBST_CTRL: + case CS35L36_ASP_TX_PIN_CTRL: + case CS35L36_ASP_RATE_CTRL: + case CS35L36_ASP_FORMAT: + case CS35L36_ASP_FRAME_CTRL: + case CS35L36_ASP_TX1_TX2_SLOT: + case CS35L36_ASP_TX3_TX4_SLOT: + case CS35L36_ASP_TX5_TX6_SLOT: + case CS35L36_ASP_TX7_TX8_SLOT: + case CS35L36_ASP_RX1_SLOT: + case CS35L36_ASP_RX_TX_EN: + case CS35L36_ASP_RX1_SEL: + case CS35L36_ASP_TX1_SEL: + case CS35L36_ASP_TX2_SEL: + case CS35L36_ASP_TX3_SEL: + case CS35L36_ASP_TX4_SEL: + case CS35L36_ASP_TX5_SEL: + case CS35L36_ASP_TX6_SEL: + case CS35L36_SWIRE_P1_TX1_SEL: + case CS35L36_SWIRE_P1_TX2_SEL: + case CS35L36_SWIRE_P2_TX1_SEL: + case CS35L36_SWIRE_P2_TX2_SEL: + case CS35L36_SWIRE_P2_TX3_SEL: + case CS35L36_SWIRE_DP1_FIFO_CFG: + case CS35L36_SWIRE_DP2_FIFO_CFG: + case CS35L36_SWIRE_DP3_FIFO_CFG: + case CS35L36_SWIRE_PCM_RX_DATA: + case CS35L36_SWIRE_FS_SEL: + case CS35L36_AMP_DIG_VOL_CTRL: + case CS35L36_VPBR_CFG: + case CS35L36_VBBR_CFG: + case CS35L36_VPBR_STATUS: + case CS35L36_VBBR_STATUS: + case CS35L36_OVERTEMP_CFG: + case CS35L36_AMP_ERR_VOL: + case CS35L36_CLASSH_CFG: + case CS35L36_CLASSH_FET_DRV_CFG: + case CS35L36_NG_CFG: + case CS35L36_AMP_GAIN_CTRL: + case CS35L36_PWM_MOD_IO_CTRL: + case CS35L36_PWM_MOD_STATUS: + case CS35L36_DAC_MSM_CFG: + case CS35L36_AMP_SLOPE_CTRL: + case CS35L36_AMP_PDM_VOLUME: + case CS35L36_AMP_PDM_RATE_CTRL: + case CS35L36_PDM_CH_SEL: + case CS35L36_AMP_NG_CTRL: + case CS35L36_PDM_HIGHFILT_CTRL: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + case CS35L36_PAC_INT0_CTRL: + case CS35L36_PAC_INT1_CTRL: + case CS35L36_PAC_INT2_CTRL: + case CS35L36_PAC_INT3_CTRL: + case CS35L36_PAC_INT4_CTRL: + case CS35L36_PAC_INT5_CTRL: + case CS35L36_PAC_INT6_CTRL: + case CS35L36_PAC_INT7_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +bool cs35l36_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_TST_FS_MON0: + return true; + default: + return false; + } +} + +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); + +static const char * const cs35l36_pcm_sftramp_text[] = { + "Off", ".5ms", "1ms", "2ms", "4ms", "8ms", "15ms", "30ms"}; + +static SOC_ENUM_SINGLE_DECL(pcm_sft_ramp, CS35L36_AMP_DIG_VOL_CTRL, 0, + cs35l36_pcm_sftramp_text); + +static int cs35l36_ldm_sel_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = cs35l36->ldm_mode_sel; + + return 0; +} + +static int cs35l36_ldm_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int val = (ucontrol->value.integer.value[0]) ? CS35L36_NG_AMP_EN_MASK : + 0; + + cs35l36->ldm_mode_sel = val; + + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_AMP_EN_MASK, val); + + return 0; +} + +static const struct snd_kcontrol_new cs35l36_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, + 3, 0x4D0, 0x390, dig_vol_tlv), + SOC_SINGLE_TLV("Analog PCM Volume", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, + amp_gain_tlv), + SOC_ENUM("PCM Soft Ramp", pcm_sft_ramp), + SOC_SINGLE("Amp Gain Zero-Cross Switch", CS35L36_AMP_GAIN_CTRL, + CS35L36_AMP_ZC_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Enter Ramp Switch", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_ENTER_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Exit Ramp Switch", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_EXIT_SHIFT, 1, 0), + SOC_SINGLE_BOOL_EXT("LDM Select Switch", 0, cs35l36_ldm_sel_get, + cs35l36_ldm_sel_put), +}; + +static int cs35l36_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + u32 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 1 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); + + if (WARN_ON_ONCE(reg & CS35L36_PLL_UNLOCK_MASK)) + dev_crit(cs35l36->dev, "PLL Unlocked\n"); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_PCM); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 0 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_ZERO); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 1 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 0 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_boost_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_EN << + CS35L36_BST_EN_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_DIS_VP << + CS35L36_BST_EN_SHIFT); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const char * const cs35l36_chan_text[] = { + "RX1", + "RX2", +}; + +static SOC_ENUM_SINGLE_DECL(chansel_enum, CS35L36_ASP_RX1_SLOT, 0, + cs35l36_chan_text); + +static const struct snd_kcontrol_new cs35l36_chan_mux = + SOC_DAPM_ENUM("Input Mux", chansel_enum); + +static const struct snd_kcontrol_new amp_enable_ctrl = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_SHIFT, 1, 1); + +static const struct snd_kcontrol_new boost_ctrl = + SOC_DAPM_SINGLE_VIRT("Switch", 1); + +static const char * const asp_tx_src_text[] = { + "Zero Fill", "ASPRX1", "VMON", "IMON", "ERRVOL", "VPMON", "VBSTMON" +}; + +static const unsigned int asp_tx_src_values[] = { + 0x00, 0x08, 0x18, 0x19, 0x20, 0x28, 0x29 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx1_src_enum, CS35L36_ASP_TX1_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx1_src = + SOC_DAPM_ENUM("ASPTX1SRC", asp_tx1_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx2_src_enum, CS35L36_ASP_TX2_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx2_src = + SOC_DAPM_ENUM("ASPTX2SRC", asp_tx2_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx3_src_enum, CS35L36_ASP_TX3_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx3_src = + SOC_DAPM_ENUM("ASPTX3SRC", asp_tx3_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx4_src_enum, CS35L36_ASP_TX4_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx4_src = + SOC_DAPM_ENUM("ASPTX4SRC", asp_tx4_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx5_src_enum, CS35L36_ASP_TX5_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx5_src = + SOC_DAPM_ENUM("ASPTX5SRC", asp_tx5_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx6_src_enum, CS35L36_ASP_TX6_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx6_src = + SOC_DAPM_ENUM("ASPTX6SRC", asp_tx6_src_enum); + +static const struct snd_soc_dapm_widget cs35l36_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Channel Mux", SND_SOC_NOPM, 0, 0, &cs35l36_chan_mux), + SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, CS35L36_ASP_RX_TX_EN, 16, 0), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L36_PWR_CTRL2, 0, 0, NULL, 0, + cs35l36_main_amp_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 1, &amp_enable_ctrl), + SND_SOC_DAPM_MIXER("CLASS H", CS35L36_PWR_CTRL3, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH_E("BOOST Enable", SND_SOC_NOPM, 0, 0, &boost_ctrl, + cs35l36_boost_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_AIF_OUT("ASPTX1", NULL, 0, CS35L36_ASP_RX_TX_EN, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX2", NULL, 1, CS35L36_ASP_RX_TX_EN, 1, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 2, CS35L36_ASP_RX_TX_EN, 2, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 3, CS35L36_ASP_RX_TX_EN, 3, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX5", NULL, 4, CS35L36_ASP_RX_TX_EN, 4, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX6", NULL, 5, CS35L36_ASP_RX_TX_EN, 5, 0), + + SND_SOC_DAPM_MUX("ASPTX1SRC", SND_SOC_NOPM, 0, 0, &asp_tx1_src), + SND_SOC_DAPM_MUX("ASPTX2SRC", SND_SOC_NOPM, 0, 0, &asp_tx2_src), + SND_SOC_DAPM_MUX("ASPTX3SRC", SND_SOC_NOPM, 0, 0, &asp_tx3_src), + SND_SOC_DAPM_MUX("ASPTX4SRC", SND_SOC_NOPM, 0, 0, &asp_tx4_src), + SND_SOC_DAPM_MUX("ASPTX5SRC", SND_SOC_NOPM, 0, 0, &asp_tx5_src), + SND_SOC_DAPM_MUX("ASPTX6SRC", SND_SOC_NOPM, 0, 0, &asp_tx6_src), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L36_PWR_CTRL2, 12, 0), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L36_PWR_CTRL2, 13, 0), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L36_PWR_CTRL2, 8, 0), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L36_PWR_CTRL2, 9, 0), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("VSENSE"), +}; + +static const struct snd_soc_dapm_route cs35l36_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "VSENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + + {"ASPTX1SRC", "IMON", "IMON ADC"}, + {"ASPTX1SRC", "VMON", "VMON ADC"}, + {"ASPTX1SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX1SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX2SRC", "IMON", "IMON ADC"}, + {"ASPTX2SRC", "VMON", "VMON ADC"}, + {"ASPTX2SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX2SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX3SRC", "IMON", "IMON ADC"}, + {"ASPTX3SRC", "VMON", "VMON ADC"}, + {"ASPTX3SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX3SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX4SRC", "IMON", "IMON ADC"}, + {"ASPTX4SRC", "VMON", "VMON ADC"}, + {"ASPTX4SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX4SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX5SRC", "IMON", "IMON ADC"}, + {"ASPTX5SRC", "VMON", "VMON ADC"}, + {"ASPTX5SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX5SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX6SRC", "IMON", "IMON ADC"}, + {"ASPTX6SRC", "VMON", "VMON ADC"}, + {"ASPTX6SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX6SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX1", NULL, "ASPTX1SRC"}, + {"ASPTX2", NULL, "ASPTX2SRC"}, + {"ASPTX3", NULL, "ASPTX3SRC"}, + {"ASPTX4", NULL, "ASPTX4SRC"}, + {"ASPTX5", NULL, "ASPTX5SRC"}, + {"ASPTX6", NULL, "ASPTX6SRC"}, + + {"AMP Capture", NULL, "ASPTX1"}, + {"AMP Capture", NULL, "ASPTX2"}, + {"AMP Capture", NULL, "ASPTX3"}, + {"AMP Capture", NULL, "ASPTX4"}, + {"AMP Capture", NULL, "ASPTX5"}, + {"AMP Capture", NULL, "ASPTX6"}, + + {"AMP Enable", "Switch", "AMP Playback"}, + {"SDIN", NULL, "AMP Enable"}, + {"Channel Mux", "RX1", "SDIN"}, + {"Channel Mux", "RX2", "SDIN"}, + {"BOOST Enable", "Switch", "Channel Mux"}, + {"CLASS H", NULL, "BOOST Enable"}, + {"Main AMP", NULL, "Channel Mux"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, + unsigned int fmt) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component_dai->component); + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode, clk_frc; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + slave_mode = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_MSTR_MASK, + slave_mode << CS35L36_SCLK_MSTR_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_MSTR_MASK, + slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + clk_frc = 1; + break; + case SND_SOC_DAIFMT_GATED: + clk_frc = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_FRC_MASK, clk_frc << + CS35L36_SCLK_FRC_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_FRC_MASK, clk_frc << + CS35L36_LRCLK_FRC_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + asp_fmt = 0; + break; + case SND_SOC_DAIFMT_I2S: + asp_fmt = 2; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + lrclk_fmt = 1; + sclk_fmt = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + lrclk_fmt = 0; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + lrclk_fmt = 1; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + lrclk_fmt = 0; + sclk_fmt = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_INV_MASK, + lrclk_fmt << CS35L36_LRCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_INV_MASK, + sclk_fmt << CS35L36_SCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FORMAT, + CS35L36_ASP_FMT_MASK, asp_fmt); + + return 0; +} + +struct cs35l36_global_fs_config { + int rate; + int fs_cfg; +}; + +static const struct cs35l36_global_fs_config cs35l36_fs_rates[] = { + {12000, 0x01}, + {24000, 0x02}, + {48000, 0x03}, + {96000, 0x04}, + {192000, 0x05}, + {384000, 0x06}, + {11025, 0x09}, + {22050, 0x0A}, + {44100, 0x0B}, + {88200, 0x0C}, + {176400, 0x0D}, + {8000, 0x11}, + {16000, 0x12}, + {32000, 0x13}, +}; + +static int cs35l36_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(dai->component); + unsigned int asp_width, global_fs = params_rate(params); + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_fs_rates); i++) { + if (global_fs == cs35l36_fs_rates[i].rate) + regmap_update_bits(cs35l36->regmap, + CS35L36_GLOBAL_CLK_CTRL, + CS35L36_GLOBAL_FS_MASK, + cs35l36_fs_rates[i].fs_cfg << + CS35L36_GLOBAL_FS_SHIFT); + } + + switch (params_width(params)) { + case 16: + asp_width = CS35L36_ASP_WIDTH_16; + break; + case 24: + asp_width = CS35L36_ASP_WIDTH_24; + break; + case 32: + asp_width = CS35L36_ASP_WIDTH_32; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_RX_WIDTH_MASK, + asp_width << CS35L36_ASP_RX_WIDTH_SHIFT); + } else { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_TX_WIDTH_MASK, + asp_width << CS35L36_ASP_TX_WIDTH_SHIFT); + } + + return 0; +} + +static int cs35l36_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int fs1, fs2; + + if (freq > CS35L36_FS_NOM_6MHZ) { + fs1 = CS35L36_FS1_DEFAULT_VAL; + fs2 = CS35L36_FS2_DEFAULT_VAL; + } else { + fs1 = 3 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + fs2 = 5 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + } + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_TST_FS_MON0, + CS35L36_FS1_WINDOW_MASK | CS35L36_FS2_WINDOW_MASK, + fs1 | (fs2 << CS35L36_FS2_WINDOW_SHIFT)); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + return 0; +} + +static const struct cs35l36_pll_config *cs35l36_get_clk_config( + struct cs35l36_private *cs35l36, int freq) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_pll_sysclk); i++) { + if (cs35l36_pll_sysclk[i].freq == freq) + return &cs35l36_pll_sysclk[i]; + } + + return NULL; +} + +static const unsigned int cs35l36_src_rates[] = { + 8000, 12000, 11025, 16000, 22050, 24000, 32000, + 44100, 48000, 88200, 96000, 176400, 192000, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { + .count = ARRAY_SIZE(cs35l36_src_rates), + .list = cs35l36_src_rates, +}; + +static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l36_ops = { + .startup = cs35l36_pcm_startup, + .set_fmt = cs35l36_set_dai_fmt, + .hw_params = cs35l36_pcm_hw_params, + .set_sysclk = cs35l36_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l36_dai[] = { + { + .name = "cs35l36-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_RX_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_TX_FORMATS, + }, + .ops = &cs35l36_ops, + .symmetric_rates = 1, + }, +}; + +static int cs35l36_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + const struct cs35l36_pll_config *clk_cfg; + int prev_clksrc; + bool pdm_switch; + + prev_clksrc = cs35l36->clksrc; + + switch (clk_id) { + case 0: + cs35l36->clksrc = CS35L36_PLLSRC_SCLK; + break; + case 1: + cs35l36->clksrc = CS35L36_PLLSRC_LRCLK; + break; + case 2: + cs35l36->clksrc = CS35L36_PLLSRC_PDMCLK; + break; + case 3: + cs35l36->clksrc = CS35L36_PLLSRC_SELF; + break; + case 4: + cs35l36->clksrc = CS35L36_PLLSRC_MCLK; + break; + default: + return -EINVAL; + } + + clk_cfg = cs35l36_get_clk_config(cs35l36, freq); + if (clk_cfg == NULL) { + dev_err(component->dev, "Invalid CLK Config Freq: %d\n", freq); + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 1 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_REFCLK_FREQ_MASK, + clk_cfg->clk_cfg << CS35L36_REFCLK_FREQ_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 0 << CS35L36_PLL_REFCLK_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_CLK_SEL_MASK, + cs35l36->clksrc); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 0 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 1 << CS35L36_PLL_REFCLK_EN_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_write(cs35l36->regmap, CS35L36_DCO_CTRL, 0x00036DA8); + regmap_write(cs35l36->regmap, CS35L36_MISC_CTRL, 0x0100EE0E); + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_IGAIN_MASK, + CS35L36_PLL_IGAIN << + CS35L36_PLL_IGAIN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_FFL_IGAIN_MASK, + clk_cfg->fll_igain); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->clksrc == CS35L36_PLLSRC_PDMCLK) { + pdm_switch = cs35l36->ldm_mode_sel && + (prev_clksrc != CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 1 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } else { + pdm_switch = cs35l36->ldm_mode_sel && + (prev_clksrc == CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 0 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + + return 0; +} + +static int cs35l36_boost_inductor(struct cs35l36_private *cs35l36, int inductor) +{ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K1_MASK, 0x3C); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K2_MASK, + 0x3C << CS35L36_BSTCVRT_K2_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SW_FREQ, + CS35L36_BSTCVRT_CCMFREQ_MASK, 0x00); + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x75 << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x00); + break; + case 1200: /* 1.2 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x6B << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x01); + break; + default: + dev_err(cs35l36->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_component_probe(struct snd_soc_component *component) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + if ((cs35l36->rev_id == CS35L36_REV_A0) && cs35l36->pdata.dcm_mode) { + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_DCM_CTRL, + CS35L36_DCM_AUTO_MASK, + CS35L36_DCM_AUTO_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_MASK, + 0 << CS35L36_BST_MAN_IPKCOMP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_EN_MASK, + CS35L36_BST_MAN_IPKCOMP_EN_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->pdata.amp_pcm_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_DIG_VOL_CTRL, + CS35L36_AMP_PCM_INV_MASK, + CS35L36_AMP_PCM_INV_MASK); + + if (cs35l36->pdata.multi_amp_mode) + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_ASP_TX_HIZ_MASK, + CS35L36_ASP_TX_HIZ_MASK); + + if (cs35l36->pdata.imon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_IMON_POL_MASK, 0); + + if (cs35l36->pdata.vmon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_VMON_POL_MASK, 0); + + if (cs35l36->pdata.bst_vctl) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + cs35l36->pdata.bst_vctl); + + if (cs35l36->pdata.bst_vctl_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L35_BSTCVRT_CTL_SEL_MASK, + cs35l36->pdata.bst_vctl_sel); + + if (cs35l36->pdata.bst_ipk) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_PEAK_CUR, + CS35L36_BST_IPK_MASK, + cs35l36->pdata.bst_ipk); + + if (cs35l36->pdata.boost_ind) { + ret = cs35l36_boost_inductor(cs35l36, cs35l36->pdata.boost_ind); + if (ret < 0) { + dev_err(cs35l36->dev, + "Boost inductor config failed(%d)\n", ret); + return ret; + } + } + + if (cs35l36->pdata.temp_warn_thld) + regmap_update_bits(cs35l36->regmap, CS35L36_DTEMP_WARN_THLD, + CS35L36_TEMP_THLD_MASK, + cs35l36->pdata.temp_warn_thld); + + if (cs35l36->pdata.irq_drv_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_DRV_SEL_MASK, + cs35l36->pdata.irq_drv_sel << + CS35L36_INT_DRV_SEL_SHIFT); + + if (cs35l36->pdata.irq_gpio_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_GPIO_SEL_MASK, + cs35l36->pdata.irq_gpio_sel << + CS35L36_INT_GPIO_SEL_SHIFT); + + /* + * Rev B0 has 2 versions + * L36 is 10V + * L37 is 12V + * If L36 we need to clamp some values for safety + * after probe has setup dt values. We want to make + * sure we dont miss any values set in probe + */ + if (cs35l36->chip_version == CS35L36_10V_L36) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_OVERVOLT_CTRL, + CS35L36_BST_OVP_THLD_MASK, + CS35L36_BST_OVP_THLD_11V); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_ANA2_TEST, + CS35L36_BST_OVP_TRIM_MASK, + CS35L36_BST_OVP_TRIM_11V << + CS35L36_BST_OVP_TRIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L36_BST_CTRL_LIM_MASK, + 1 << CS35L36_BST_CTRL_LIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + CS35L36_BST_CTRL_10V_CLAMP); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + /* + * RevA and B require the disabling of + * SYNC_GLOBAL_OVR when GLOBAL_EN = 0. + * Just turn it off from default + */ + regmap_update_bits(cs35l36->regmap, CS35L36_CTRL_OVRRIDE, + CS35L36_SYNC_GLOBAL_OVR_MASK, + 0 << CS35L36_SYNC_GLOBAL_OVR_SHIFT); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_cs35l36 = { + .probe = &cs35l36_component_probe, + .set_sysclk = cs35l36_component_set_sysclk, + .dapm_widgets = cs35l36_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l36_dapm_widgets), + .dapm_routes = cs35l36_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l36_audio_map), + .controls = cs35l36_aud_controls, + .num_controls = ARRAY_SIZE(cs35l36_aud_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct regmap_config cs35l36_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = CS35L36_PAC_PMEM_WORD1023, + .reg_defaults = cs35l36_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l36_reg), + .precious_reg = cs35l36_precious_reg, + .volatile_reg = cs35l36_volatile_reg, + .readable_reg = cs35l36_readable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l36_irq(int irq, void *data) +{ + struct cs35l36_private *cs35l36 = data; + unsigned int status[4]; + unsigned int masks[4]; + int ret = IRQ_NONE; + + /* ack the irq by reading all status registers */ + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_STATUS, status, + ARRAY_SIZE(status)); + + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_MASK, masks, + ARRAY_SIZE(masks)); + + /* Check to see if unmasked bits are active */ + if (!(status[0] & ~masks[0]) && !(status[1] & ~masks[1]) && + !(status[2] & ~masks[2]) && !(status[3] & ~masks[3])) { + return IRQ_NONE; + } + + /* + * The following interrupts require a + * protection release cycle to get the + * speaker out of Safe-Mode. + */ + if (status[2] & CS35L36_AMP_SHORT_ERR) { + dev_crit(cs35l36->dev, "Amp short error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, + CS35L36_AMP_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT3_STATUS, + CS35L36_AMP_SHORT_ERR, + CS35L36_AMP_SHORT_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_WARN) { + dev_crit(cs35l36->dev, "Over temperature warning\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, + CS35L36_TEMP_WARN_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_WARN, CS35L36_TEMP_WARN); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_ERR) { + dev_crit(cs35l36->dev, "Over temperature error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_ERR, CS35L36_TEMP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_OVP_ERR) { + dev_crit(cs35l36->dev, "VBST Over Voltage error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_OVP_ERR, CS35L36_BST_OVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_DCM_UVP_ERR) { + dev_crit(cs35l36->dev, "DCM VBST Under Voltage Error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, + CS35L36_BST_UVP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_DCM_UVP_ERR, + CS35L36_BST_DCM_UVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_SHORT_ERR) { + dev_crit(cs35l36->dev, "LBST SHORT error!\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, + CS35L36_BST_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_SHORT_ERR, + CS35L36_BST_SHORT_ERR); + ret = IRQ_HANDLED; + } + + return ret; +} + +static int cs35l36_handle_of_data(struct i2c_client *i2c_client, + struct cs35l36_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct cs35l36_vpbr_cfg *vpbr_config = &pdata->vpbr_config; + struct device_node *vpbr_node; + unsigned int val; + int ret; + + if (!np) + return 0; + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val); + if (!ret) { + if (val < 2550 || val > 12000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + pdata->bst_vctl = (((val - 2550) / 100) + 1) << 1; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-ctl-millivolt'"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "cirrus,boost-ctl-select", &val); + if (!ret) + pdata->bst_vctl_sel = val | CS35L36_VALID_PDATA; + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val); + if (!ret) { + if (val < 1600 || val > 4500) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val); + return -EINVAL; + } + + pdata->bst_ipk = (val - 1600) / 50; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-peak-milliamp'"); + return -EINVAL; + } + + pdata->multi_amp_mode = of_property_read_bool(np, + "cirrus,multi-amp-mode"); + + pdata->dcm_mode = of_property_read_bool(np, + "cirrus,dcm-mode-enable"); + + pdata->amp_pcm_inv = of_property_read_bool(np, + "cirrus,amp-pcm-inv"); + + pdata->imon_pol_inv = of_property_read_bool(np, + "cirrus,imon-pol-inv"); + + pdata->vmon_pol_inv = of_property_read_bool(np, + "cirrus,vmon-pol-inv"); + + if (of_property_read_u32(np, "cirrus,temp-warn-threshold", &val) >= 0) + pdata->temp_warn_thld = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + if (of_property_read_u32(np, "cirrus,irq-drive-select", &val) >= 0) + pdata->irq_drv_sel = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,irq-gpio-select", &val) >= 0) + pdata->irq_gpio_sel = val | CS35L36_VALID_PDATA; + + /* VPBR Config */ + vpbr_node = of_get_child_by_name(np, "cirrus,vpbr-config"); + vpbr_config->is_present = vpbr_node ? true : false; + if (vpbr_config->is_present) { + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-en", + &val) >= 0) + vpbr_config->vpbr_en = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-thld", + &val) >= 0) + vpbr_config->vpbr_thld = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-rate", + &val) >= 0) + vpbr_config->vpbr_atk_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-vol", + &val) >= 0) + vpbr_config->vpbr_atk_vol = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-max-attn", + &val) >= 0) + vpbr_config->vpbr_max_attn = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-wait", + &val) >= 0) + vpbr_config->vpbr_wait = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-rel-rate", + &val) >= 0) + vpbr_config->vpbr_rel_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-mute-en", + &val) >= 0) + vpbr_config->vpbr_mute_en = val; + } + of_node_put(vpbr_node); + + return 0; +} + +static int cs35l36_pac(struct cs35l36_private *cs35l36) +{ + int ret, count; + unsigned int val; + + if (cs35l36->rev_id != CS35L36_REV_B0) + return 0; + + /* + * Magic code for internal PAC + */ + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + usleep_range(9500, 10500); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_RESET); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS); + regmap_write(cs35l36->regmap, CS35L36_PAC_PMEM_WORD0, + CS35L36_B0_PAC_PATCH); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS_CLR); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK); + + usleep_range(9500, 10500); + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", ret); + return ret; + } + + count = 0; + while (!(val & CS35L36_MCU_CONFIG_CLR)) { + usleep_range(100, 200); + count++; + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, + &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + if (count >= 100) + return -EINVAL; + } + + regmap_write(cs35l36->regmap, CS35L36_INT4_STATUS, + CS35L36_MCU_CONFIG_CLR); + regmap_update_bits(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK, 0); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + + return 0; +} + +static void cs35l36_apply_vpbr_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct cs35l36_vpbr_cfg *vpbr_config = &pdata->vpbr_config; + + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL3, + CS35L36_VPBR_EN_MASK, + vpbr_config->vpbr_en << + CS35L36_VPBR_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_THLD_MASK, + vpbr_config->vpbr_thld << + CS35L36_VPBR_THLD_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MAX_ATTN_MASK, + vpbr_config->vpbr_max_attn << + CS35L36_VPBR_MAX_ATTN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_VOL_MASK, + vpbr_config->vpbr_atk_vol << + CS35L36_VPBR_ATK_VOL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_RATE_MASK, + vpbr_config->vpbr_atk_rate << + CS35L36_VPBR_ATK_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_WAIT_MASK, + vpbr_config->vpbr_wait << + CS35L36_VPBR_WAIT_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_REL_RATE_MASK, + vpbr_config->vpbr_rel_rate << + CS35L36_VPBR_REL_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MUTE_EN_MASK, + vpbr_config->vpbr_mute_en << + CS35L36_VPBR_MUTE_EN_SHIFT); +} + +static const struct reg_sequence cs35l36_reva0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + /* Errata Writes */ + { CS35L36_OTP_CTRL1, 0x00002060 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { CS35L36_OTP_CTRL1, 0x00002460 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { 0x00002088, 0x012A1838 }, + { 0x00003014, 0x0100EE0E }, + { 0x00003008, 0x0008184A }, + { 0x00007418, 0x509001C8 }, + { 0x00007064, 0x0929A800 }, + { 0x00002D10, 0x0002C01C }, + { 0x0000410C, 0x00000A11 }, + { 0x00006E08, 0x8B19140C }, + { 0x00006454, 0x0300000A }, + { CS35L36_AMP_NG_CTRL, 0x000020EF }, + { 0x00007E34, 0x0000000E }, + { 0x0000410C, 0x00000A11 }, + { 0x00007410, 0x20514B00 }, + /* PAC Config */ + { CS35L36_CTRL_OVRRIDE, 0x00000000 }, + { CS35L36_PAC_INT0_CTRL, 0x00860001 }, + { CS35L36_PAC_INT1_CTRL, 0x00860001 }, + { CS35L36_PAC_INT2_CTRL, 0x00860001 }, + { CS35L36_PAC_INT3_CTRL, 0x00860001 }, + { CS35L36_PAC_INT4_CTRL, 0x00860001 }, + { CS35L36_PAC_INT5_CTRL, 0x00860001 }, + { CS35L36_PAC_INT6_CTRL, 0x00860001 }, + { CS35L36_PAC_INT7_CTRL, 0x00860001 }, + { CS35L36_PAC_INT_FLUSH_CTRL, 0x000000FF }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_revb0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { 0x00007064, 0x0929A800 }, + { 0x00007850, 0x00002FA9 }, + { 0x00007854, 0x0003F1D5 }, + { 0x00007858, 0x0003F5E3 }, + { 0x0000785C, 0x00001137 }, + { 0x00007860, 0x0001A7A5 }, + { 0x00007864, 0x0002F16A }, + { 0x00007868, 0x00003E21 }, + { 0x00007848, 0x00000001 }, + { 0x00003854, 0x05180240 }, + { 0x00007418, 0x509001C8 }, + { 0x0000394C, 0x028764BD }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static int cs35l36_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l36_private *cs35l36; + struct device *dev = &i2c_client->dev; + struct cs35l36_platform_data *pdata = dev_get_platdata(dev); + struct irq_data *irq_d; + int ret, irq_pol, chip_irq_pol, i; + u32 reg_id, reg_revid, l37_id_reg; + + cs35l36 = devm_kzalloc(dev, sizeof(struct cs35l36_private), GFP_KERNEL); + if (!cs35l36) + return -ENOMEM; + + cs35l36->dev = dev; + + i2c_set_clientdata(i2c_client, cs35l36); + cs35l36->regmap = devm_regmap_init_i2c(i2c_client, &cs35l36_regmap); + if (IS_ERR(cs35l36->regmap)) { + ret = PTR_ERR(cs35l36->regmap); + dev_err(dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); + for (i = 0; i < ARRAY_SIZE(cs35l36_supplies); i++) + cs35l36->supplies[i].supply = cs35l36_supplies[i]; + + ret = devm_regulator_bulk_get(dev, cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request core supplies: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l36->pdata = *pdata; + } else { + pdata = devm_kzalloc(dev, sizeof(struct cs35l36_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + if (i2c_client->dev.of_node) { + ret = cs35l36_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + + cs35l36->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l36->num_supplies, cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs35l36->reset_gpio)) { + ret = PTR_ERR(cs35l36->reset_gpio); + cs35l36->reset_gpio = NULL; + if (ret == -EBUSY) { + dev_info(dev, "Reset line busy, assuming shared reset\n"); + } else { + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); + goto err_disable_regs; + } + } + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 1); + + usleep_range(2000, 2100); + + /* initialize amplifier */ + ret = regmap_read(cs35l36->regmap, CS35L36_SW_RESET, &reg_id); + if (ret < 0) { + dev_err(dev, "Get Device ID failed %d\n", ret); + goto err; + } + + if (reg_id != CS35L36_CHIP_ID) { + dev_err(dev, "Device ID (%X). Expected ID %X\n", reg_id, + CS35L36_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l36->regmap, CS35L36_REV_ID, &reg_revid); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed %d\n", ret); + goto err; + } + + cs35l36->rev_id = reg_revid >> 8; + + ret = regmap_read(cs35l36->regmap, CS35L36_OTP_MEM30, &l37_id_reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to read otp_id Register %d\n", + ret); + return ret; + } + + if ((l37_id_reg & CS35L36_OTP_REV_MASK) == CS35L36_OTP_REV_L37) + cs35l36->chip_version = CS35L36_12V_L37; + else + cs35l36->chip_version = CS35L36_10V_L36; + + switch (cs35l36->rev_id) { + case CS35L36_REV_A0: + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_reva0_errata_patch, + ARRAY_SIZE(cs35l36_reva0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0 errata patch %d\n", + ret); + goto err; + } + break; + case CS35L36_REV_B0: + ret = cs35l36_pac(cs35l36); + if (ret < 0) { + dev_err(dev, "Failed to Trim OTP %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_revb0_errata_patch, + ARRAY_SIZE(cs35l36_revb0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply B0 errata patch %d\n", + ret); + goto err; + } + break; + } + + if (pdata->vpbr_config.is_present) + cs35l36_apply_vpbr_config(cs35l36); + + irq_d = irq_get_irq_data(i2c_client->irq); + if (IS_ERR(irq_d)) { + dev_err(&i2c_client->dev, "Invalid IRQ: %d\n", i2c_client->irq); + ret = PTR_ERR(irq_d); + goto err; + } + + irq_pol = irqd_get_trigger_type(irq_d); + + switch (irq_pol) { + case IRQF_TRIGGER_FALLING: + case IRQF_TRIGGER_LOW: + chip_irq_pol = 0; + break; + case IRQF_TRIGGER_RISING: + case IRQF_TRIGGER_HIGH: + chip_irq_pol = 1; + break; + default: + dev_err(cs35l36->dev, "Invalid IRQ polarity: %d\n", irq_pol); + ret = -EINVAL; + goto err; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_POL_SEL_MASK, + chip_irq_pol << CS35L36_INT_POL_SEL_SHIFT); + + ret = devm_request_threaded_irq(dev, i2c_client->irq, NULL, cs35l36_irq, + IRQF_ONESHOT | irq_pol, "cs35l36", + cs35l36); + if (ret != 0) { + dev_err(dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_OUTPUT_EN_MASK, 1); + + /* Set interrupt masks for critical errors */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_DEFAULT); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_DEFAULT); + + dev_info(&i2c_client->dev, "Cirrus Logic CS35L%d, Revision: %02X\n", + cs35l36->chip_version, reg_revid >> 8); + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_cs35l36, + cs35l36_dai, + ARRAY_SIZE(cs35l36_dai)); + if (ret < 0) { + dev_err(dev, "%s: Register component failed %d\n", __func__, + ret); + goto err; + } + + return 0; + +err: + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + +err_disable_regs: + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + return ret; +} + +static int cs35l36_i2c_remove(struct i2c_client *client) +{ + struct cs35l36_private *cs35l36 = i2c_get_clientdata(client); + + /* Reset interrupt masks for device removal */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_RESET); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_RESET); + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + + return 0; +} +static const struct of_device_id cs35l36_of_match[] = { + {.compatible = "cirrus,cs35l36"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l36_of_match); + +static const struct i2c_device_id cs35l36_id[] = { + {"cs35l36", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l36_id); + +static struct i2c_driver cs35l36_i2c_driver = { + .driver = { + .name = "cs35l36", + .of_match_table = cs35l36_of_match, + }, + .id_table = cs35l36_id, + .probe = cs35l36_i2c_probe, + .remove = cs35l36_i2c_remove, +}; +module_i2c_driver(cs35l36_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L36 driver"); +MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l36.h b/sound/soc/codecs/cs35l36.h new file mode 100644 index 0000000..2a31319 --- /dev/null +++ b/sound/soc/codecs/cs35l36.h @@ -0,0 +1,446 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * cs35l36.h -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H__ +#define __CS35L36_H__ + +#include <linux/regmap.h> + +#define CS35L36_FIRSTREG 0x00000000 +#define CS35L36_LASTREG 0x00E037FC +#define CS35L36_SW_RESET 0x00000000 +#define CS35L36_SW_REV 0x00000004 +#define CS35L36_HW_REV 0x00000008 +#define CS35L36_TESTKEY_CTRL 0x00000020 +#define CS35L36_USERKEY_CTL 0x00000024 +#define CS35L36_OTP_MEM30 0x00000478 +#define CS35L36_OTP_CTRL1 0x00000500 +#define CS35L36_OTP_CTRL2 0x00000504 +#define CS35L36_OTP_CTRL3 0x00000508 +#define CS35L36_OTP_CTRL4 0x0000050C +#define CS35L36_OTP_CTRL5 0x00000510 +#define CS35L36_PAC_CTL1 0x00000C00 +#define CS35L36_PAC_CTL2 0x00000C04 +#define CS35L36_PAC_CTL3 0x00000C08 +#define CS35L36_DEVICE_ID 0x00002004 +#define CS35L36_FAB_ID 0x00002008 +#define CS35L36_REV_ID 0x0000200C +#define CS35L36_PWR_CTRL1 0x00002014 +#define CS35L36_PWR_CTRL2 0x00002018 +#define CS35L36_PWR_CTRL3 0x0000201C +#define CS35L36_CTRL_OVRRIDE 0x00002020 +#define CS35L36_AMP_OUT_MUTE 0x00002024 +#define CS35L36_OTP_TRIM_STATUS 0x00002028 +#define CS35L36_DISCH_FILT 0x0000202C +#define CS35L36_OSC_TRIM 0x00002030 +#define CS35L36_PROTECT_REL_ERR 0x00002034 +#define CS35L36_PAD_INTERFACE 0x00002400 +#define CS35L36_PLL_CLK_CTRL 0x00002C04 +#define CS35L36_GLOBAL_CLK_CTRL 0x00002C0C +#define CS35L36_ADC_CLK_CTRL 0x00002C10 +#define CS35L36_SWIRE_CLK_CTRL 0x00002C14 +#define CS35L36_SP_SCLK_CLK_CTRL 0x00002D00 +#define CS35L36_TST_FS_MON0 0x00002D10 +#define CS35L36_PLL_LOOP_PARAMS 0x00003008 +#define CS35L36_DCO_CTRL 0x00003010 +#define CS35L36_MISC_CTRL 0x00003014 +#define CS35L36_MDSYNC_EN 0x00003404 +#define CS35L36_MDSYNC_TX_ID 0x00003408 +#define CS35L36_MDSYNC_PWR_CTRL 0x0000340C +#define CS35L36_MDSYNC_DATA_TX 0x00003410 +#define CS35L36_MDSYNC_TX_STATUS 0x0000341C +#define CS35L36_MDSYNC_RX_STATUS 0x00003420 +#define CS35L36_MDSYNC_ERR_STATUS 0x00003424 +#define CS35L36_BSTCVRT_VCTRL1 0x00003800 +#define CS35L36_BSTCVRT_VCTRL2 0x00003804 +#define CS35L36_BSTCVRT_PEAK_CUR 0x00003808 +#define CS35L36_BSTCVRT_SFT_RAMP 0x0000380C +#define CS35L36_BSTCVRT_COEFF 0x00003810 +#define CS35L36_BSTCVRT_SLOPE_LBST 0x00003814 +#define CS35L36_BSTCVRT_SW_FREQ 0x00003818 +#define CS35L36_BSTCVRT_DCM_CTRL 0x0000381C +#define CS35L36_BSTCVRT_DCM_MODE_FORCE 0x00003820 +#define CS35L36_BSTCVRT_OVERVOLT_CTRL 0x00003830 +#define CS35L36_BST_TST_MANUAL 0x0000393C +#define CS35L36_BST_ANA2_TEST 0x0000394C +#define CS35L36_VPI_LIMIT_MODE 0x00003C04 +#define CS35L36_VPI_LIMIT_MINMAX 0x00003C08 +#define CS35L36_VPI_VP_THLD 0x00003C0C +#define CS35L36_VPI_TRACK_CTRL 0x00003C10 +#define CS35L36_VPI_TRIG_MODE_CTRL 0x00003C14 +#define CS35L36_VPI_TRIG_STEPS 0x00003C18 +#define CS35L36_VI_SPKMON_FILT 0x00004004 +#define CS35L36_VI_SPKMON_GAIN 0x00004008 +#define CS35L36_VI_SPKMON_IP_SEL 0x00004100 +#define CS35L36_DTEMP_WARN_THLD 0x00004220 +#define CS35L36_DTEMP_STATUS 0x00004300 +#define CS35L36_VPVBST_FS_SEL 0x00004400 +#define CS35L36_VPVBST_VP_CTRL 0x00004440 +#define CS35L36_VPVBST_VBST_CTRL 0x00004444 +#define CS35L36_ASP_TX_PIN_CTRL 0x00004800 +#define CS35L36_ASP_RATE_CTRL 0x00004804 +#define CS35L36_ASP_FORMAT 0x00004808 +#define CS35L36_ASP_FRAME_CTRL 0x00004818 +#define CS35L36_ASP_TX1_TX2_SLOT 0x0000481C +#define CS35L36_ASP_TX3_TX4_SLOT 0x00004820 +#define CS35L36_ASP_TX5_TX6_SLOT 0x00004824 +#define CS35L36_ASP_TX7_TX8_SLOT 0x00004828 +#define CS35L36_ASP_RX1_SLOT 0x0000482C +#define CS35L36_ASP_RX_TX_EN 0x0000483C +#define CS35L36_ASP_RX1_SEL 0x00004C00 +#define CS35L36_ASP_TX1_SEL 0x00004C20 +#define CS35L36_ASP_TX2_SEL 0x00004C24 +#define CS35L36_ASP_TX3_SEL 0x00004C28 +#define CS35L36_ASP_TX4_SEL 0x00004C2C +#define CS35L36_ASP_TX5_SEL 0x00004C30 +#define CS35L36_ASP_TX6_SEL 0x00004C34 +#define CS35L36_SWIRE_P1_TX1_SEL 0x00004C40 +#define CS35L36_SWIRE_P1_TX2_SEL 0x00004C44 +#define CS35L36_SWIRE_P2_TX1_SEL 0x00004C60 +#define CS35L36_SWIRE_P2_TX2_SEL 0x00004C64 +#define CS35L36_SWIRE_P2_TX3_SEL 0x00004C68 +#define CS35L36_SWIRE_DP1_FIFO_CFG 0x00005000 +#define CS35L36_SWIRE_DP2_FIFO_CFG 0x00005004 +#define CS35L36_SWIRE_DP3_FIFO_CFG 0x00005008 +#define CS35L36_SWIRE_PCM_RX_DATA 0x0000500C +#define CS35L36_SWIRE_FS_SEL 0x00005010 +#define CS35L36_SPARE_CP_BITS 0x00005C00 +#define CS35L36_AMP_DIG_VOL_CTRL 0x00006000 +#define CS35L36_VPBR_CFG 0x00006404 +#define CS35L36_VBBR_CFG 0x00006408 +#define CS35L36_VPBR_STATUS 0x0000640C +#define CS35L36_VBBR_STATUS 0x00006410 +#define CS35L36_OVERTEMP_CFG 0x00006414 +#define CS35L36_AMP_ERR_VOL 0x00006418 +#define CS35L36_CLASSH_CFG 0x00006800 +#define CS35L36_CLASSH_FET_DRV_CFG 0x00006804 +#define CS35L36_NG_CFG 0x00006808 +#define CS35L36_AMP_GAIN_CTRL 0x00006C04 +#define CS35L36_PWM_MOD_IO_CTRL 0x0000706C +#define CS35L36_PWM_MOD_STATUS 0x00007070 +#define CS35L36_DAC_MSM_CFG 0x00007400 +#define CS35L36_AMP_SLOPE_CTRL 0x00007410 +#define CS35L36_AMP_PDM_VOLUME 0x00007E04 +#define CS35L36_AMP_PDM_RATE_CTRL 0x00007E08 +#define CS35L36_PDM_CH_SEL 0x00007E10 +#define CS35L36_AMP_NG_CTRL 0x00007E14 +#define CS35L36_PDM_HIGHFILT_CTRL 0x00007E3C +#define CS35L36_INT1_STATUS 0x00D00000 +#define CS35L36_INT2_STATUS 0x00D00004 +#define CS35L36_INT3_STATUS 0x00D00008 +#define CS35L36_INT4_STATUS 0x00D0000C +#define CS35L36_INT1_RAW_STATUS 0x00D00020 +#define CS35L36_INT2_RAW_STATUS 0x00D00024 +#define CS35L36_INT3_RAW_STATUS 0x00D00028 +#define CS35L36_INT4_RAW_STATUS 0x00D0002C +#define CS35L36_INT1_MASK 0x00D00040 +#define CS35L36_INT2_MASK 0x00D00044 +#define CS35L36_INT3_MASK 0x00D00048 +#define CS35L36_INT4_MASK 0x00D0004C +#define CS35L36_INT1_EDGE_LVL_CTRL 0x00D00060 +#define CS35L36_INT3_EDGE_LVL_CTRL 0x00D00068 +#define CS35L36_PAC_INT_STATUS 0x00D00200 +#define CS35L36_PAC_INT_RAW_STATUS 0x00D00210 +#define CS35L36_PAC_INT_FLUSH_CTRL 0x00D00218 +#define CS35L36_PAC_INT0_CTRL 0x00D00220 +#define CS35L36_PAC_INT1_CTRL 0x00D00224 +#define CS35L36_PAC_INT2_CTRL 0x00D00228 +#define CS35L36_PAC_INT3_CTRL 0x00D0022C +#define CS35L36_PAC_INT4_CTRL 0x00D00230 +#define CS35L36_PAC_INT5_CTRL 0x00D00234 +#define CS35L36_PAC_INT6_CTRL 0x00D00238 +#define CS35L36_PAC_INT7_CTRL 0x00D0023C +#define CS35L36_PAC_PMEM_WORD0 0x00E02800 +#define CS35L36_PAC_PMEM_WORD1 0x00E02804 +#define CS35L36_PAC_PMEM_WORD1023 0x00E037FC + +#define CS35L36_INTPAC_REG_COUNT 25 +#define CS35L36_CHIP_ID 0x00035A36 + +#define CS35L36_INT_OUTPUT_EN_MASK 0x01 +#define CS35L36_INT_GPIO_SEL_MASK 0x02 +#define CS35L36_INT_GPIO_SEL_SHIFT 1 +#define CS35L36_INT_POL_SEL_MASK 0x04 +#define CS35L36_INT_POL_SEL_SHIFT 2 +#define CS35L36_INT_DRV_SEL_MASK 0x20 +#define CS35L36_INT_DRV_SEL_SHIFT 5 +#define CS35L36_IRQ_SRC_MASK 0x08 +#define CS35L36_IRQ_SRC_SHIFT 3 + +#define CS35L36_SCLK_MSTR_MASK 0x40 +#define CS35L36_SCLK_MSTR_SHIFT 6 +#define CS35L36_LRCLK_MSTR_MASK 0x01 +#define CS35L36_LRCLK_MSTR_SHIFT 0 +#define CS35L36_SCLK_INV_MASK 0x100 +#define CS35L36_SCLK_INV_SHIFT 8 +#define CS35L36_LRCLK_INV_MASK 0x04 +#define CS35L36_LRCLK_INV_SHIFT 2 +#define CS35L36_SCLK_FRC_MASK 0x80 +#define CS35L36_SCLK_FRC_SHIFT 7 +#define CS35L36_LRCLK_FRC_MASK 0x02 +#define CS35L36_LRCLK_FRC_SHIFT 1 + +#define CS35L36_PDM_MODE_MASK 0x01 +#define CS35L36_PDM_MODE_SHIFT 0 + +#define CS35L36_ASP_FMT_MASK 0x07 +#define CS35L36_ASP_FMT_SHIFT 0 + +#define CS35L36_ASP_RX_WIDTH_MASK 0xFF0000 +#define CS35L36_ASP_RX_WIDTH_SHIFT 16 +#define CS35L36_ASP_TX_WIDTH_MASK 0xFF +#define CS35L36_ASP_TX_WIDTH_SHIFT 0 +#define CS35L36_ASP_WIDTH_16 0x10 +#define CS35L36_ASP_WIDTH_24 0x18 +#define CS35L36_ASP_WIDTH_32 0x20 + +#define CS35L36_ASP_RX1_SLOT_MASK 0x3F +#define CS35L36_ASP_RX1_EN_MASK 0x00010000 +#define CS35L36_ASP_RX1_EN_SHIFT 16 + +#define CS35L36_ASP_TX1_SLOT_MASK 0x3F +#define CS35L36_ASP_TX2_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX2_SLOT_SHIFT 16 +#define CS35L36_ASP_TX3_SLOT_MASK 0x3F +#define CS35L36_ASP_TX4_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX4_SLOT_SHIFT 16 +#define CS35L36_ASP_TX5_SLOT_MASK 0x3F +#define CS35L36_ASP_TX6_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX6_SLOT_SHIFT 16 +#define CS35L36_ASP_TX7_SLOT_MASK 0x3F +#define CS35L36_ASP_TX8_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX8_SLOT_SHIFT 16 +#define CS35L36_ASP_TX_HIZ_MASK 0x200000 + +#define CS35L36_APS_TX_SEL_MASK 0x7F + +#define CS35L36_ASP_TX1_EN_MASK 0x01 +#define CS35L36_ASP_TX2_EN_MASK 0x02 +#define CS35L36_ASP_TX2_EN_SHIFT 1 +#define CS35L36_ASP_TX3_EN_MASK 0x04 +#define CS35L36_ASP_TX3_EN_SHIFT 2 +#define CS35L36_ASP_TX4_EN_MASK 0x08 +#define CS35L36_ASP_TX4_EN_SHIFT 3 +#define CS35L36_ASP_TX5_EN_MASK 0x10 +#define CS35L36_ASP_TX5_EN_SHIFT 4 +#define CS35L36_ASP_TX6_EN_MASK 0x20 +#define CS35L36_ASP_TX6_EN_SHIFT 5 +#define CS35L36_ASP_TX7_EN_MASK 0x40 +#define CS35L36_ASP_TX7_EN_SHIFT 6 +#define CS35L36_ASP_TX8_EN_MASK 0x80 +#define CS35L36_ASP_TX8_EN_SHIFT 7 + + +#define CS35L36_PLL_CLK_SEL_MASK 0x07 +#define CS35L36_PLL_CLK_SEL_SHIFT 0 +#define CS35L36_PLLSRC_SCLK 0 +#define CS35L36_PLLSRC_LRCLK 1 +#define CS35L36_PLLSRC_SELF 3 +#define CS35L36_PLLSRC_PDMCLK 4 +#define CS35L36_PLLSRC_MCLK 5 +#define CS35L36_PLLSRC_SWIRE 7 +#define CS35L36_REFCLK_FREQ_MASK 0x7E0 +#define CS35L36_REFCLK_FREQ_SHIFT 5 +#define CS35L36_PLL_OPENLOOP_MASK 0x800 +#define CS35L36_PLL_OPENLOOP_SHIFT 11 +#define CS35L36_PLL_REFCLK_EN_MASK 0x10 +#define CS35L36_PLL_REFCLK_EN_SHIFT 4 + + +#define CS35L36_GLOBAL_FS_MASK 0x1F +#define CS35L36_GLOBAL_FS_SHIFT 0 + +#define CS35L36_HPF_PCM_EN_MASK 0x800 +#define CS35L36_HPF_PCM_EN_SHIFT 15 +#define CS35L36_PCM_RX_SEL_MASK 0x7F +#define CS35L36_PCM_RX_SEL_SHIFT 0 + +#define CS35L36_PCM_RX_SEL_ZERO 0x00 +#define CS35L36_PCM_RX_SEL_PCM 0x08 +#define CS35L36_PCM_RX_SEL_SWIRE 0x10 +#define CS35L36_PCM_RX_SEL_DIAG 0x04 + +#define CS35L36_GLOBAL_EN_MASK 0x01 +#define CS35L36_GLOBAL_EN_SHIFT 0x00 + +#define CS35L36_AMP_PCM_INV_MASK 0x4000 +#define CS35L36_AMP_PCM_INV_SHIFT 14 + +#define CS35L36_AMP_VOL_PCM_MASK 0x3FF8 +#define CS35L36_AMP_VOL_PCM_SHIFT 3 +#define CS35L36_DIGITAL_MUTE 0x04CF + +#define CS35L36_AMP_RAMP_MASK 0x0007 +#define CS35L36_AMP_RAMP_SHIFT 0 + +#define CS35L36_AMP_MUTE_MASK 0x0010 +#define CS35L36_AMP_MUTE_SHIFT 4 + +#define CS35L36_GLOBAL_RESYNC_FS1_MASK 0x00000200 +#define CS35L36_GLOBAL_RESYNC_FS2_MASK 0x00000400 +#define CS35L36_SYNC_GLOBAL_OVR_MASK 0x00000002 +#define CS35L36_SYNC_GLOBAL_OVR_SHIFT 1 + +#define CS35L36_REFCLK_IN_MASK 0x00100000 +#define CS35L36_PLL_UNLOCK_MASK 0x00002000 + +#define CS35L36_ASP_RX_UDF_MASK 0x00000040 +#define CS35L36_ASP_RX_OVF_MASK 0x00000080 + +#define CS35L36_IMON_POL_MASK 0x02 +#define CS35L36_IMON_POL_SHIFT 1 + +#define CS35L36_VMON_POL_MASK 0x01 +#define CS35L36_VMON_POL_SHIFT 0 + +#define CS35L36_PDN_DONE 0x40 +#define CS35L36_PDN_DONE_SHIFT 6 +#define CS35L36_PUP_DONE 0x80 +#define CS35L36_PUP_DONE_SHIFT 7 +#define CS35L36_GLOBAL_EN_ASSRT 0x20 +#define CS35L36_PUP_DONE_IRQ_UNMASK 0x7F +#define CS35L36_PUP_DONE_IRQ_MASK 0xBF + +#define CS35L36_FS1_WINDOW_MASK 0x000007FF +#define CS35L36_FS2_WINDOW_MASK 0x00FFF800 +#define CS35L36_FS2_WINDOW_SHIFT 12 + +#define CS35L36_PLL_FFL_IGAIN_MASK 0x0F +#define CS35L36_PLL_IGAIN_MASK 0x3F0 +#define CS35L36_PLL_IGAIN_SHIFT 4 +#define CS35L36_PLL_IGAIN 0x04 + +#define CS35L36_BST_EN_MASK 0x30 +#define CS35L36_BST_EN 0x02 +#define CS35L36_BST_DIS_VP 0x01 +#define CS35L36_BST_DIS_EXTN 0x00 +#define CS35L36_BST_EN_SHIFT 4 +#define CS35L36_BST_MAN_IPKCOMP_MASK 0x200 +#define CS35L36_BST_MAN_IPKCOMP_SHIFT 9 + +#define CS35L36_BST_MAN_IPKCOMP_EN_MASK 0x100 +#define CS35L36_BST_MAN_IPKCOMP_EN_SHIFT 8 + +#define CS35L36_BST_IPK_MASK 0x7F +#define CS35L36_BST_OVP_THLD_MASK 0x3F +#define CS35L36_BST_OVP_THLD_11V 0x10 +#define CS35L36_BST_OVP_TRIM_MASK 0x00078000 +#define CS35L36_BST_OVP_TRIM_SHIFT 15 +#define CS35L36_BST_OVP_TRIM_11V 0x0C +#define CS35L36_BST_CTRL_LIM_MASK 0x04 +#define CS35L36_BST_CTRL_LIM_SHIFT 2 +#define CS35L36_BST_CTRL_10V_CLAMP 0x96 + +#define CS35L36_NG_AMP_EN_MASK 0x3F00 +#define CS35L36_NG_DELAY_MASK 0x70 +#define CS35L36_NG_DELAY_SHIFT 4 +#define CS35L36_AMP_ZC_SHIFT 10 +#define CS35L36_PDM_LDM_ENTER_SHIFT 3 +#define CS35L36_PDM_LDM_EXIT_SHIFT 4 + +#define CS35L36_BSTCVRT_K1_MASK 0xFF +#define CS35L36_BSTCVRT_K2_MASK 0xFF00 +#define CS35L36_BSTCVRT_K2_SHIFT 8 +#define CS35L36_BSTCVRT_SLOPE_MASK 0xFF00 +#define CS35L36_BSTCVRT_SLOPE_SHIFT 8 +#define CS35L36_BSTCVRT_CCMFREQ_MASK 0x0F +#define CS35L36_BSTCVRT_LBSTVAL_MASK 0x03 +#define CS35L35_BSTCVRT_CTL_MASK 0xFF +#define CS35L35_BSTCVRT_CTL_SEL_MASK 0x03 +#define CS35L36_DCM_AUTO_MASK 0x01 + +#define CS35L36_INT1_MASK_DEFAULT 0xF9BA7FFF +#define CS35L36_INT1_MASK_RESET 0xFFFFFFFF +#define CS35L36_INT3_MASK_DEFAULT 0xFFFFEFFF +#define CS35L36_INT3_MASK_RESET 0xFFFFFFFF + + +#define CS35L36_AMP_SHORT_ERR 0x1000 +#define CS35L36_BST_SHORT_ERR 0x40000 +#define CS35L36_TEMP_WARN 0x2000000 +#define CS35L36_TEMP_ERR 0x4000000 +#define CS35L36_BST_OVP_ERR 0x10000 +#define CS35L36_BST_DCM_UVP_ERR 0x20000 + +#define CS35L36_AMP_SHORT_ERR_RLS 0x02 +#define CS35L36_BST_SHORT_ERR_RLS 0x04 +#define CS35L36_BST_OVP_ERR_RLS 0x08 +#define CS35L36_BST_UVP_ERR_RLS 0x10 +#define CS35L36_TEMP_WARN_ERR_RLS 0x20 +#define CS35L36_TEMP_ERR_RLS 0x40 +#define CS35L36_TEMP_THLD_MASK 0x03 + +#define CS35L36_REV_B0 0xb0 +#define CS35L36_REV_A0 0xa0 +#define CS35L36_B0_PAC_PATCH 0x00DD0102 + +#define CS35L36_OTP_ECC_EN_MASK 0x400 +#define CS35L36_OTP_ECC_EN_SHIFT 10 +#define CS35L36_OTP_RUN_BOOT_MASK 0x01 +#define CS35L36_OTP_BOOT_DONE 0x2000000 +#define CS35L36_PAC_RESET_MASK 0x04 +#define CS35L36_PAC_RESET_SHIFT 2 +#define CS35L36_PAC_STALL_MASK 0x02 +#define CS35L36_PAC_STALL_SHIFT 1 +#define CS35L36_PAC_ENABLE_MASK 0x00000001 +#define CS35L36_PAC_MEM_ACCESS 0x01 +#define CS35L36_PAC_MEM_ACCESS_CLR 0 +#define CS35L36_SOFT_RESET 0x5AAA +#define CS35L36_MCU_BOOT_COMPLETE 0x02 +#define CS35L36_MCU_CONFIG_UNMASK 0x00FEFFFF +#define CS35L36_MCU_CONFIG_CLR 0x00010000 +#define CS35L36_MCU_CONFIG_MASK 0x00FFFFFF +#define CS35L36_GPIO_INT_SEL_MASK 0x0000003B +#define CS35L36_GPIO_INT_SEL_UNMASK 0x0000003A +#define CS35L36_PAC_RESET 0x00000000 +#define CS35L36_OTP_REV_MASK 0x00FF0000 +#define CS35L36_OTP_REV_L37 0x00CC0000 +#define CS35L36_12V_L37 37 +#define CS35L36_10V_L36 36 + +#define CS35L36_VPBR_EN_MASK 0x00001000 +#define CS35L36_VPBR_EN_SHIFT 12 + +#define CS35L36_VPBR_THLD_MASK 0x0000001F +#define CS35L36_VPBR_THLD_SHIFT 0 +#define CS35L36_VPBR_MAX_ATTN_MASK 0x00000F00 +#define CS35L36_VPBR_MAX_ATTN_SHIFT 8 +#define CS35L36_VPBR_ATK_VOL_MASK 0x0000F000 +#define CS35L36_VPBR_ATK_VOL_SHIFT 12 +#define CS35L36_VPBR_ATK_RATE_MASK 0x00070000 +#define CS35L36_VPBR_ATK_RATE_SHIFT 16 +#define CS35L36_VPBR_WAIT_MASK 0x00180000 +#define CS35L36_VPBR_WAIT_SHIFT 19 +#define CS35L36_VPBR_REL_RATE_MASK 0x00E00000 +#define CS35L36_VPBR_REL_RATE_SHIFT 21 +#define CS35L36_VPBR_MUTE_EN_MASK 0x01000000 +#define CS35L36_VPBR_MUTE_EN_SHIFT 24 + +#define CS35L36_OSC_FREQ_TRIM_MASK 0x070 +#define CS35L36_OSC_TRIM_DONE 0x08 + +#define CS35L36_FS1_DEFAULT_VAL 16 +#define CS35L36_FS2_DEFAULT_VAL 36 +#define CS35L36_FS_NOM_6MHZ 6000000 + +#define CS35L36_TEST_UNLOCK1 0x00005555 +#define CS35L36_TEST_UNLOCK2 0x0000AAAA +#define CS35L36_TEST_LOCK1 0x0000CCCC +#define CS35L36_TEST_LOCK2 0x00003333 + +#define CS35L36_PAC_PROG_MEM 512 + +#define CS35L36_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define CS35L36_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +extern const int cs35l36_a0_pac_patch[CS35L36_PAC_PROG_MEM]; + +#endif -- 1.9.1

3 2

[alsa-devel] [PATCH] ALSA: hda - Add quirk for HP EliteBook 840 G5
by Jurica Vukadin 07 Feb '19

07 Feb '19

This enables mute LED support and fixes switching jacks when the laptop is docked. Signed-off-by: Jurica Vukadin <jurica.vukadin(a)rt-rk.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 152f54137082..a4ee7656d9ee 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -924,6 +924,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), -- 2.20.1

2 1

[alsa-devel] Applied "dt-bindings: sound: Document jz4740-codec bindings" to the asoc tree
by Mark Brown 07 Feb '19

07 Feb '19

The patch dt-bindings: sound: Document jz4740-codec bindings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7c536bade37f11833ea358fbb84aca045b157fa4 Mon Sep 17 00:00:00 2001 From: Paul Cercueil <paul(a)crapouillou.net> Date: Thu, 7 Feb 2019 10:31:38 -0300 Subject: [PATCH] dt-bindings: sound: Document jz4740-codec bindings Add documentation about how to probe the jz4740-codec driver from devicetree. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/ingenic,jz4740-codec.txt | 20 +++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt new file mode 100644 index 000000000000..1ffcade87e7b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt @@ -0,0 +1,20 @@ +Ingenic JZ4740 codec controller + +Required properties: +- compatible : "ingenic,jz4740-codec" +- reg : codec registers location and length +- clocks : phandle to the AIC clock. +- clock-names: must be set to "aic". +- #sound-dai-cells: Must be set to 0. + +Example: + +codec: audio-codec@10020080 { + compatible = "ingenic,jz4740-codec"; + reg = <0x10020080 0x8>; + + #sound-dai-cells = <0>; + + clocks = <&cgu JZ4740_CLK_AIC>; + clock-names = "aic"; +}; -- 2.20.1

1 0

[alsa-devel] Applied "dt-bindings: sound: Document jz4725b-codec bindings" to the asoc tree
by Mark Brown 07 Feb '19

07 Feb '19

The patch dt-bindings: sound: Document jz4725b-codec bindings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f58f2b0a9f6ecfd76cb08fe1c4894ab009a30c66 Mon Sep 17 00:00:00 2001 From: Paul Cercueil <paul(a)crapouillou.net> Date: Thu, 7 Feb 2019 10:31:39 -0300 Subject: [PATCH] dt-bindings: sound: Document jz4725b-codec bindings Add documentation about how to probe the jz4725b-codec driver from devicetree. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/ingenic,jz4725b-codec.txt | 20 +++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt new file mode 100644 index 000000000000..05adc0d47b13 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt @@ -0,0 +1,20 @@ +Ingenic JZ4725B codec controller + +Required properties: +- compatible : "ingenic,jz4725b-codec" +- reg : codec registers location and length +- clocks : phandle to the AIC clock. +- clock-names: must be set to "aic". +- #sound-dai-cells: Must be set to 0. + +Example: + +codec: audio-codec@100200a4 { + compatible = "ingenic,jz4725b-codec"; + reg = <0x100200a4 0x8>; + + #sound-dai-cells = <0>; + + clocks = <&cgu JZ4725B_CLK_AIC>; + clock-names = "aic"; +}; -- 2.20.1

1 0

[alsa-devel] [PATCH v2 1/6] dt-bindings: sound: Document jz4740-codec bindings
by Paul Cercueil 07 Feb '19

07 Feb '19

Add documentation about how to probe the jz4740-codec driver from devicetree. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- Changes: v2: No change .../bindings/sound/ingenic,jz4740-codec.txt | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt new file mode 100644 index 000000000000..1ffcade87e7b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt @@ -0,0 +1,20 @@ +Ingenic JZ4740 codec controller + +Required properties: +- compatible : "ingenic,jz4740-codec" +- reg : codec registers location and length +- clocks : phandle to the AIC clock. +- clock-names: must be set to "aic". +- #sound-dai-cells: Must be set to 0. + +Example: + +codec: audio-codec@10020080 { + compatible = "ingenic,jz4740-codec"; + reg = <0x10020080 0x8>; + + #sound-dai-cells = <0>; + + clocks = <&cgu JZ4740_CLK_AIC>; + clock-names = "aic"; +}; -- 2.11.0

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[alsa-devel] Applied "ASoC: codecs: jz4740: Add support for devicetree" to the asoc tree
by Mark Brown 07 Feb '19

07 Feb '19

The patch ASoC: codecs: jz4740: Add support for devicetree has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 030a79e97730976f54299634fa3bbb7a99e32b71 Mon Sep 17 00:00:00 2001 From: Paul Cercueil <paul(a)crapouillou.net> Date: Thu, 7 Feb 2019 10:31:41 -0300 Subject: [PATCH] ASoC: codecs: jz4740: Add support for devicetree Add support for probing the driver from devicetree. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/jz4740.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 9b3e1227a971..974e17fa1911 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -346,10 +346,19 @@ static int jz4740_codec_probe(struct platform_device *pdev) return ret; } +#ifdef CONFIG_OF +static const struct of_device_id jz4740_codec_of_matches[] = { + { .compatible = "ingenic,jz4740-codec", }, + { } +}; +MODULE_DEVICE_TABLE(of, jz4740_codec_of_matches); +#endif + static struct platform_driver jz4740_codec_driver = { .probe = jz4740_codec_probe, .driver = { .name = "jz4740-codec", + .of_match_table = of_match_ptr(jz4740_codec_of_matches), }, }; -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: samsung: Prevent clk_get_rate() calls in atomic context
by Sylwester Nawrocki 07 Feb '19

07 Feb '19

This patch moves clk_get_rate() call from trigger() to hw_params() callback to avoid calling sleeping clk API from atomic context and prevent deadlock as indicated below. Before this change clk_get_rate() was being called with same spinlock held as the one passed to the clk API when registering clocks exposed by the I2S driver. [ 82.109780] BUG: sleeping function called from invalid context at kernel/locking/mutex.c:908 [ 82.117009] in_atomic(): 1, irqs_disabled(): 128, pid: 1554, name: speaker-test [ 82.124235] 3 locks held by speaker-test/1554: [ 82.128653] #0: cc8c5328 (snd_pcm_link_rwlock){...-}, at: snd_pcm_stream_lock_irq+0x20/0x38 [ 82.137058] #1: ec9eda17 (&(&substream->self_group.lock)->rlock){..-.}, at: snd_pcm_ioctl+0x900/0x1268 [ 82.146417] #2: 6ac279bf (&(&pri_dai->spinlock)->rlock){..-.}, at: i2s_trigger+0x64/0x6d4 [ 82.154650] irq event stamp: 8144 [ 82.157949] hardirqs last enabled at (8143): [<c0a0f574>] _raw_read_unlock_irq+0x24/0x5c [ 82.166089] hardirqs last disabled at (8144): [<c0a0f6a8>] _raw_read_lock_irq+0x18/0x58 [ 82.174063] softirqs last enabled at (8004): [<c01024e4>] __do_softirq+0x3a4/0x66c [ 82.181688] softirqs last disabled at (7997): [<c012d730>] irq_exit+0x140/0x168 [ 82.188964] Preemption disabled at: [ 82.188967] [<00000000>] (null) [ 82.195728] CPU: 6 PID: 1554 Comm: speaker-test Not tainted 5.0.0-rc5-00192-ga6e6caca8f03 #191 [ 82.204302] Hardware name: SAMSUNG EXYNOS (Flattened Device Tree) [ 82.210376] [<c0111a54>] (unwind_backtrace) from [<c010d8f4>] (show_stack+0x10/0x14) [ 82.218084] [<c010d8f4>] (show_stack) from [<c09ef004>] (dump_stack+0x90/0xc8) [ 82.225278] [<c09ef004>] (dump_stack) from [<c0152980>] (___might_sleep+0x22c/0x2c8) [ 82.232990] [<c0152980>] (___might_sleep) from [<c0a0a2e4>] (__mutex_lock+0x28/0xa3c) [ 82.240788] [<c0a0a2e4>] (__mutex_lock) from [<c0a0ad80>] (mutex_lock_nested+0x1c/0x24) [ 82.248763] [<c0a0ad80>] (mutex_lock_nested) from [<c04923dc>] (clk_prepare_lock+0x78/0xec) [ 82.257079] [<c04923dc>] (clk_prepare_lock) from [<c049538c>] (clk_core_get_rate+0xc/0x5c) [ 82.265309] [<c049538c>] (clk_core_get_rate) from [<c0766b18>] (i2s_trigger+0x490/0x6d4) [ 82.273369] [<c0766b18>] (i2s_trigger) from [<c074fec4>] (soc_pcm_trigger+0x100/0x140) [ 82.281254] [<c074fec4>] (soc_pcm_trigger) from [<c07378a0>] (snd_pcm_do_start+0x2c/0x30) [ 82.289400] [<c07378a0>] (snd_pcm_do_start) from [<c07376cc>] (snd_pcm_action_single+0x38/0x78) [ 82.298065] [<c07376cc>] (snd_pcm_action_single) from [<c073a450>] (snd_pcm_ioctl+0x910/0x1268) [ 82.306734] [<c073a450>] (snd_pcm_ioctl) from [<c0292344>] (do_vfs_ioctl+0x90/0x9ec) [ 82.314443] [<c0292344>] (do_vfs_ioctl) from [<c0292cd4>] (ksys_ioctl+0x34/0x60) [ 82.321808] [<c0292cd4>] (ksys_ioctl) from [<c0101000>] (ret_fast_syscall+0x0/0x28) [ 82.329431] Exception stack(0xeb875fa8 to 0xeb875ff0) [ 82.334459] 5fa0: 00033c18 b6e31000 00000004 00004142 00033d80 00033d80 [ 82.342605] 5fc0: 00033c18 b6e31000 00008000 00000036 00008000 00000000 beea38a8 00008000 [ 82.350748] 5fe0: b6e3142c beea384c b6da9a30 b6c9212c [ 82.355789] [ 82.357245] ====================================================== [ 82.363397] WARNING: possible circular locking dependency detected [ 82.369551] 5.0.0-rc5-00192-ga6e6caca8f03 #191 Tainted: G W [ 82.376395] ------------------------------------------------------ [ 82.382548] speaker-test/1554 is trying to acquire lock: [ 82.387834] 6d2007f4 (prepare_lock){+.+.}, at: clk_prepare_lock+0x78/0xec [ 82.394593] [ 82.394593] but task is already holding lock: [ 82.400398] 6ac279bf (&(&pri_dai->spinlock)->rlock){..-.}, at: i2s_trigger+0x64/0x6d4 [ 82.408197] [ 82.408197] which lock already depends on the new lock. [ 82.416343] [ 82.416343] the existing dependency chain (in reverse order) is: [ 82.423795] [ 82.423795] -> #1 (&(&pri_dai->spinlock)->rlock){..-.}: [ 82.430472] clk_mux_set_parent+0x34/0xb8 [ 82.434975] clk_core_set_parent_nolock+0x1c4/0x52c [ 82.440347] clk_set_parent+0x38/0x6c [ 82.444509] of_clk_set_defaults+0xc8/0x308 [ 82.449186] of_clk_add_provider+0x84/0xd0 [ 82.453779] samsung_i2s_probe+0x408/0x5f8 [ 82.458376] platform_drv_probe+0x48/0x98 [ 82.462879] really_probe+0x224/0x3f4 [ 82.467037] driver_probe_device+0x70/0x1c4 [ 82.471716] bus_for_each_drv+0x44/0x8c [ 82.476049] __device_attach+0xa0/0x138 [ 82.480382] bus_probe_device+0x88/0x90 [ 82.484715] deferred_probe_work_func+0x6c/0xbc [ 82.489741] process_one_work+0x200/0x740 [ 82.494246] worker_thread+0x2c/0x4c8 [ 82.498408] kthread+0x128/0x164 [ 82.502131] ret_from_fork+0x14/0x20 [ 82.506204] (null) [ 82.508976] [ 82.508976] -> #0 (prepare_lock){+.+.}: [ 82.514264] __mutex_lock+0x60/0xa3c [ 82.518336] mutex_lock_nested+0x1c/0x24 [ 82.522756] clk_prepare_lock+0x78/0xec [ 82.527088] clk_core_get_rate+0xc/0x5c [ 82.531421] i2s_trigger+0x490/0x6d4 [ 82.535494] soc_pcm_trigger+0x100/0x140 [ 82.539913] snd_pcm_do_start+0x2c/0x30 [ 82.544246] snd_pcm_action_single+0x38/0x78 [ 82.549012] snd_pcm_ioctl+0x910/0x1268 [ 82.553345] do_vfs_ioctl+0x90/0x9ec [ 82.557417] ksys_ioctl+0x34/0x60 [ 82.561229] ret_fast_syscall+0x0/0x28 [ 82.565477] 0xbeea384c [ 82.568421] [ 82.568421] other info that might help us debug this: [ 82.568421] [ 82.576394] Possible unsafe locking scenario: [ 82.576394] [ 82.582285] CPU0 CPU1 [ 82.586792] ---- ---- [ 82.591297] lock(&(&pri_dai->spinlock)->rlock); [ 82.595977] lock(prepare_lock); [ 82.601782] lock(&(&pri_dai->spinlock)->rlock); [ 82.608975] lock(prepare_lock); [ 82.612268] [ 82.612268] *** DEADLOCK *** Fixes: 647d04f8e07a ("ASoC: samsung: i2s: Ensure the RCLK rate is properly determined") Reported-by: Krzysztof Kozłowski <krzk(a)kernel.org> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d6c62aa13041..ce00fe2f6aae 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -700,6 +700,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, { struct i2s_dai *i2s = to_info(dai); u32 mod, mask = 0, val = 0; + struct clk *rclksrc; unsigned long flags; WARN_ON(!pm_runtime_active(dai->dev)); @@ -782,6 +783,10 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, i2s->frmclk = params_rate(params); + rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC]; + if (rclksrc && !IS_ERR(rclksrc)) + i2s->rclk_srcrate = clk_get_rate(rclksrc); + return 0; } @@ -886,11 +891,6 @@ static int config_setup(struct i2s_dai *i2s) return 0; if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { - struct clk *rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC]; - - if (rclksrc && !IS_ERR(rclksrc)) - i2s->rclk_srcrate = clk_get_rate(rclksrc); - psr = i2s->rclk_srcrate / i2s->frmclk / rfs; writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR); dev_dbg(&i2s->pdev->dev, -- 2.20.1

2 3

[alsa-devel] [PATCH] ASoC: rsnd: ssiu: correct shift bit for ssiu9
by Jiada Wang 07 Feb '19

07 Feb '19

Currently "0xf << 36" is used to clear SSIU-9 internal buffer state, which overflows 32-bit value according to user reference manual, it is always bit4 ~ bit7 of SSI_SYS_STATUS[1,3,5,7] registers indicate SSIU-9's buffer state, so "0xf << 4" should be used. This patch fix incorrect shifting issue in SSIU-9 case Fixes: commit b7169ddea2f2 ("ASoC: rsnd: remove RSND_REG_ from rsnd_reg") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/ssiu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 2f3085a54eba..b11677489a5e 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -79,7 +79,7 @@ static int rsnd_ssiu_init(struct rsnd_mod *mod, break; case 9: for (i = 0; i < 4; i++) - rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << (id * 4)); + rsnd_mod_write(mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << 4); break; } -- 2.19.2

3 2

[alsa-devel] [asoc:for-5.1 98/129] sound/soc/codecs/wcd-clsh-v2.c:545:2-3: Unneeded semicolon
by kbuild test robot 07 Feb '19

07 Feb '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.1 head: 0f747bb273790f49be4660521d86d7cebd1bbe1e commit: cc2e324d39b26d62599d056f5cb905a025b909a3 [98/129] ASoC: wcd9335: add CLASS-H Controller support coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/wcd-clsh-v2.c:545:2-3: Unneeded semicolon sound/soc/codecs/wcd-clsh-v2.c:211:2-3: Unneeded semicolon sound/soc/codecs/wcd-clsh-v2.c:250:2-3: Unneeded semicolon Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [PATCH][next] ASoC: codecs: jz4725b: fix spelling mistake "Deemphatize" -> "Deemphasize"
by Colin King 07 Feb '19

07 Feb '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in the SOC_SINGLE control name. Fix this. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/jz4725b.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index e3dba92f30bd..24b1b23b99c9 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -161,7 +161,7 @@ static const struct snd_kcontrol_new jz4725b_codec_controls[] = { SOC_SINGLE("Master Playback Switch", JZ4725B_CODEC_REG_CR1, REG_CR1_DAC_MUTE_OFFSET, 1, 1), - SOC_SINGLE("Deemphatize Filter Playback Switch", + SOC_SINGLE("Deemphasize Filter Playback Switch", JZ4725B_CODEC_REG_CR2, REG_CR2_DAC_DEEMP_OFFSET, 1, 0), -- 2.20.1

2 1

[alsa-devel] [PATCH v15 0/8] ASoC: sun4i-codec: Add Line-In, FM-In, Mic 2
by Danny Milosavljevic 07 Feb '19

07 Feb '19

This patchset adds some mixer controls to sun4i-codec for the Allwinner A10 and the Allwinner A20. It also adds the PGA for the MIC2 preamp. Where possible, it uses SOC_DAPM_DOUBLE in order to cut down on the number of distinct controls in alsamixer. v15 changes compared to v14 are: - Instead of adding controls dynamically, add an extra sun7i_codec_controls structure and duplicate the sun4i_codec_controls there, with small changes. - Split Mic Playback Volume into extra patch. - LEFT OFF adding Capture Selects since it's unclear how we want to structure it. - LEFT OFF adding Differential Line Source since it might also change if the Capture Select implementation is different. v14 changes compared to v13 are: - Merged some of the patches together if it made sense to test them together. - Use snd_soc_component_driver. - Moved SUN4I_CODEC_DAC_ACTL_LFMS, SUN4I_CODEC_DAC_ACTL_RFMS to the correct patch in the series. - Kept "Left Mixer Left DAC Playback Switch", "Right Mixer Left DAC Playback Switch" and "Right Mixer Right DAC Playback Switch" unchanged compared to the released version - for backward compatibility. v13 changes compared to v12 are: - Added my "Signed-off-by". - Clarified some commit message text. v12 changes compared to v11 are: - Split up patchset in another way. - Renamed "Mic1 Capture Volume" to "Mic1 Boost Volume". - Renamed "Mic2 Capture Volume" to "Mic2 Boost Volume". - Renamed "Line Capture Volume" to "Line Boost Volume". - Renamed "Differential Line Capture Switch" to "Differential Line Source". v11 changes compared to v10 are: - Split up patchset. - Fixed typo in Differential Line Capture Switch. - Renamed "Non-Differential" value to "Stereo". - Removed duplicate PA Volume mixer control. v10 changes compared to v9 are: - Use SOC_DAPM_DOUBLE where possible and it makes sense in order to cut down on the number of controls. v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (8): ASoC: sun4i-codec: Add MIC2 Pre-Amplifier, Mic2 ASoC: sun4i-codec: Add Mic Playback Volume ASoC: sun4i-codec: Add sun7i_codec_controls, sun7i_codec_codec. ASoC: sun4i-codec: Add Mic1 Boost Volume, Mic2 Boost Volume ASoC: sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls ASoC: sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch ASoC: sun4i-codec: Add FM Playback Volume, FM Left, FM Right, FM Playback Switch ASoC: sun4i-codec: Add Line Playback Volume, Line Boost Volume, Line Right, Line Left, Line Playback Switch sound/soc/sunxi/sun4i-codec.c | 149 ++++++++++++++++++++++++++++++---- 1 file changed, 134 insertions(+), 15 deletions(-)

3 15

[alsa-devel] Applied "ASoC: sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls" to the asoc tree
by Mark Brown 07 Feb '19

07 Feb '19

The patch ASoC: sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0bbb8e83cfe02d670380798ae1876668f8bbd00c Mon Sep 17 00:00:00 2001 From: Danny Milosavljevic <dannym(a)scratchpost.org> Date: Wed, 30 Jan 2019 11:39:41 +0100 Subject: [PATCH] ASoC: sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls Since it's now possible to have a DAPM mixer control with multiple channels, use it to cut down the total number of controls. Keep "Left Mixer Left DAC Playback Switch" and "Right Mixer Right DAC Playback Switch" name & layout the same as before for compatibility. Signed-off-by: Danny Milosavljevic <dannym(a)scratchpost.org> Reviewed-by: Chen-Yu Tsai <wens(a)csie.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/sun4i-codec.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 9d509ede22c7..279a7880d623 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -718,15 +718,15 @@ static const struct snd_kcontrol_new sun7i_codec_controls[] = { sun7i_codec_micin_preamp_gain_scale), }; -static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, - SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0), -}; - -static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { - SOC_DAPM_SINGLE("Right DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, - SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0), - SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, +static const struct snd_kcontrol_new sun4i_codec_mixer_controls[] = { + SOC_DAPM_SINGLE("Left Mixer Left DAC Playback Switch", + SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACLMIXS, + 1, 0), + SOC_DAPM_SINGLE("Right Mixer Right DAC Playback Switch", + SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_RDACRMIXS, + 1, 0), + SOC_DAPM_SINGLE("Right Mixer Left DAC Playback Switch", + SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), }; @@ -762,11 +762,11 @@ static const struct snd_soc_dapm_widget sun4i_codec_codec_dapm_widgets[] = { /* Mixers */ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, - sun4i_codec_left_mixer_controls, - ARRAY_SIZE(sun4i_codec_left_mixer_controls)), + sun4i_codec_mixer_controls, + ARRAY_SIZE(sun4i_codec_mixer_controls)), SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, - sun4i_codec_right_mixer_controls, - ARRAY_SIZE(sun4i_codec_right_mixer_controls)), + sun4i_codec_mixer_controls, + ARRAY_SIZE(sun4i_codec_mixer_controls)), /* Global Mixer Enable */ SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL, @@ -808,12 +808,12 @@ static const struct snd_soc_dapm_route sun4i_codec_codec_dapm_routes[] = { /* Right Mixer Routes */ { "Right Mixer", NULL, "Mixer Enable" }, - { "Right Mixer", "Left DAC Playback Switch", "Left DAC" }, - { "Right Mixer", "Right DAC Playback Switch", "Right DAC" }, + { "Right Mixer", "Right Mixer Left DAC Playback Switch", "Left DAC" }, + { "Right Mixer", "Right Mixer Right DAC Playback Switch", "Right DAC" }, /* Left Mixer Routes */ { "Left Mixer", NULL, "Mixer Enable" }, - { "Left Mixer", "Left DAC Playback Switch", "Left DAC" }, + { "Left Mixer", "Left Mixer Left DAC Playback Switch", "Left DAC" }, /* Power Amplifier Routes */ { "Power Amplifier", "Mixer Playback Switch", "Left Mixer" }, -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch" to the asoc tree
by Mark Brown 07 Feb '19

07 Feb '19

The patch ASoC: sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 44a1f4e8cfcc5492c4e33b983987506719714786 Mon Sep 17 00:00:00 2001 From: Danny Milosavljevic <dannym(a)scratchpost.org> Date: Wed, 30 Jan 2019 11:39:42 +0100 Subject: [PATCH] ASoC: sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch Add Mic1 Playback Switch and Mic2 Playback Switch for Allwinner A10 and Allwinner A20. Signed-off-by: Danny Milosavljevic <dannym(a)scratchpost.org> Reviewed-by: Chen-Yu Tsai <wens(a)csie.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/sun4i-codec.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 279a7880d623..87bb16e59da7 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -68,6 +68,10 @@ #define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15) #define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14) #define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13) +#define SUN4I_CODEC_DAC_ACTL_MIC1LS (12) +#define SUN4I_CODEC_DAC_ACTL_MIC1RS (11) +#define SUN4I_CODEC_DAC_ACTL_MIC2LS (10) +#define SUN4I_CODEC_DAC_ACTL_MIC2RS (9) #define SUN4I_CODEC_DAC_ACTL_DACPAS (8) #define SUN4I_CODEC_DAC_ACTL_MIXPAS (7) #define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6) @@ -728,6 +732,12 @@ static const struct snd_kcontrol_new sun4i_codec_mixer_controls[] = { SOC_DAPM_SINGLE("Right Mixer Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), + SOC_DAPM_DOUBLE("Mic1 Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1LS, + SUN4I_CODEC_DAC_ACTL_MIC1RS, 1, 0), + SOC_DAPM_DOUBLE("Mic2 Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2LS, + SUN4I_CODEC_DAC_ACTL_MIC2RS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { @@ -810,10 +820,14 @@ static const struct snd_soc_dapm_route sun4i_codec_codec_dapm_routes[] = { { "Right Mixer", NULL, "Mixer Enable" }, { "Right Mixer", "Right Mixer Left DAC Playback Switch", "Left DAC" }, { "Right Mixer", "Right Mixer Right DAC Playback Switch", "Right DAC" }, + { "Right Mixer", "Mic1 Playback Switch", "MIC1 Pre-Amplifier" }, + { "Right Mixer", "Mic2 Playback Switch", "MIC2 Pre-Amplifier" }, /* Left Mixer Routes */ { "Left Mixer", NULL, "Mixer Enable" }, { "Left Mixer", "Left Mixer Left DAC Playback Switch", "Left DAC" }, + { "Left Mixer", "Mic1 Playback Switch", "MIC1 Pre-Amplifier" }, + { "Left Mixer", "Mic2 Playback Switch", "MIC2 Pre-Amplifier" }, /* Power Amplifier Routes */ { "Power Amplifier", "Mixer Playback Switch", "Left Mixer" }, -- 2.20.1

1 0

[alsa-devel] [PATCH] ALSA: pcm: Revert capture stream behavior change in blocking mode
by Takashi Iwai 07 Feb '19

07 Feb '19

In the commit 62ba568f7aef ("ALSA: pcm: Return 0 when size < start_threshold in capture"), we changed the behavior of __snd_pcm_lib_xfer() to return immediately with 0 when a capture stream has a high start_threshold. This was intended to be a correction of the behavior consistency and looked harmless, but this was the culprit of the recent breakage reported by syzkaller, which was fixed by the commit e190161f96b8 ("ALSA: pcm: Fix tight loop of OSS capture stream"). At the time for the OSS fix, I didn't touch the behavior for ALSA native API, as assuming that this behavior actually is good. But this turned out to be also broken actually for a similar deployment, e.g. one thread goes to a write loop in blocking mode while another thread controls the start/stop of the stream manually. Overall, the original commit is harmful, and it brings less merit to keep that behavior. Let's revert it. Fixes: 62ba568f7aef ("ALSA: pcm: Return 0 when size < start_threshold in capture") Fixes: e190161f96b8 ("ALSA: pcm: Fix tight loop of OSS capture stream") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 20 ++++---------------- 1 file changed, 4 insertions(+), 16 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 5957aeb1099e..bcb06bd3d81d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2112,13 +2112,6 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream, return 0; } -/* allow waiting for a capture stream that hasn't been started */ -#if IS_ENABLED(CONFIG_SND_PCM_OSS) -#define wait_capture_start(substream) ((substream)->oss.oss) -#else -#define wait_capture_start(substream) false -#endif - /* the common loop for read/write data */ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, void *data, bool interleaved, @@ -2184,16 +2177,11 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, snd_pcm_update_hw_ptr(substream); if (!is_playback && - runtime->status->state == SNDRV_PCM_STATE_PREPARED) { - if (size >= runtime->start_threshold) { - err = snd_pcm_start(substream); - if (err < 0) - goto _end_unlock; - } else if (!wait_capture_start(substream)) { - /* nothing to do */ - err = 0; + runtime->status->state == SNDRV_PCM_STATE_PREPARED && + size >= runtime->start_threshold) { + err = snd_pcm_start(substream); + if (err < 0) goto _end_unlock; - } } avail = snd_pcm_avail(substream); -- 2.16.4

1 0

[alsa-devel] [PATCH] UCM: bytcr/PlatformEnableSeq.conf update some comments
by Hans de Goede 07 Feb '19

07 Feb '19

Commit f91cc3c7d6b7 ("Update chtrt5645 ucm variants to use bytcr/PlatformEnableSeq.conf component") updated the following 2 comments: # codec0_out settings (used if SSP2 is connected to aif1) # modem_out settings (used if SSP0 is connected to aif2) Specifically it added the " to aif1" resp. " to aif2" part of the comments. This is not correct, AIF1 / AIF2 are something which is present on Realtek codecs only, and either one can be used indepedent of SSP0 or SSP2 being used (the comments in the chtrt5645 UCM profile before this change were wrong / outdated). Besides there not being any relationship between SSP0 or SSP2 being used, bytcr/PlatformEnableSeq.conf is also used with other codecs, e.g. the ESS8316 codec where this is not applicable at all. Therefor this commit removes the " to aif?" part of the comments again to avoid confusing people reading this in the future. Cc: Russell Parker <russell.parker7(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf b/src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf index 6f5e899c..b5ee2b41 100644 --- a/src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf +++ b/src/conf/ucm/platforms/bytcr/PlatformEnableSeq.conf @@ -29,7 +29,7 @@ cset "name='pcm0_in Gain 0 Volume' 0" cset "name='pcm1_in Gain 0 Switch' off" cset "name='pcm1_in Gain 0 Volume' 0%" -# codec0_out settings (used if SSP2 is connected to aif1) +# codec0_out settings (used if SSP2 is connected) cset "name='codec_out0 mix 0 codec_in0 Switch' off" cset "name='codec_out0 mix 0 codec_in1 Switch' off" cset "name='codec_out0 mix 0 media_loop1_in Switch' off" @@ -40,7 +40,7 @@ cset "name='codec_out0 mix 0 sprot_loop_in Switch' off" cset "name='codec_out0 Gain 0 Switch' on" cset "name='codec_out0 Gain 0 Volume' 0" -# modem_out settings (used if SSP0 is connected to aif2) +# modem_out settings (used if SSP0 is connected) cset "name='modem_out mix 0 codec_in0 Switch' off" cset "name='modem_out mix 0 codec_in1 Switch' off" cset "name='modem_out mix 0 media_loop1_in Switch' off" -- 2.20.1

3 3

[alsa-devel] [PATCH v2 2/2] ASoC: cs35l36: Add device tree documentation for CS35L36
by James Schulman 06 Feb '19

06 Feb '19

Add device tree documentation for Cirrus Logic CS35L36 speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- .../devicetree/bindings/sound/cs35l36.txt | 168 +++++++++++++++++++++ 1 file changed, 168 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l36.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l36.txt b/Documentation/devicetree/bindings/sound/cs35l36.txt new file mode 100644 index 0000000..912bd16 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l36.txt @@ -0,0 +1,168 @@ +CS35L36 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l36" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2550mV to 12000mV with + increments of 50mV. + (Default) VP + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1600mA and goes to a maximum of 4500mA with increments of + 50mA. + (Default) 4.50 Amps + + - cirrus,boost-ind-nanohenry : Inductor estimation LBST reference value. + Seeds the digital boost converter's inductor estimation block with the initial + inductance value to reference. + + 1000 = 1uH (Default) + 1200 = 1.2uH + +Optional properties: + - cirrus,multi-amp-mode : Boolean to determine if there are more than + one amplifier in the system. If more than one it is best to Hi-Z the ASP + port to prevent bus contention on the output signal + + - cirrus,boost-ctl-select : Boost conerter control source selection. + Selects the source of the BST_CTL target VBST voltage for the boost + converter to generate. + 0x00 - Control Port Value + 0x01 - Class H Tracking (Default) + 0x10 - MultiDevice Sync Value + + - cirrus,amp-pcm-inv : Boolean to determine Amplifier will invert incoming + PCM data + + - cirrus,imon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound IMON feedback data + + - cirrus,vmon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound VMON feedback data + + - cirrus,dcm-mode-enable : Boost converter automatic DCM Mode enable. + This enables the digital boost converter to operate in a low power + (Discontinuous Conduction) mode during low loading conditions. + + - cirrus,weak-fet-disable : Boolean : The strength of the output drivers is + reduced when operating in a Weak-FET Drive Mode and must not be used to drive + a large load. + + - cirrus,classh-wk-fet-delay : Weak-FET entry delay. Controls the delay + (in ms) before the Class H algorithm switches to the weak-FET voltage + (after the audio falls and remains below the value specified in WKFET_AMP_THLD). + + 0 = 0ms + 1 = 5ms + 2 = 10ms + 3 = 50ms + 4 = 100ms (Default) + 5 = 200ms + 6 = 500ms + 7 = 1000ms + + - cirrus,classh-weak-fet-thld-millivolt : Weak-FET amplifier drive threshold. + Configures the signal threshold at which the PWM output stage enters + weak-FET operation. The range is 50mV to 700mV in 50mV increments. + + - cirrus,temp-warn-threshold : Amplifier overtemperature warning threshold. + Configures the threshold at which the overtemperature warning condition occurs. + When the threshold is met, the overtemperature warning attenuation is applied + and the TEMP_WARN_EINT interrupt status bit is set. + If TEMP_WARN_MASK = 0, INTb is asserted. + + 0 = 105C + 1 = 115C + 2 = 125C (Default) + 3 = 135C + + - cirrus,irq-drive-select : Selects the driver type of the selected interrupt + output. + + 0 = Open-drain + 1 = Push-pull (Default) + + - cirrus,irq-gpio-select : Selects the pin to serve as the programmable + interrupt output. + + 0 = PDM_DATA / SWIRE_SD / INT (Default) + 1 = GPIO + +Optional properties for the "cirrus,vpbr-config" Sub-node + + - cirrus,vpbr-en : VBST brownout prevention enable. Configures whether the + VBST brownout prevention algorithm is enabled or disabled. + + 0 = VBST brownout prevention disabled (default) + 1 = VBST brownout prevention enabled + + See Section 7.31.1 VPBR Config for configuration options & further details + + - cirrus,vpbr-thld : Initial VPBR threshold. Configures the VP brownout + threshold voltage + + - cirrus,cirrus,vpbr-atk-rate : Attenuation attack step rate. Configures the + amount delay between consecutive volume attenuation steps when a brownout + condition is present and the VP brownout condition is in an attacking state. + + - cirrus,vpbr-atk-vol : VP brownout prevention step size. Configures the VP + brownout prevention attacking attenuation step size when operating in either + digital volume or analog gain modes. + + - cirrus,vpbr-max-attn : Maximum attenuation that the VP brownout prevention + can apply to the audio signal. + + - cirrus,vpbr-wait : Configures the delay time between a brownout condition + no longer being present and the VP brownout prevention entering an attenuation + release state. + + - cirrus,vpbr-rel-rate : Attenuation release step rate. Configures the delay + between consecutive volume attenuation release steps when a brownout condition + is not longer present and the VP brownout is in an attenuation release state. + + - cirrus,vpbr-mute-en : During the attack state, if the vpbr-max-attn value + is reached, the error condition still remains, and this bit is set, the audio + is muted. + +Example: + +cs35l36: cs35l36@40 { + compatible = "cirrus,cs35l36"; + reg = <0x40>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio0 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-ctl-millivolt = <10000>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ctl-select = <0x00>; + cirrus,weak-fet-delay = <0x04>; + cirrus,weak-fet-thld = <0x01>; + cirrus,temp-warn-threshold = <0x01>; + cirrus,multi-amp-mode; + cirrus,irq-drive-select = <0x01>; + cirrus,irq-gpio-select = <0x01>; + + cirrus,vpbr-config { + cirrus,vpbr-en = <0x00>; + cirrus,vpbr-thld = <0x05>; + cirrus,vpbr-atk-rate = <0x02>; + cirrus,vpbr-atk-vol = <0x01>; + cirrus,vpbr-max-attn = <0x09>; + cirrus,vpbr-wait = <0x01>; + cirrus,vpbr-rel-rate = <0x05>; + cirrus,vpbr-mute-en = <0x00>; + }; +}; -- 1.9.1

1 0

[alsa-devel] Applied "ASoC: ssm2602: switch to SPDX identifier" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: ssm2602: switch to SPDX identifier has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 47306401835a095843df8d87a51848f7e5e5099e Mon Sep 17 00:00:00 2001 From: Marco Felsch <m.felsch(a)pengutronix.de> Date: Wed, 6 Feb 2019 18:00:03 +0100 Subject: [PATCH] ASoC: ssm2602: switch to SPDX identifier Drop old license header and switch to SPDX-License-Identifier. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ssm2602.c | 39 ++++++++++++-------------------------- 1 file changed, 12 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 6f79cb0fbc63..464a4d7873bb 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -1,30 +1,15 @@ -/* - * File: sound/soc/codecs/ssm2602.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: Driver for ssm2602 sound chip - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// File: sound/soc/codecs/ssm2602.c +// Author: Cliff Cai <Cliff.Cai(a)analog.com> +// +// Created: Tue June 06 2008 +// Description: Driver for ssm2602 sound chip +// +// Modified: +// Copyright 2008 Analog Devices Inc. +// +// Bugs: Enter bugs at http://blackfin.uclinux.org/ #include <linux/delay.h> #include <linux/module.h> -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c16e12010060c6c7a31f08b4a99513064cb53b7d Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Tue, 5 Feb 2019 10:22:27 -0600 Subject: [PATCH] ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables KASAN reports and additional traces point to out-of-bounds accesses to the dapm_up_seq and dapm_down_seq lookup tables. The indices used are larger than the array definition. Fix by adding missing entries for the new widget types in these two lookup tables, and align them with PGA values. Also the sequences for the following widgets were not defined. Since their values defaulted to zero, assign them explicitly snd_soc_dapm_input snd_soc_dapm_output snd_soc_dapm_vmid snd_soc_dapm_siggen snd_soc_dapm_sink Fixes: 8a70b4544ef4 ('ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.'). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c4c13419539..20bad755888b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -70,12 +70,16 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, + [snd_soc_dapm_vmid] = 3, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai_in] = 4, [snd_soc_dapm_dai_out] = 4, [snd_soc_dapm_aif_in] = 4, [snd_soc_dapm_aif_out] = 4, [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_siggen] = 5, + [snd_soc_dapm_input] = 5, + [snd_soc_dapm_output] = 5, [snd_soc_dapm_mux] = 6, [snd_soc_dapm_demux] = 6, [snd_soc_dapm_dac] = 7, @@ -83,11 +87,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer] = 8, [snd_soc_dapm_mixer_named_ctl] = 8, [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_buffer] = 9, + [snd_soc_dapm_scheduler] = 9, + [snd_soc_dapm_effect] = 9, + [snd_soc_dapm_src] = 9, + [snd_soc_dapm_asrc] = 9, + [snd_soc_dapm_encoder] = 9, + [snd_soc_dapm_decoder] = 9, [snd_soc_dapm_adc] = 10, [snd_soc_dapm_out_drv] = 11, [snd_soc_dapm_hp] = 11, [snd_soc_dapm_spk] = 11, [snd_soc_dapm_line] = 11, + [snd_soc_dapm_sink] = 11, [snd_soc_dapm_kcontrol] = 12, [snd_soc_dapm_post] = 13, }; @@ -100,13 +112,25 @@ static int dapm_down_seq[] = { [snd_soc_dapm_spk] = 3, [snd_soc_dapm_line] = 3, [snd_soc_dapm_out_drv] = 3, + [snd_soc_dapm_sink] = 3, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_buffer] = 4, + [snd_soc_dapm_scheduler] = 4, + [snd_soc_dapm_effect] = 4, + [snd_soc_dapm_src] = 4, + [snd_soc_dapm_asrc] = 4, + [snd_soc_dapm_encoder] = 4, + [snd_soc_dapm_decoder] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_siggen] = 7, + [snd_soc_dapm_input] = 7, + [snd_soc_dapm_output] = 7, [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_vmid] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_demux] = 9, [snd_soc_dapm_aif_in] = 10, -- 2.20.1

1 0

[alsa-devel] [PATCH v4 1/2] ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables
by Pierre-Louis Bossart 06 Feb '19

06 Feb '19

KASAN reports and additional traces point to out-of-bounds accesses to the dapm_up_seq and dapm_down_seq lookup tables. The indices used are larger than the array definition. Fix by adding missing entries for the new widget types in these two lookup tables, and align them with PGA values. Also the sequences for the following widgets were not defined. Since their values defaulted to zero, assign them explicitly snd_soc_dapm_input snd_soc_dapm_output snd_soc_dapm_vmid snd_soc_dapm_siggen snd_soc_dapm_sink Fixes: 8a70b4544ef4 ('ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.'). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5b74dffc9c11..2bf5bab579de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -70,12 +70,16 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, + [snd_soc_dapm_vmid] = 3, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai_in] = 4, [snd_soc_dapm_dai_out] = 4, [snd_soc_dapm_aif_in] = 4, [snd_soc_dapm_aif_out] = 4, [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_siggen] = 5, + [snd_soc_dapm_input] = 5, + [snd_soc_dapm_output] = 5, [snd_soc_dapm_mux] = 6, [snd_soc_dapm_demux] = 6, [snd_soc_dapm_dac] = 7, @@ -83,11 +87,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer] = 8, [snd_soc_dapm_mixer_named_ctl] = 8, [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_buffer] = 9, + [snd_soc_dapm_scheduler] = 9, + [snd_soc_dapm_effect] = 9, + [snd_soc_dapm_src] = 9, + [snd_soc_dapm_asrc] = 9, + [snd_soc_dapm_encoder] = 9, + [snd_soc_dapm_decoder] = 9, [snd_soc_dapm_adc] = 10, [snd_soc_dapm_out_drv] = 11, [snd_soc_dapm_hp] = 11, [snd_soc_dapm_spk] = 11, [snd_soc_dapm_line] = 11, + [snd_soc_dapm_sink] = 11, [snd_soc_dapm_kcontrol] = 12, [snd_soc_dapm_post] = 13, }; @@ -100,13 +112,25 @@ static int dapm_down_seq[] = { [snd_soc_dapm_spk] = 3, [snd_soc_dapm_line] = 3, [snd_soc_dapm_out_drv] = 3, + [snd_soc_dapm_sink] = 3, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_buffer] = 4, + [snd_soc_dapm_scheduler] = 4, + [snd_soc_dapm_effect] = 4, + [snd_soc_dapm_src] = 4, + [snd_soc_dapm_asrc] = 4, + [snd_soc_dapm_encoder] = 4, + [snd_soc_dapm_decoder] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_siggen] = 7, + [snd_soc_dapm_input] = 7, + [snd_soc_dapm_output] = 7, [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_vmid] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_demux] = 9, [snd_soc_dapm_aif_in] = 10, -- 2.17.1

2 2

[alsa-devel] Applied "ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d9111d36024de07784f2e1ba2ccf70b16035f378 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 5 Feb 2019 09:46:43 +0900 Subject: [PATCH] ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") added new rsnd_ssi_prepare() and moved rsnd_ssi_master_clk_start() to .prepare. But, ssi user count (= ssi->usrcnt) is incremented at .init (= rsnd_ssi_init()). Because of these timing exchange, ssi->usrcnt check at rsnd_ssi_master_clk_start() should be adjusted. Otherwise, 2nd master clock setup will be no check. This patch fixup this issue. Fixes: commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Reported-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Reported-by: Valentine Barshak <valentine.barshak(a)cogentembedded.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 45ef295743ec..f5afab631abb 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -286,7 +286,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->usrcnt > 1) { + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: codecs: Add jz4725b-codec driver" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: codecs: Add jz4725b-codec driver has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e9d97b05a80f27d5ba7379b108db19b0d93cf267 Mon Sep 17 00:00:00 2001 From: Paul Cercueil <paul(a)crapouillou.net> Date: Tue, 5 Feb 2019 00:11:08 -0300 Subject: [PATCH] ASoC: codecs: Add jz4725b-codec driver Add jz4725b-codec driver to support the internal CODEC found in the JZ4725B SoC from Ingenic. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/jz4725b.c | 599 +++++++++++++++++++++++++++++++++++++ 3 files changed, 613 insertions(+) create mode 100644 sound/soc/codecs/jz4725b.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a15710c8a95f..fec894c725d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -90,6 +90,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_INNO_RK3036 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC + select SND_SOC_JZ4725B_CODEC select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C @@ -581,6 +582,17 @@ config SND_SOC_JZ4740_CODEC select REGMAP_MMIO tristate +config SND_SOC_JZ4725B_CODEC + depends on MIPS || COMPILE_TEST + select REGMAP + tristate "Ingenic JZ4725B internal CODEC" + help + Enable support for the internal CODEC found in the JZ4725B SoC + from Ingenic. + + This driver can also be built as a module. If so, the module + will be called snd-soc-jz4725b-codec. + config SND_SOC_L3 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d7a59761c08..b07dfb5fa700 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -86,6 +86,7 @@ snd-soc-ics43432-objs := ics43432.o snd-soc-inno-rk3036-objs := inno_rk3036.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o +snd-soc-jz4725b-codec-objs := jz4725b.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o @@ -357,6 +358,7 @@ obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_INNO_RK3036) += snd-soc-inno-rk3036.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o +obj-$(CONFIG_SND_SOC_JZ4725B_CODEC) += snd-soc-jz4725b-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c new file mode 100644 index 000000000000..e3dba92f30bd --- /dev/null +++ b/sound/soc/codecs/jz4725b.c @@ -0,0 +1,599 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * JZ4725B CODEC driver + * + * Copyright (C) 2019, Paul Cercueil <paul(a)crapouillou.net> + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/iopoll.h> +#include <linux/regmap.h> +#include <linux/clk.h> + +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define ICDC_RGADW_OFFSET 0x00 +#define ICDC_RGDATA_OFFSET 0x04 + +/* ICDC internal register access control register(RGADW) */ +#define ICDC_RGADW_RGWR BIT(16) + +#define ICDC_RGADW_RGADDR_OFFSET 8 +#define ICDC_RGADW_RGADDR_MASK GENMASK(14, ICDC_RGADW_RGADDR_OFFSET) + +#define ICDC_RGADW_RGDIN_OFFSET 0 +#define ICDC_RGADW_RGDIN_MASK GENMASK(7, ICDC_RGADW_RGDIN_OFFSET) + +/* ICDC internal register data output register (RGDATA)*/ +#define ICDC_RGDATA_IRQ BIT(8) + +#define ICDC_RGDATA_RGDOUT_OFFSET 0 +#define ICDC_RGDATA_RGDOUT_MASK GENMASK(7, ICDC_RGDATA_RGDOUT_OFFSET) + +/* JZ internal register space */ +enum { + JZ4725B_CODEC_REG_AICR, + JZ4725B_CODEC_REG_CR1, + JZ4725B_CODEC_REG_CR2, + JZ4725B_CODEC_REG_CCR1, + JZ4725B_CODEC_REG_CCR2, + JZ4725B_CODEC_REG_PMR1, + JZ4725B_CODEC_REG_PMR2, + JZ4725B_CODEC_REG_CRR, + JZ4725B_CODEC_REG_ICR, + JZ4725B_CODEC_REG_IFR, + JZ4725B_CODEC_REG_CGR1, + JZ4725B_CODEC_REG_CGR2, + JZ4725B_CODEC_REG_CGR3, + JZ4725B_CODEC_REG_CGR4, + JZ4725B_CODEC_REG_CGR5, + JZ4725B_CODEC_REG_CGR6, + JZ4725B_CODEC_REG_CGR7, + JZ4725B_CODEC_REG_CGR8, + JZ4725B_CODEC_REG_CGR9, + JZ4725B_CODEC_REG_CGR10, + JZ4725B_CODEC_REG_TR1, + JZ4725B_CODEC_REG_TR2, + JZ4725B_CODEC_REG_CR3, + JZ4725B_CODEC_REG_AGC1, + JZ4725B_CODEC_REG_AGC2, + JZ4725B_CODEC_REG_AGC3, + JZ4725B_CODEC_REG_AGC4, + JZ4725B_CODEC_REG_AGC5, +}; + +#define REG_AICR_CONFIG1_OFFSET 0 +#define REG_AICR_CONFIG1_MASK (0xf << REG_AICR_CONFIG1_OFFSET) + +#define REG_CR1_SB_MICBIAS_OFFSET 7 +#define REG_CR1_MONO_OFFSET 6 +#define REG_CR1_DAC_MUTE_OFFSET 5 +#define REG_CR1_HP_DIS_OFFSET 4 +#define REG_CR1_DACSEL_OFFSET 3 +#define REG_CR1_BYPASS_OFFSET 2 + +#define REG_CR2_DAC_DEEMP_OFFSET 7 +#define REG_CR2_DAC_ADWL_OFFSET 5 +#define REG_CR2_DAC_ADWL_MASK (0x3 << REG_CR2_DAC_ADWL_OFFSET) +#define REG_CR2_ADC_ADWL_OFFSET 3 +#define REG_CR2_ADC_ADWL_MASK (0x3 << REG_CR2_ADC_ADWL_OFFSET) +#define REG_CR2_ADC_HPF_OFFSET 2 + +#define REG_CR3_SB_MIC1_OFFSET 7 +#define REG_CR3_SB_MIC2_OFFSET 6 +#define REG_CR3_SIDETONE1_OFFSET 5 +#define REG_CR3_SIDETONE2_OFFSET 4 +#define REG_CR3_MICDIFF_OFFSET 3 +#define REG_CR3_MICSTEREO_OFFSET 2 +#define REG_CR3_INSEL_OFFSET 0 +#define REG_CR3_INSEL_MASK (0x3 << REG_CR3_INSEL_OFFSET) + +#define REG_CCR1_CONFIG4_OFFSET 0 +#define REG_CCR1_CONFIG4_MASK (0xf << REG_CCR1_CONFIG4_OFFSET) + +#define REG_CCR2_DFREQ_OFFSET 4 +#define REG_CCR2_DFREQ_MASK (0xf << REG_CCR2_DFREQ_OFFSET) +#define REG_CCR2_AFREQ_OFFSET 0 +#define REG_CCR2_AFREQ_MASK (0xf << REG_CCR2_AFREQ_OFFSET) + +#define REG_PMR1_SB_DAC_OFFSET 7 +#define REG_PMR1_SB_OUT_OFFSET 6 +#define REG_PMR1_SB_MIX_OFFSET 5 +#define REG_PMR1_SB_ADC_OFFSET 4 +#define REG_PMR1_SB_LIN_OFFSET 3 +#define REG_PMR1_SB_IND_OFFSET 0 + +#define REG_PMR2_LRGI_OFFSET 7 +#define REG_PMR2_RLGI_OFFSET 6 +#define REG_PMR2_LRGOD_OFFSET 5 +#define REG_PMR2_RLGOD_OFFSET 4 +#define REG_PMR2_GIM_OFFSET 3 +#define REG_PMR2_SB_MC_OFFSET 2 +#define REG_PMR2_SB_OFFSET 1 +#define REG_PMR2_SB_SLEEP_OFFSET 0 + +#define REG_IFR_RAMP_UP_DONE_OFFSET 3 +#define REG_IFR_RAMP_DOWN_DONE_OFFSET 2 + +#define REG_CGR1_GODL_OFFSET 4 +#define REG_CGR1_GODL_MASK (0xf << REG_CGR1_GODL_OFFSET) +#define REG_CGR1_GODR_OFFSET 0 +#define REG_CGR1_GODR_MASK (0xf << REG_CGR1_GODR_OFFSET) + +#define REG_CGR2_GO1R_OFFSET 0 +#define REG_CGR2_GO1R_MASK (0x1f << REG_CGR2_GO1R_OFFSET) + +#define REG_CGR3_GO1L_OFFSET 0 +#define REG_CGR3_GO1L_MASK (0x1f << REG_CGR3_GO1L_OFFSET) + +struct jz_icdc { + struct regmap *regmap; + void __iomem *base; + struct clk *clk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_dac_tlv, -2250, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_line_tlv, -1500, 600); + +static const struct snd_kcontrol_new jz4725b_codec_controls[] = { + SOC_DOUBLE_TLV("Master Playback Volume", + JZ4725B_CODEC_REG_CGR1, + REG_CGR1_GODL_OFFSET, + REG_CGR1_GODR_OFFSET, + 0xf, 1, jz4725b_dac_tlv), + SOC_DOUBLE_R_TLV("Master Capture Volume", + JZ4725B_CODEC_REG_CGR3, + JZ4725B_CODEC_REG_CGR2, + REG_CGR2_GO1R_OFFSET, + 0x1f, 1, jz4725b_line_tlv), + + SOC_SINGLE("Master Playback Switch", JZ4725B_CODEC_REG_CR1, + REG_CR1_DAC_MUTE_OFFSET, 1, 1), + + SOC_SINGLE("Deemphatize Filter Playback Switch", + JZ4725B_CODEC_REG_CR2, + REG_CR2_DAC_DEEMP_OFFSET, 1, 0), + + SOC_SINGLE("High-Pass Filter Capture Switch", + JZ4725B_CODEC_REG_CR2, + REG_CR2_ADC_HPF_OFFSET, 1, 0), +}; + +static const char * const jz4725b_codec_adc_src_texts[] = { + "Mic 1", "Mic 2", "Line In", "Mixer", +}; +static const unsigned int jz4725b_codec_adc_src_values[] = { 0, 1, 2, 3, }; +static const SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, + JZ4725B_CODEC_REG_CR3, + REG_CR3_INSEL_OFFSET, + REG_CR3_INSEL_MASK, + jz4725b_codec_adc_src_texts, + jz4725b_codec_adc_src_values); +static const struct snd_kcontrol_new jz4725b_codec_adc_src_ctrl = + SOC_DAPM_ENUM("Route", jz4725b_codec_adc_src_enum); + +static const struct snd_kcontrol_new jz4725b_codec_mixer_controls[] = { + SOC_DAPM_SINGLE("Line In Bypass", JZ4725B_CODEC_REG_CR1, + REG_CR1_BYPASS_OFFSET, 1, 0), +}; + +static int jz4725b_out_stage_enable(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct jz_icdc *icdc = snd_soc_component_get_drvdata(codec); + struct regmap *map = icdc->regmap; + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return regmap_update_bits(map, JZ4725B_CODEC_REG_IFR, + BIT(REG_IFR_RAMP_UP_DONE_OFFSET), 0); + case SND_SOC_DAPM_POST_PMU: + return regmap_read_poll_timeout(map, JZ4725B_CODEC_REG_IFR, + val, val & BIT(REG_IFR_RAMP_UP_DONE_OFFSET), + 100000, 500000); + case SND_SOC_DAPM_PRE_PMD: + return regmap_update_bits(map, JZ4725B_CODEC_REG_IFR, + BIT(REG_IFR_RAMP_DOWN_DONE_OFFSET), 0); + case SND_SOC_DAPM_POST_PMD: + return regmap_read_poll_timeout(map, JZ4725B_CODEC_REG_IFR, + val, val & BIT(REG_IFR_RAMP_DOWN_DONE_OFFSET), + 100000, 500000); + default: + return -EINVAL; + } +} + +static const struct snd_soc_dapm_widget jz4725b_codec_dapm_widgets[] = { + /* DAC */ + SND_SOC_DAPM_DAC("DAC", "Playback", + JZ4725B_CODEC_REG_PMR1, REG_PMR1_SB_DAC_OFFSET, 1), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC", "Capture", + JZ4725B_CODEC_REG_PMR1, REG_PMR1_SB_ADC_OFFSET, 1), + + SND_SOC_DAPM_MUX("ADC Source", SND_SOC_NOPM, 0, 0, + &jz4725b_codec_adc_src_ctrl), + + /* Mixer */ + SND_SOC_DAPM_MIXER("Mixer", JZ4725B_CODEC_REG_PMR1, + REG_PMR1_SB_MIX_OFFSET, 1, + jz4725b_codec_mixer_controls, + ARRAY_SIZE(jz4725b_codec_mixer_controls)), + SND_SOC_DAPM_MIXER("DAC to Mixer", JZ4725B_CODEC_REG_CR1, + REG_CR1_DACSEL_OFFSET, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Line In", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("HP Out", JZ4725B_CODEC_REG_CR1, + REG_CR1_HP_DIS_OFFSET, 1, NULL, 0), + + SND_SOC_DAPM_MIXER("Mic 1", JZ4725B_CODEC_REG_CR3, + REG_CR3_SB_MIC1_OFFSET, 1, NULL, 0), + SND_SOC_DAPM_MIXER("Mic 2", JZ4725B_CODEC_REG_CR3, + REG_CR3_SB_MIC2_OFFSET, 1, NULL, 0), + + SND_SOC_DAPM_MIXER_E("Out Stage", JZ4725B_CODEC_REG_PMR1, + REG_PMR1_SB_OUT_OFFSET, 1, NULL, 0, + jz4725b_out_stage_enable, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("Mixer to ADC", JZ4725B_CODEC_REG_PMR1, + REG_PMR1_SB_IND_OFFSET, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Mic Bias", JZ4725B_CODEC_REG_CR1, + REG_CR1_SB_MICBIAS_OFFSET, 1, NULL, 0), + + /* Pins */ + SND_SOC_DAPM_INPUT("MIC1P"), + SND_SOC_DAPM_INPUT("MIC1N"), + SND_SOC_DAPM_INPUT("MIC2P"), + SND_SOC_DAPM_INPUT("MIC2N"), + + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), +}; + +static const struct snd_soc_dapm_route jz4725b_codec_dapm_routes[] = { + {"Mic 1", NULL, "MIC1P"}, + {"Mic 1", NULL, "MIC1N"}, + {"Mic 2", NULL, "MIC2P"}, + {"Mic 2", NULL, "MIC2N"}, + + {"Line In", NULL, "LLINEIN"}, + {"Line In", NULL, "RLINEIN"}, + + {"Mixer", "Line In Bypass", "Line In"}, + {"DAC to Mixer", NULL, "DAC"}, + {"Mixer", NULL, "DAC to Mixer"}, + + {"Mixer to ADC", NULL, "Mixer"}, + {"ADC Source", "Mixer", "Mixer to ADC"}, + {"ADC Source", "Line In", "Line In"}, + {"ADC Source", "Mic 1", "Mic 1"}, + {"ADC Source", "Mic 2", "Mic 2"}, + {"ADC", NULL, "ADC Source"}, + + {"Out Stage", NULL, "Mixer"}, + {"HP Out", NULL, "Out Stage"}, + {"LHPOUT", NULL, "HP Out"}, + {"RHPOUT", NULL, "HP Out"}, +}; + +static int jz4725b_codec_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct jz_icdc *icdc = snd_soc_component_get_drvdata(component); + struct regmap *map = icdc->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_SLEEP_OFFSET), 0); + break; + case SND_SOC_BIAS_PREPARE: + /* Enable sound hardware */ + regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_OFFSET), 0); + msleep(224); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_SLEEP_OFFSET), + BIT(REG_PMR2_SB_SLEEP_OFFSET)); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(map, JZ4725B_CODEC_REG_PMR2, + BIT(REG_PMR2_SB_OFFSET), + BIT(REG_PMR2_SB_OFFSET)); + break; + } + + return 0; +} + +static int jz4725b_codec_dev_probe(struct snd_soc_component *component) +{ + struct jz_icdc *icdc = snd_soc_component_get_drvdata(component); + struct regmap *map = icdc->regmap; + + clk_prepare_enable(icdc->clk); + + /* Write CONFIGn (n=1 to 8) bits. + * The value 0x0f is specified in the datasheet as a requirement. + */ + regmap_write(map, JZ4725B_CODEC_REG_AICR, + 0xf << REG_AICR_CONFIG1_OFFSET); + regmap_write(map, JZ4725B_CODEC_REG_CCR1, + 0x0 << REG_CCR1_CONFIG4_OFFSET); + + return 0; +} + +static void jz4725b_codec_dev_remove(struct snd_soc_component *component) +{ + struct jz_icdc *icdc = snd_soc_component_get_drvdata(component); + + clk_disable_unprepare(icdc->clk); +} + +static const struct snd_soc_component_driver jz4725b_codec = { + .probe = jz4725b_codec_dev_probe, + .remove = jz4725b_codec_dev_remove, + .set_bias_level = jz4725b_codec_set_bias_level, + .controls = jz4725b_codec_controls, + .num_controls = ARRAY_SIZE(jz4725b_codec_controls), + .dapm_widgets = jz4725b_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(jz4725b_codec_dapm_widgets), + .dapm_routes = jz4725b_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(jz4725b_codec_dapm_routes), + .suspend_bias_off = 1, + .use_pmdown_time = 1, +}; + +static const unsigned int jz4725b_codec_sample_rates[] = { + 96000, 48000, 44100, 32000, + 24000, 22050, 16000, 12000, + 11025, 9600, 8000, +}; + +static int jz4725b_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct jz_icdc *icdc = snd_soc_component_get_drvdata(dai->component); + unsigned int rate, bit_width; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + bit_width = 0; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + bit_width = 1; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + bit_width = 2; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + bit_width = 3; + break; + default: + return -EINVAL; + } + + for (rate = 0; rate < ARRAY_SIZE(jz4725b_codec_sample_rates); rate++) { + if (jz4725b_codec_sample_rates[rate] == params_rate(params)) + break; + } + + if (rate == ARRAY_SIZE(jz4725b_codec_sample_rates)) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(icdc->regmap, + JZ4725B_CODEC_REG_CR2, + REG_CR2_DAC_ADWL_MASK, + bit_width << REG_CR2_DAC_ADWL_OFFSET); + + regmap_update_bits(icdc->regmap, + JZ4725B_CODEC_REG_CCR2, + REG_CCR2_DFREQ_MASK, + rate << REG_CCR2_DFREQ_OFFSET); + } else { + regmap_update_bits(icdc->regmap, + JZ4725B_CODEC_REG_CR2, + REG_CR2_ADC_ADWL_MASK, + bit_width << REG_CR2_ADC_ADWL_OFFSET); + + regmap_update_bits(icdc->regmap, + JZ4725B_CODEC_REG_CCR2, + REG_CCR2_AFREQ_MASK, + rate << REG_CCR2_AFREQ_OFFSET); + } + + return 0; +} + +static const struct snd_soc_dai_ops jz4725b_codec_dai_ops = { + .hw_params = jz4725b_codec_hw_params, +}; + +#define JZ_ICDC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_3LE) + +static struct snd_soc_dai_driver jz4725b_codec_dai = { + .name = "jz4725b-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = JZ_ICDC_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = JZ_ICDC_FORMATS, + }, + .ops = &jz4725b_codec_dai_ops, +}; + +static bool jz4725b_codec_volatile(struct device *dev, unsigned int reg) +{ + return reg == JZ4725B_CODEC_REG_IFR; +} + +static bool jz4725b_codec_can_access_reg(struct device *dev, unsigned int reg) +{ + return (reg != JZ4725B_CODEC_REG_TR1) && (reg != JZ4725B_CODEC_REG_TR2); +} + +static int jz4725b_codec_io_wait(struct jz_icdc *icdc) +{ + u32 reg; + + return readl_poll_timeout(icdc->base + ICDC_RGADW_OFFSET, reg, + !(reg & ICDC_RGADW_RGWR), 1000, 10000); +} + +static int jz4725b_codec_reg_read(void *context, unsigned int reg, + unsigned int *val) +{ + struct jz_icdc *icdc = context; + unsigned int i; + u32 tmp; + int ret; + + ret = jz4725b_codec_io_wait(icdc); + if (ret) + return ret; + + tmp = readl(icdc->base + ICDC_RGADW_OFFSET); + tmp = (tmp & ~ICDC_RGADW_RGADDR_MASK) + | (reg << ICDC_RGADW_RGADDR_OFFSET); + writel(tmp, icdc->base + ICDC_RGADW_OFFSET); + + /* wait 6+ cycles */ + for (i = 0; i < 6; i++) + *val = readl(icdc->base + ICDC_RGDATA_OFFSET) & + ICDC_RGDATA_RGDOUT_MASK; + + return 0; +} + +static int jz4725b_codec_reg_write(void *context, unsigned int reg, + unsigned int val) +{ + struct jz_icdc *icdc = context; + int ret; + + ret = jz4725b_codec_io_wait(icdc); + if (ret) + return ret; + + writel(ICDC_RGADW_RGWR | (reg << ICDC_RGADW_RGADDR_OFFSET) | val, + icdc->base + ICDC_RGADW_OFFSET); + + ret = jz4725b_codec_io_wait(icdc); + if (ret) + return ret; + + return 0; +} + +static const u8 jz4725b_codec_reg_defaults[] = { + 0x0c, 0xaa, 0x78, 0x00, 0x00, 0xff, 0x03, 0x51, + 0x3f, 0x00, 0x00, 0x04, 0x04, 0x04, 0x04, 0x04, + 0x04, 0x0a, 0x0a, 0x00, 0x00, 0x00, 0xc0, 0x34, + 0x07, 0x44, 0x1f, 0x00, +}; + +static const struct regmap_config jz4725b_codec_regmap_config = { + .reg_bits = 7, + .val_bits = 8, + + .max_register = JZ4725B_CODEC_REG_AGC5, + .volatile_reg = jz4725b_codec_volatile, + .readable_reg = jz4725b_codec_can_access_reg, + .writeable_reg = jz4725b_codec_can_access_reg, + + .reg_read = jz4725b_codec_reg_read, + .reg_write = jz4725b_codec_reg_write, + + .reg_defaults_raw = jz4725b_codec_reg_defaults, + .num_reg_defaults_raw = ARRAY_SIZE(jz4725b_codec_reg_defaults), + .cache_type = REGCACHE_FLAT, +}; + +static int jz4725b_codec_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct jz_icdc *icdc; + struct resource *mem; + int ret; + + icdc = devm_kzalloc(dev, sizeof(*icdc), GFP_KERNEL); + if (!icdc) + return -ENOMEM; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + icdc->base = devm_ioremap_resource(dev, mem); + if (IS_ERR(icdc->base)) + return PTR_ERR(icdc->base); + + icdc->regmap = devm_regmap_init(dev, NULL, icdc, + &jz4725b_codec_regmap_config); + if (IS_ERR(icdc->regmap)) + return PTR_ERR(icdc->regmap); + + icdc->clk = devm_clk_get(&pdev->dev, "aic"); + if (IS_ERR(icdc->clk)) + return PTR_ERR(icdc->clk); + + platform_set_drvdata(pdev, icdc); + + ret = devm_snd_soc_register_component(dev, &jz4725b_codec, + &jz4725b_codec_dai, 1); + if (ret) + dev_err(dev, "Failed to register codec\n"); + + return ret; +} + +#ifdef CONFIG_OF +static const struct of_device_id jz4725b_codec_of_matches[] = { + { .compatible = "ingenic,jz4725b-codec", }, + { } +}; +MODULE_DEVICE_TABLE(of, jz4725b_codec_of_matches); +#endif + +static struct platform_driver jz4725b_codec_driver = { + .probe = jz4725b_codec_probe, + .driver = { + .name = "jz4725b-codec", + .of_match_table = of_match_ptr(jz4725b_codec_of_matches), + }, +}; +module_platform_driver(jz4725b_codec_driver); + +MODULE_DESCRIPTION("JZ4725B SoC internal codec driver"); +MODULE_AUTHOR("Paul Cercueil <paul(a)crapouillou.net>"); +MODULE_LICENSE("GPL v2"); -- 2.20.1

1 0

[alsa-devel] [PATCH 1/6] dt-bindings: sound: Document jz4740-codec bindings
by Paul Cercueil 06 Feb '19

06 Feb '19

Add documentation about how to probe the jz4740-codec driver from devicetree. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- .../bindings/sound/ingenic,jz4740-codec.txt | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt diff --git a/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt new file mode 100644 index 000000000000..1ffcade87e7b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt @@ -0,0 +1,20 @@ +Ingenic JZ4740 codec controller + +Required properties: +- compatible : "ingenic,jz4740-codec" +- reg : codec registers location and length +- clocks : phandle to the AIC clock. +- clock-names: must be set to "aic". +- #sound-dai-cells: Must be set to 0. + +Example: + +codec: audio-codec@10020080 { + compatible = "ingenic,jz4740-codec"; + reg = <0x10020080 0x8>; + + #sound-dai-cells = <0>; + + clocks = <&cgu JZ4740_CLK_AIC>; + clock-names = "aic"; +}; -- 2.11.0

2 7

[alsa-devel] [PATCH 0/3] ALSA: Remove superfluous snd_info_register() calls
by Takashi Iwai 06 Feb '19

06 Feb '19

Hi, this is a fallout of the previous patchset, covering more places that call snd_info_register() unnecessarily. These should be applied on top of my previous patchset. The complete patches are found in topic/proc-fixes branch of sound git tree. thanks, Takashi === Takashi Iwai (3): ALSA: firewire: Remove superfluous snd_info_register() calls ALSA: opl4: Remove superfluous snd_info_register() calls ALSA: emux: Remove superfluous snd_info_register() calls sound/drivers/opl4/opl4_proc.c | 4 ---- sound/firewire/bebob/bebob_proc.c | 12 ++---------- sound/firewire/dice/dice-proc.c | 12 ++---------- sound/firewire/digi00x/digi00x-proc.c | 16 ++-------------- sound/firewire/fireface/ff-proc.c | 12 ++---------- sound/firewire/fireworks/fireworks_proc.c | 12 ++---------- sound/firewire/motu/motu-proc.c | 12 ++---------- sound/firewire/oxfw/oxfw-proc.c | 12 ++---------- sound/firewire/tascam/tascam-proc.c | 12 ++---------- sound/synth/emux/emux_proc.c | 4 ---- 10 files changed, 16 insertions(+), 92 deletions(-) -- 2.16.4

3 6

[alsa-devel] [PATCH v3 1/2] ASoC: ssm2602: switch to SPDX identifier
by Marco Felsch 06 Feb '19

06 Feb '19

Drop old license header and switch to SPDX-License-Identifier. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- Changelog: v3: - switch to C++ comment style v2: - new created sound/soc/codecs/ssm2602.c | 39 ++++++++++++-------------------------- 1 file changed, 12 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 501a4e73b185..30b6d30e8e7c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -1,30 +1,15 @@ -/* - * File: sound/soc/codecs/ssm2602.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: Driver for ssm2602 sound chip - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// File: sound/soc/codecs/ssm2602.c +// Author: Cliff Cai <Cliff.Cai(a)analog.com> +// +// Created: Tue June 06 2008 +// Description: Driver for ssm2602 sound chip +// +// Modified: +// Copyright 2008 Analog Devices Inc. +// +// Bugs: Enter bugs at http://blackfin.uclinux.org/ #include <linux/module.h> #include <linux/regmap.h> -- 2.20.1

1 2

[alsa-devel] Applied "ASoC: dapm: Check for NULL widget in dapm_update_dai_unlocked" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: dapm: Check for NULL widget in dapm_update_dai_unlocked has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From cf17a5ffd27234371d10748bf1c716ef172877f3 Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Wed, 6 Feb 2019 11:13:59 +0000 Subject: [PATCH] ASoC: dapm: Check for NULL widget in dapm_update_dai_unlocked DAIs linked to the dummy will not have an associated playback/capture widget, so we need to skip the update in that case. Fixes: 078a85f2806f ("ASoC: dapm: Only power up active channels from a DAI") Reported-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Tested-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Tested-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5b74dffc9c11..111a23a9708a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2580,6 +2580,9 @@ static int dapm_update_dai_unlocked(struct snd_pcm_substream *substream, else w = dai->capture_widget; + if (!w) + return 0; + dev_dbg(dai->dev, "Update DAI routes for %s %s\n", dai->name, dir == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: ssm2602: Fix ADC powerup sequencing" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: ssm2602: Fix ADC powerup sequencing has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 37768e3917405e82611af8e693ccac1a20df38b5 Mon Sep 17 00:00:00 2001 From: Philipp Zabel <p.zabel(a)pengutronix.de> Date: Wed, 6 Feb 2019 16:29:47 +0100 Subject: [PATCH] ASoC: ssm2602: Fix ADC powerup sequencing According to the ssm2603 data sheet (control register sequencing), the digital core should be activated only after all necessary bits in the power register are enabled, and a delay determined by the decoupling capacitor on the VMID pin has passed. If the digital core is activated too early, or even before the ADC is powered up, audible artifacts appear at the beginning of the recorded signal. The digital core is also needed for playback, so when recording starts it may already be enabled. This means we cannot get the power sequence correct when we want to be able to start recording after playback. As a workaround put the MIC mute switch into the DAPM routes. This way we can keep the recording disabled until the MIC Bias has settled and thus get rid of audible artifacts. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> m.felsch(a)pengutronix.de: adapt commit message m.felsch(a)pengutronix.de: drop of configuration as mentioned by Mark: https://patchwork.kernel.org/patch/10407449/ Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ssm2602.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 501a4e73b185..6f79cb0fbc63 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -26,6 +26,7 @@ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA */ +#include <linux/delay.h> #include <linux/module.h> #include <linux/regmap.h> #include <linux/slab.h> @@ -111,7 +112,6 @@ SOC_SINGLE_TLV("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1, SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0), -SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1), }; /* Output Mixer */ @@ -121,10 +121,31 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0), SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0), }; +static const struct snd_kcontrol_new mic_ctl = + SOC_DAPM_SINGLE("Switch", SSM2602_APANA, 1, 1, 1); + /* Input mux */ static const struct snd_kcontrol_new ssm2602_input_mux_controls = SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]); +static int ssm2602_mic_switch_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* + * According to the ssm2603 data sheet (control register sequencing), + * the digital core should be activated only after all necessary bits + * in the power register are enabled, and a delay determined by the + * decoupling capacitor on the VMID pin has passed. If the digital core + * is activated too early, or even before the ADC is powered up, audible + * artifacts appear at the beginning and end of the recorded signal. + * + * In practice, audible artifacts disappear well over 500 ms. + */ + msleep(500); + + return 0; +} + static const struct snd_soc_dapm_widget ssm260x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), @@ -146,6 +167,9 @@ SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1, SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls), SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1), +SND_SOC_DAPM_SWITCH_E("Mic Switch", SSM2602_APANA, 1, 1, &mic_ctl, + ssm2602_mic_switch_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_OUTPUT("LHPOUT"), SND_SOC_DAPM_OUTPUT("RHPOUT"), SND_SOC_DAPM_INPUT("MICIN"), @@ -178,9 +202,11 @@ static const struct snd_soc_dapm_route ssm2602_routes[] = { {"LHPOUT", NULL, "Output Mixer"}, {"Input Mux", "Line", "Line Input"}, - {"Input Mux", "Mic", "Mic Bias"}, + {"Input Mux", "Mic", "Mic Switch"}, {"ADC", NULL, "Input Mux"}, + {"Mic Switch", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "MICIN"}, }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: wcd9335: remove some unnecessary NULL checks" to the asoc tree
by Mark Brown 06 Feb '19

06 Feb '19

The patch ASoC: wcd9335: remove some unnecessary NULL checks has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d22b4117538d42f3dcf7e28210f07f968e46222e Mon Sep 17 00:00:00 2001 From: Dan Carpenter <dan.carpenter(a)oracle.com> Date: Tue, 5 Feb 2019 12:09:27 +0300 Subject: [PATCH] ASoC: wcd9335: remove some unnecessary NULL checks These are arrays, not pointers, and they can't be NULL. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wcd9335.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 3878187bb512..981f88a5f615 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2001,20 +2001,16 @@ static int wcd9335_set_channel_map(struct snd_soc_dai *dai, return -EINVAL; } - if (wcd->rx_chs) { - wcd->num_rx_port = rx_num; - for (i = 0; i < rx_num; i++) { - wcd->rx_chs[i].ch_num = rx_slot[i]; - INIT_LIST_HEAD(&wcd->rx_chs[i].list); - } + wcd->num_rx_port = rx_num; + for (i = 0; i < rx_num; i++) { + wcd->rx_chs[i].ch_num = rx_slot[i]; + INIT_LIST_HEAD(&wcd->rx_chs[i].list); } - if (wcd->tx_chs) { - wcd->num_tx_port = tx_num; - for (i = 0; i < tx_num; i++) { - wcd->tx_chs[i].ch_num = tx_slot[i]; - INIT_LIST_HEAD(&wcd->tx_chs[i].list); - } + wcd->num_tx_port = tx_num; + for (i = 0; i < tx_num; i++) { + wcd->tx_chs[i].ch_num = tx_slot[i]; + INIT_LIST_HEAD(&wcd->tx_chs[i].list); } return 0; -- 2.20.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: compress: Clarify the intent of current compressed ops handling
by Charles Keepax 06 Feb '19

06 Feb '19

For callbacks configuring the state of the components (trigger, set_params, ack and set_metadata) simplify the code a little and make intention clearer by aborting as soon as an error is encountered. The operation has already failed and there is nothing to be gained from processing the callbacks on additional components. The operations currently abort after the callbacks, so this simply shortens the error path. For callbacks returning information from the driver (copy, get_metadata, pointer, get_codec_caps, get_caps and get_params) only look for the first callback provided, currently the code will call every callback only returning the information provided by the last. Since we can only return one set of data, it makes no sense to request the data from every component. Again this just makes the currently supported feature set a little more clear. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-compress.c | 106 +++++++++++++++++++++-------------------------- 1 file changed, 48 insertions(+), 58 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 699397a091670..fc8742383b230 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -353,7 +353,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0, __ret; + int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -364,12 +364,10 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) !component->driver->compr_ops->trigger) continue; - __ret = component->driver->compr_ops->trigger(cstream, cmd); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->trigger(cstream, cmd); + if (ret < 0) + goto out; } - if (ret < 0) - goto out; if (cpu_dai->driver->cops && cpu_dai->driver->cops->trigger) cpu_dai->driver->cops->trigger(cstream, cmd, cpu_dai); @@ -394,7 +392,7 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = fe->cpu_dai; - int ret = 0, __ret, stream; + int ret, stream; if (cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN || cmd == SND_COMPR_TRIGGER_DRAIN) { @@ -406,9 +404,10 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) !component->driver->compr_ops->trigger) continue; - __ret = component->driver->compr_ops->trigger(cstream, cmd); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->trigger(cstream, + cmd); + if (ret < 0) + return ret; } return ret; } @@ -433,12 +432,10 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) !component->driver->compr_ops->trigger) continue; - __ret = component->driver->compr_ops->trigger(cstream, cmd); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->trigger(cstream, cmd); + if (ret < 0) + goto out; } - if (ret < 0) - goto out; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; @@ -472,7 +469,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0, __ret; + int ret; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -496,12 +493,10 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, !component->driver->compr_ops->set_params) continue; - __ret = component->driver->compr_ops->set_params(cstream, params); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->set_params(cstream, params); + if (ret < 0) + goto err; } - if (ret < 0) - goto err; if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) { ret = rtd->dai_link->compr_ops->set_params(cstream); @@ -522,7 +517,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, cancel_delayed_work_sync(&rtd->delayed_work); - return ret; + return 0; err: mutex_unlock(&rtd->pcm_mutex); @@ -538,7 +533,7 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = fe->cpu_dai; - int ret = 0, __ret, stream; + int ret, stream; if (cstream->direction == SND_COMPRESS_PLAYBACK) stream = SNDRV_PCM_STREAM_PLAYBACK; @@ -578,12 +573,10 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, !component->driver->compr_ops->set_params) continue; - __ret = component->driver->compr_ops->set_params(cstream, params); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->set_params(cstream, params); + if (ret < 0) + goto out; } - if (ret < 0) - goto out; if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->set_params) { ret = fe->dai_link->compr_ops->set_params(cstream); @@ -607,7 +600,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0, __ret; + int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -624,9 +617,8 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, !component->driver->compr_ops->get_params) continue; - __ret = component->driver->compr_ops->get_params(cstream, params); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->get_params(cstream, params); + break; } err: @@ -640,7 +632,7 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream, struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; - int ret = 0, __ret; + int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -651,9 +643,8 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream, !component->driver->compr_ops->get_caps) continue; - __ret = component->driver->compr_ops->get_caps(cstream, caps); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->get_caps(cstream, caps); + break; } mutex_unlock(&rtd->pcm_mutex); @@ -666,7 +657,7 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; - int ret = 0, __ret; + int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -677,9 +668,9 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, !component->driver->compr_ops->get_codec_caps) continue; - __ret = component->driver->compr_ops->get_codec_caps(cstream, codec); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->get_codec_caps(cstream, + codec); + break; } mutex_unlock(&rtd->pcm_mutex); @@ -692,7 +683,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0, __ret; + int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -709,9 +700,9 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) !component->driver->compr_ops->ack) continue; - __ret = component->driver->compr_ops->ack(cstream, bytes); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->ack(cstream, bytes); + if (ret < 0) + goto err; } err: @@ -725,7 +716,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; - int ret = 0, __ret; + int ret = 0; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -740,9 +731,8 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, !component->driver->compr_ops->pointer) continue; - __ret = component->driver->compr_ops->pointer(cstream, tstamp); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->pointer(cstream, tstamp); + break; } mutex_unlock(&rtd->pcm_mutex); @@ -781,7 +771,7 @@ static int soc_compr_set_metadata(struct snd_compr_stream *cstream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0, __ret; + int ret; if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_metadata) { ret = cpu_dai->driver->cops->set_metadata(cstream, metadata, cpu_dai); @@ -796,12 +786,13 @@ static int soc_compr_set_metadata(struct snd_compr_stream *cstream, !component->driver->compr_ops->set_metadata) continue; - __ret = component->driver->compr_ops->set_metadata(cstream, metadata); - if (__ret < 0) - ret = __ret; + ret = component->driver->compr_ops->set_metadata(cstream, + metadata); + if (ret < 0) + return ret; } - return ret; + return 0; } static int soc_compr_get_metadata(struct snd_compr_stream *cstream, @@ -811,7 +802,7 @@ static int soc_compr_get_metadata(struct snd_compr_stream *cstream, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0, __ret; + int ret; if (cpu_dai->driver->cops && cpu_dai->driver->cops->get_metadata) { ret = cpu_dai->driver->cops->get_metadata(cstream, metadata, cpu_dai); @@ -826,12 +817,11 @@ static int soc_compr_get_metadata(struct snd_compr_stream *cstream, !component->driver->compr_ops->get_metadata) continue; - __ret = component->driver->compr_ops->get_metadata(cstream, metadata); - if (__ret < 0) - ret = __ret; + return component->driver->compr_ops->get_metadata(cstream, + metadata); } - return ret; + return 0; } /* ASoC Compress operations */ -- 2.11.0

2 3

[alsa-devel] [PATCH v2 1/2] ASoC: ssm2602: switch to SPDX identifier
by Marco Felsch 06 Feb '19

06 Feb '19

Drop old license header and switch to SPDX-License-Identifier. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- Changelog: v2: - new created sound/soc/codecs/ssm2602.c | 16 +--------------- 1 file changed, 1 insertion(+), 15 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 501a4e73b185..580ab8916854 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -1,3 +1,4 @@ +// SPDX-License-Identifier: GPL-2.0-or-later /* * File: sound/soc/codecs/ssm2602.c * Author: Cliff Cai <Cliff.Cai(a)analog.com> @@ -9,21 +10,6 @@ * Copyright 2008 Analog Devices Inc. * * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA */ #include <linux/module.h> -- 2.20.1

3 5

[alsa-devel] [BUG BISECT] NULL pointer after commit "ASoC: dapm: Only power up active channels from a DAI"
by Krzysztof Kozlowski 06 Feb '19

06 Feb '19

Hi, Bisect pointed to commit: commit 078a85f2806f0ffd11289009462a6a390f9adb5c Author: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Thu Jan 31 13:30:18 2019 +0000 ASoC: dapm: Only power up active channels from a DAI as a bad commit for NULL pointer on my Odroid XU3 and Odroid U3 board when doing "aplay /usr/share/sounds/alsa/Front_Right.wav". 1. Arch ARM Linux 2. exynos_defconfig 3. Odroid U3, XU3, Exynos SoC, ARMv7 Last address is in calltrace: c079552c dapm_update_dai_unlocked sound/soc/soc-dapm.c:2586 Full log: Unable to handle kernel NULL pointer dereference at virtual address 0000007c pgd = d5b32ef7 [0000007c] *pgd=00000000 Internal error: Oops: 5 [#1] PREEMPT SMP ARM Modules linked in: s5p_mfc s5p_jpeg exynos_gsc v4l2_mem2mem v4l2_common videobuf2_dma_contig videobuf2_memops videobuf2_v4l2 videobuf2_common videodev media s5p_cec CPU: 7 PID: 359 Comm: aplay Not tainted 5.0.0-rc4-00177-g078a85f2806f #32 Hardware name: SAMSUNG EXYNOS (Flattened Device Tree) PC is at dapm_update_dai_unlocked+0x38/0x138 LR is at snd_soc_dapm_update_dai+0x34/0x4c pc : [<c079552c>] lr : [<c0795660>] psr: 60010013 sp : eb9d9be8 ip : eabc8b64 fp : ecce0e10 r10: c100b448 r9 : ecce0e0c r8 : 00000000 r7 : 00000000 r6 : 00000000 r5 : 00000002 r4 : ecce0800 r3 : 00000000 r2 : eccaa080 r1 : ecdab000 r0 : 00000000 Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none Control: 10c5387d Table: 6c09806a DAC: 00000051 Process aplay (pid: 359, stack limit = 0x4ae0a814) Stack: (0xeb9d9be8 to 0xeb9da000) 9be0: ecce0e0c c100b448 ecce0e10 ecce0800 ecd64400 ecdab000 9c00: eccaa080 c0795660 ee3c480c ecd64400 ecce0800 ecdab000 00000000 c079d760 9c20: eccaa080 ecce080c 00000000 00000001 00000000 00000000 00000000 00000000 9c40: 00000000 00000000 00000000 00000004 00000000 00000000 00000000 00000000 9c60: 00000000 00000000 00000000 00000001 00000000 00000000 00000000 00000000 9c80: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9ca0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9cc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9ce0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9d00: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9d20: 00000000 00000000 00000000 00000010 00000010 00000004 00000020 00000020 9d40: 00000004 00000002 00000002 00000004 0000bb80 0000bb80 00000004 0001e848 9d60: 0001e848 00000004 00001770 00001770 00000004 00005dc0 00005dc0 00000004 9d80: 00000004 00000004 00000004 0007a120 0007a120 00000004 00005dc0 00005dc0 9da0: 00000004 00017700 00017700 00000004 00000000 00000000 00000004 00000000 9dc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9de0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9e00: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9e20: 00000000 00000000 00000000 000fff07 00200103 00000010 0000bb80 00000001 9e40: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9e60: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 9e80: 00000000 8c6972bb ecdab000 ecd64400 ecdab000 00000000 c0b97b1c ebfe8000 9ea0: 00000000 bea4eea0 b6effb88 c07825e4 c100b448 ecdab000 ecd64400 00000000 9ec0: ecfdd180 ec1bde40 bea4eea0 c07836e0 eb9d9f78 00000000 00000000 00000001 9ee0: 61746164 c0294d70 00000001 8c6972bb c100b448 bea4eea0 ecfdd180 c02a9bac 9f00: bea4eea0 eb9d8000 ed4aafa8 c02a9244 eb9cbb80 c100b448 00000000 00000000 9f20: 00000000 00000000 00000000 8c6972bb eb9d9fb0 ec6f11c0 eb9cbb80 00000002 9f40: eb9cbb88 00000001 b6e85b6f 00000001 61746164 c0295028 00000000 00000000 9f60: c100b448 8c6972bb 00000000 ecfdd180 ecfdd180 00000004 c25c4111 bea4eea0 9f80: eb9d8000 00000036 b6effb88 c02a9bac 004cf1b8 bea4eea0 004cf208 00000036 9fa0: c01011c4 c0101000 004cf1b8 bea4eea0 00000004 c25c4111 bea4eea0 00020001 9fc0: 004cf1b8 bea4eea0 004cf208 00000036 bea4ee10 00000000 b6fb5900 b6effb88 9fe0: b6f6c82c bea4edf4 b6efbfc4 b6e0bbdc 200d0010 00000004 00000000 00000000 [<c079552c>] (dapm_update_dai_unlocked) from [<c0795660>] (snd_soc_dapm_update_dai+0x34/0x4c) [<c0795660>] (snd_soc_dapm_update_dai) from [<c079d760>] (soc_pcm_hw_params+0x22c/0x624) [<c079d760>] (soc_pcm_hw_params) from [<c07825e4>] (snd_pcm_hw_params+0x128/0x378) [<c07825e4>] (snd_pcm_hw_params) from [<c07836e0>] (snd_pcm_ioctl+0x26c/0x1258) [<c07836e0>] (snd_pcm_ioctl) from [<c02a9244>] (do_vfs_ioctl+0xb0/0x9e4) [<c02a9244>] (do_vfs_ioctl) from [<c02a9bac>] (ksys_ioctl+0x34/0x58) [<c02a9bac>] (ksys_ioctl) from [<c0101000>] (ret_fast_syscall+0x0/0x28) Exception stack(0xeb9d9fa8 to 0xeb9d9ff0) 9fa0: 004cf1b8 bea4eea0 00000004 c25c4111 bea4eea0 00020001 9fc0: 004cf1b8 bea4eea0 004cf208 00000036 bea4ee10 00000000 b6fb5900 b6effb88 9fe0: b6f6c82c bea4edf4 b6efbfc4 b6e0bbdc Code: e3530000 e5926020 1a00003a e1a07006 (e5b7407c) ---[ end trace 8561600efcd2b2f1 ]--- Let me know if you need any other data. Best regards, Krzysztof

3 11

[alsa-devel] [PATCH 00/20] ALSA: Drop superfluous PCM preallocation error checks
by Takashi Iwai 06 Feb '19

06 Feb '19

Hi, this is yet another patchset for trivial cleanups. At this time, it's about the superfluous error check from snd_pcm_lib_preallocate_pages() and co. They always succeed and never return a failure although many codes do the error check. The patches simply remove the error checks and finally change the function to be void function. The patches are applied on top of my previous NULL device fix patches, so not applicable to ASoC tree, I guess. As all these are trivial, applying through my tree altogether would be simpler. So, Mark, please give your ACK if the changes to ASoC drivers are OK. thanks, Takashi === Takashi Iwai (20): ALSA: atmel: Drop superfluous PCM preallocation error checks ALSA: parisc: Drop superfluous PCM preallocation error checks ALSA: pci: Drop superfluous PCM preallocation error checks ALSA: ppc: Drop superfluous PCM preallocation error checks ALSA: sh: Drop superfluous PCM preallocation error checks ALSA: sparc: Drop superfluous PCM preallocation error checks ALSA: spi: Drop superfluous PCM preallocation error checks ALSA: usb: Drop superfluous PCM preallocation error checks ASoC: amd: Drop superfluous PCM preallocation error checks ASoC: dwc: Drop superfluous PCM preallocation error checks ASoC: intel: Drop superfluous PCM preallocation error checks ASoC: mediatek: Drop superfluous PCM preallocation error checks ASoC: meson: Drop superfluous PCM preallocation error checks ASoC: sh: Drop superfluous PCM preallocation error checks ASoC: stm: Drop superfluous PCM preallocation error checks ASoC: txx9: Drop superfluous PCM preallocation error checks ASoC: uniphier: Drop superfluous PCM preallocation error checks ASoC: xtensa: Drop superfluous PCM preallocation error checks ASoC: dmaengine: Drop superfluous PCM preallocation error checks ALSA: pcm: Define snd_pcm_lib_preallocate_*() as returning void include/sound/pcm.h | 8 ++--- sound/atmel/ac97c.c | 4 +-- sound/core/pcm_memory.c | 29 +++++----------- sound/parisc/harmony.c | 10 ++---- sound/pci/ad1889.c | 7 +--- sound/pci/aw2/aw2-alsa.c | 40 +++++++--------------- sound/pci/bt87x.c | 10 +++--- sound/pci/ca0106/ca0106_main.c | 16 ++++----- sound/pci/echoaudio/echoaudio.c | 16 ++++----- sound/pci/emu10k1/emupcm.c | 22 ++++++++---- sound/pci/emu10k1/p16v.c | 17 ++++----- sound/pci/lx6464es/lx6464es.c | 8 ++--- sound/pci/rme9652/hdspm.c | 20 +++-------- sound/pci/via82xx_modem.c | 8 ++--- sound/ppc/snd_ps3.c | 6 +--- sound/sh/aica.c | 14 ++++---- sound/soc/amd/acp-pcm-dma.c | 26 ++++++-------- sound/soc/amd/raven/acp3x-pcm-dma.c | 9 +++-- sound/soc/dwc/dwc-pcm.c | 3 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 9 ++--- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 15 +++----- sound/soc/intel/haswell/sst-haswell-pcm.c | 10 ++---- sound/soc/intel/skylake/skl-pcm.c | 9 ++--- .../soc/mediatek/common/mtk-afe-platform-driver.c | 6 ++-- sound/soc/meson/axg-fifo.c | 7 ++-- sound/soc/sh/fsi.c | 3 +- sound/soc/sh/rcar/core.c | 5 +-- sound/soc/sh/siu_pcm.c | 13 +------ sound/soc/soc-generic-dmaengine-pcm.c | 5 +-- sound/soc/stm/stm32_adfsdm.c | 5 +-- sound/soc/txx9/txx9aclc.c | 4 ++- sound/soc/uniphier/aio-dma.c | 3 +- sound/soc/xtensa/xtfpga-i2s.c | 6 ++-- sound/sparc/dbri.c | 9 ++--- sound/spi/at73c213.c | 2 +- sound/usb/usx2y/usbusx2yaudio.c | 21 ++++++------ sound/usb/usx2y/usx2yhwdeppcm.c | 19 +++++----- 37 files changed, 157 insertions(+), 267 deletions(-) -- 2.16.4

4 25

[alsa-devel] [PATCH] ASoC: dapm: Check for NULL widget in dapm_update_dai_unlocked
by Charles Keepax 06 Feb '19

06 Feb '19

DAIs linked to the dummy will not have an associated playback/capture widget, so we need to skip the update in that case. Fixes: 078a85f2806f ("ASoC: dapm: Only power up active channels from a DAI") Reported-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Tested-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Tested-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5b74dffc9c11d..111a23a9708a2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2580,6 +2580,9 @@ static int dapm_update_dai_unlocked(struct snd_pcm_substream *substream, else w = dai->capture_widget; + if (!w) + return 0; + dev_dbg(dai->dev, "Update DAI routes for %s %s\n", dai->name, dir == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); -- 2.11.0

1 0

[alsa-devel] [sound:topic/memory-device-fixes 83/83] drivers/media/pci/solo6x10/solo6x10-g723.c:363:6: error: void value not ignored as it ought to be
by kbuild test robot 06 Feb '19

06 Feb '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/memory-device-fixes head: c898c3496b8e5c583e874bbfbbcba3d185cac051 commit: c898c3496b8e5c583e874bbfbbcba3d185cac051 [83/83] ALSA: pcm: Define snd_pcm_lib_preallocate_*() as returning void config: x86_64-allmodconfig (attached as .config) compiler: gcc-8 (Debian 8.2.0-14) 8.2.0 reproduce: git checkout c898c3496b8e5c583e874bbfbbcba3d185cac051 # save the attached .config to linux build tree make ARCH=x86_64 All error/warnings (new ones prefixed by >>): >> drivers/media/pci/solo6x10/solo6x10-g723.c:363:13: sparse: warning: incorrect type in assignment (different base types) drivers/media/pci/solo6x10/solo6x10-g723.c:363:13: sparse: expected int [assigned] ret drivers/media/pci/solo6x10/solo6x10-g723.c:363:13: sparse: got void -- drivers/media/pci/solo6x10/solo6x10-g723.c: In function 'solo_snd_pcm_init': >> drivers/media/pci/solo6x10/solo6x10-g723.c:363:6: error: void value not ignored as it ought to be ret = snd_pcm_lib_preallocate_pages_for_all(pcm, ^ -- >> drivers/media/pci/tw686x/tw686x-audio.c:304:53: sparse: warning: incorrect type in return expression (different base types) drivers/media/pci/tw686x/tw686x-audio.c:304:53: sparse: expected int drivers/media/pci/tw686x/tw686x-audio.c:304:53: sparse: got void -- drivers/media/pci/tw686x/tw686x-audio.c: In function 'tw686x_snd_pcm_init': >> drivers/media/pci/tw686x/tw686x-audio.c:304:9: error: void value not ignored as it ought to be return snd_pcm_lib_preallocate_pages_for_all(pcm, ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ SNDRV_DMA_TYPE_DEV, ~~~~~~~~~~~~~~~~~~~ snd_dma_pci_data(dev->pci_dev), ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ TW686X_AUDIO_PAGE_MAX * AUDIO_DMA_SIZE_MAX, ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ TW686X_AUDIO_PAGE_MAX * AUDIO_DMA_SIZE_MAX); ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ >> drivers/media/pci/tw686x/tw686x-audio.c:309:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ sparse warnings: (new ones prefixed by >>) drivers/media/pci/solo6x10/solo6x10-g723.c:363:13: sparse: warning: incorrect type in assignment (different base types) >> drivers/media/pci/solo6x10/solo6x10-g723.c:363:13: sparse: expected int [assigned] ret >> drivers/media/pci/solo6x10/solo6x10-g723.c:363:13: sparse: got void -- drivers/media/pci/tw686x/tw686x-audio.c:304:53: sparse: warning: incorrect type in return expression (different base types) >> drivers/media/pci/tw686x/tw686x-audio.c:304:53: sparse: expected int >> drivers/media/pci/tw686x/tw686x-audio.c:304:53: sparse: got void vim +363 drivers/media/pci/solo6x10/solo6x10-g723.c faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 338 decebabf drivers/staging/solo6x10/g723.c Krzysztof Hałasa 2011-02-11 339 static int solo_snd_pcm_init(struct solo_dev *solo_dev) faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 340 { faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 341 struct snd_card *card = solo_dev->snd_card; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 342 struct snd_pcm *pcm; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 343 struct snd_pcm_substream *ss; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 344 int ret; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 345 int i; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 346 faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 347 ret = snd_pcm_new(card, card->driver, 0, 0, solo_dev->nr_chans, faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 348 &pcm); faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 349 if (ret < 0) faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 350 return ret; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 351 faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 352 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 353 &snd_solo_pcm_ops); faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 354 faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 355 snd_pcm_chip(pcm) = solo_dev; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 356 pcm->info_flags = 0; cc1e6315 drivers/media/pci/solo6x10/solo6x10-g723.c Mauro Carvalho Chehab 2018-09-10 357 strscpy(pcm->name, card->shortname, sizeof(pcm->name)); faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 358 faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 359 for (i = 0, ss = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 360 ss; ss = ss->next, i++) faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 361 sprintf(ss->name, "Camera #%d Audio", i); faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 362 faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 @363 ret = snd_pcm_lib_preallocate_pages_for_all(pcm, faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 364 SNDRV_DMA_TYPE_CONTINUOUS, faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 365 snd_dma_continuous_data(GFP_KERNEL), 9d1b1f61 drivers/media/pci/solo6x10/solo6x10-g723.c Krzysztof Hałasa 2015-06-08 366 G723_PERIOD_BYTES * PERIODS, 9d1b1f61 drivers/media/pci/solo6x10/solo6x10-g723.c Krzysztof Hałasa 2015-06-08 367 G723_PERIOD_BYTES * PERIODS); faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 368 if (ret < 0) faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 369 return ret; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 370 faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 371 solo_dev->snd_pcm = pcm; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 372 faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 373 return 0; faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 374 } faa4fd2a drivers/staging/solo6x10/solo6010-g723.c Ben Collins 2010-06-17 375 :::::: The code at line 363 was first introduced by commit :::::: faa4fd2a09517b39cc1f5d622453f97a59acfdac Staging: solo6x10: New driver (staging) for Softlogic 6x10 :::::: TO: Ben Collins <bcollins(a)bluecherry.net> :::::: CC: Greg Kroah-Hartman <gregkh(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v10 0/4] Media Device Allocator API
by Shuah Khan 06 Feb '19

06 Feb '19

Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - This patch series is tested on 5.0-rc3 and addresses comments on v9 series from Hans Verkuil. - v9 was tested on 4.20-rc6. - Tested sharing resources with kaffeine, vlc, xawtv, tvtime, and arecord. When analog is streaming, digital and audio user-space applications detect that the tuner is busy and exit. When digital is streaming, analog and audio applications detect that the tuner is busy and exit. When arecord is owns the tuner, digital and analog detect that the tuner is busy and exit. - Tested media device allocator API with bind/unbind testing on snd-usb-audio and au0828 drivers to make sure /dev/mediaX is released only when the last driver is unbound. - Addressed review comments from Hans on the RFC v8 (rebased on 4.19) - Updated change log to describe the use-case more clearly. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. Changes since v9: - Patch 1: Fix mutex assert warning from find_module() calls. This code was written before the change to find_module() that requires callers to hold module_mutex. I missed this during my testing on 4.20-rc6. Hans Verkuil reported the problem. - Patch 4: sound/usb: Initializes all the entities it can before registering the device based on comments from Hans Verkuil - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - No changes to Patches 2 and 3. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (4): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 41 ++++ drivers/media/Makefile | 4 + drivers/media/media-dev-allocator.c | 144 +++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 327 +++++++++++++++++++++++++ sound/usb/media.h | 74 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 723 insertions(+), 22 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.1

4 17

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Fix build error without CONFIG_PCI
by Takashi Iwai 06 Feb '19

06 Feb '19

A call of pci_iounmap() call without CONFIG_PCI leads to a build error on some architectures. We tried to address this and add a check of IS_ENABLED(CONFIG_PCI), but this still doesn't seem enough for sh. Ideally we should fix it globally, it's really a corner case, so let's paper over it with a simpler ifdef. Fixes: 1e73359a24fa ("ALSA: hda/ca0132 - make pci_iounmap() call conditional") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e5bdbc245682..29882bda7632 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -8451,8 +8451,10 @@ static void ca0132_free(struct hda_codec *codec) ca0132_exit_chip(codec); snd_hda_power_down(codec); - if (IS_ENABLED(CONFIG_PCI) && spec->mem_base) +#ifdef CONFIG_PCI + if (spec->mem_base) pci_iounmap(codec->bus->pci, spec->mem_base); +#endif kfree(spec->spec_init_verbs); kfree(codec->spec); } -- 2.16.4

2 2

[alsa-devel] [sound:topic/proc-fixes 71/82] htmldocs: sound/core/info.c:886: warning: Function parameter or member 'write' not described in 'snd_card_rw_proc_new'
by kbuild test robot 05 Feb '19

05 Feb '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/proc-fixes head: 0f66ae449f2ec8e3f4ca30334bb60ce93a3d9822 commit: 2c0283ac599c343937679b178a00ecc453c6899a [71/82] ALSA: info: Add standard helpers for card proc file entries reproduce: make htmldocs All warnings (new ones prefixed by >>): WARNING: convert(1) not found, for SVG to PDF conversion install ImageMagick (https://www.imagemagick.org) include/linux/interrupt.h:268: warning: Function parameter or member 'is_managed' not described in 'irq_affinity_desc' block/blk-core.c:685: warning: Excess function parameter 'request_count' description in 'blk_attempt_plug_merge' block/blk-core.c:685: warning: Excess function parameter 'request_count' description in 'blk_attempt_plug_merge' include/linux/rcupdate_wait.h:1: warning: no structured comments found include/linux/rcutree.h:1: warning: no structured comments found kernel/rcu/tree.c:710: warning: Excess function parameter 'irq' description in 'rcu_nmi_exit' include/linux/gfp.h:1: warning: no structured comments found include/net/cfg80211.h:4687: warning: Function parameter or member 'wext.ibss' not described in 'wireless_dev' include/net/cfg80211.h:4687: warning: Function parameter or member 'wext.connect' not described in 'wireless_dev' include/net/cfg80211.h:4687: warning: Function parameter or member 'wext.keys' not described in 'wireless_dev' include/net/cfg80211.h:4687: warning: Function parameter or member 'wext.ie' not described in 'wireless_dev' include/net/cfg80211.h:4687: warning: Function parameter or member 'wext.ie_len' not described in 'wireless_dev' include/net/cfg80211.h:4687: warning: Function parameter or member 'wext.bssid' not described in 'wireless_dev' include/net/cfg80211.h:4687: warning: Function parameter or member 'wext.ssid' not described in 'wireless_dev' include/net/cfg80211.h:4687: warning: Function parameter or member 'wext.default_key' not described in 'wireless_dev' include/net/cfg80211.h:4687: warning: Function parameter or member 'wext.default_mgmt_key' not described in 'wireless_dev' include/net/cfg80211.h:4687: warning: Function parameter or member 'wext.prev_bssid_valid' not described in 'wireless_dev' include/net/mac80211.h:2393: warning: Function parameter or member 'radiotap_timestamp.units_pos' not described in 'ieee80211_hw' include/net/mac80211.h:2393: warning: Function parameter or member 'radiotap_timestamp.accuracy' not described in 'ieee80211_hw' include/net/mac80211.h:1004: warning: Function parameter or member 'control.rates' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'control.rts_cts_rate_idx' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'control.use_rts' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'control.use_cts_prot' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'control.short_preamble' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'control.skip_table' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'control.jiffies' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'control.vif' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'control.hw_key' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'control.flags' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'control.enqueue_time' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'ack' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'ack.cookie' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'status.rates' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'status.ack_signal' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'status.ampdu_ack_len' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'status.ampdu_len' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'status.antenna' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'status.tx_time' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'status.is_valid_ack_signal' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'status.status_driver_data' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'driver_rates' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'pad' not described in 'ieee80211_tx_info' include/net/mac80211.h:1004: warning: Function parameter or member 'rate_driver_data' not described in 'ieee80211_tx_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'rx_stats_avg' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'rx_stats_avg.signal' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'rx_stats_avg.chain_signal' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.filtered' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.retry_failed' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.retry_count' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.lost_packets' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.last_tdls_pkt_time' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.msdu_retries' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.msdu_failed' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.last_ack' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.last_ack_signal' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.ack_signal_filled' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'status_stats.avg_ack_signal' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'tx_stats.packets' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'tx_stats.bytes' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'tx_stats.last_rate' not described in 'sta_info' net/mac80211/sta_info.h:590: warning: Function parameter or member 'tx_stats.msdu' not described in 'sta_info' kernel/rcu/tree.c:711: warning: Excess function parameter 'irq' description in 'rcu_nmi_exit' include/linux/dma-buf.h:304: warning: Function parameter or member 'cb_excl.cb' not described in 'dma_buf' include/linux/dma-buf.h:304: warning: Function parameter or member 'cb_excl.poll' not described in 'dma_buf' include/linux/dma-buf.h:304: warning: Function parameter or member 'cb_excl.active' not described in 'dma_buf' include/linux/dma-buf.h:304: warning: Function parameter or member 'cb_shared.cb' not described in 'dma_buf' include/linux/dma-buf.h:304: warning: Function parameter or member 'cb_shared.poll' not described in 'dma_buf' include/linux/dma-buf.h:304: warning: Function parameter or member 'cb_shared.active' not described in 'dma_buf' include/linux/dma-fence-array.h:54: warning: Function parameter or member 'work' not described in 'dma_fence_array' include/linux/firmware/intel/stratix10-svc-client.h:1: warning: no structured comments found include/linux/gpio/driver.h:371: warning: Function parameter or member 'init_valid_mask' not described in 'gpio_chip' include/linux/iio/hw-consumer.h:1: warning: no structured comments found include/linux/input/sparse-keymap.h:46: warning: Function parameter or member 'sw' not described in 'key_entry' drivers/mtd/nand/raw/nand_base.c:420: warning: Function parameter or member 'chip' not described in 'nand_fill_oob' drivers/mtd/nand/raw/nand_bbt.c:173: warning: Function parameter or member 'this' not described in 'read_bbt' drivers/mtd/nand/raw/nand_bbt.c:173: warning: Excess function parameter 'chip' description in 'read_bbt' include/linux/regulator/machine.h:199: warning: Function parameter or member 'max_uV_step' not described in 'regulation_constraints' include/linux/regulator/driver.h:228: warning: Function parameter or member 'resume' not described in 'regulator_ops' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw0' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw1' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw2' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.esw3' not described in 'irb' arch/s390/include/asm/cio.h:245: warning: Function parameter or member 'esw.eadm' not described in 'irb' drivers/slimbus/stream.c:1: warning: no structured comments found >> sound/core/info.c:886: warning: Function parameter or member 'write' not described in 'snd_card_rw_proc_new' include/linux/spi/spi.h:180: warning: Function parameter or member 'driver_override' not described in 'spi_device' drivers/target/target_core_device.c:1: warning: no structured comments found drivers/usb/typec/bus.c:1: warning: no structured comments found drivers/usb/typec/class.c:1: warning: no structured comments found include/linux/w1.h:281: warning: Function parameter or member 'of_match_table' not described in 'w1_family' fs/direct-io.c:257: warning: Excess function parameter 'offset' description in 'dio_complete' fs/file_table.c:1: warning: no structured comments found fs/libfs.c:477: warning: Excess function parameter 'available' description in 'simple_write_end' fs/posix_acl.c:646: warning: Function parameter or member 'inode' not described in 'posix_acl_update_mode' fs/posix_acl.c:646: warning: Function parameter or member 'mode_p' not described in 'posix_acl_update_mode' fs/posix_acl.c:646: warning: Function parameter or member 'acl' not described in 'posix_acl_update_mode' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:294: warning: Excess function parameter 'mm' description in 'amdgpu_mn_invalidate_range_start_hsa' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:294: warning: Excess function parameter 'start' description in 'amdgpu_mn_invalidate_range_start_hsa' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:294: warning: Excess function parameter 'end' description in 'amdgpu_mn_invalidate_range_start_hsa' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:343: warning: Excess function parameter 'mm' description in 'amdgpu_mn_invalidate_range_end' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:343: warning: Excess function parameter 'start' description in 'amdgpu_mn_invalidate_range_end' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:343: warning: Excess function parameter 'end' description in 'amdgpu_mn_invalidate_range_end' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:183: warning: Function parameter or member 'blockable' not described in 'amdgpu_mn_read_lock' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:295: warning: Function parameter or member 'range' not described in 'amdgpu_mn_invalidate_range_start_hsa' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:295: warning: Excess function parameter 'mm' description in 'amdgpu_mn_invalidate_range_start_hsa' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:295: warning: Excess function parameter 'start' description in 'amdgpu_mn_invalidate_range_start_hsa' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:295: warning: Excess function parameter 'end' description in 'amdgpu_mn_invalidate_range_start_hsa' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:344: warning: Function parameter or member 'range' not described in 'amdgpu_mn_invalidate_range_end' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:344: warning: Excess function parameter 'mm' description in 'amdgpu_mn_invalidate_range_end' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:344: warning: Excess function parameter 'start' description in 'amdgpu_mn_invalidate_range_end' drivers/gpu/drm/amd/amdgpu/amdgpu_mn.c:344: warning: Excess function parameter 'end' description in 'amdgpu_mn_invalidate_range_end' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:382: warning: cannot understand function prototype: 'struct amdgpu_vm_pt_cursor ' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:383: warning: cannot understand function prototype: 'struct amdgpu_vm_pt_cursor ' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:555: warning: Function parameter or member 'adev' not described in 'for_each_amdgpu_vm_pt_leaf' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:555: warning: Function parameter or member 'vm' not described in 'for_each_amdgpu_vm_pt_leaf' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:555: warning: Function parameter or member 'start' not described in 'for_each_amdgpu_vm_pt_leaf' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:555: warning: Function parameter or member 'end' not described in 'for_each_amdgpu_vm_pt_leaf' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:555: warning: Function parameter or member 'cursor' not described in 'for_each_amdgpu_vm_pt_leaf' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:603: warning: Function parameter or member 'adev' not described in 'for_each_amdgpu_vm_pt_dfs_safe' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:603: warning: Function parameter or member 'vm' not described in 'for_each_amdgpu_vm_pt_dfs_safe' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:603: warning: Function parameter or member 'cursor' not described in 'for_each_amdgpu_vm_pt_dfs_safe' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:603: warning: Function parameter or member 'entry' not described in 'for_each_amdgpu_vm_pt_dfs_safe' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:845: warning: Function parameter or member 'level' not described in 'amdgpu_vm_bo_param' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1353: warning: Function parameter or member 'params' not described in 'amdgpu_vm_update_func' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1353: warning: Function parameter or member 'bo' not described in 'amdgpu_vm_update_func' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1353: warning: Function parameter or member 'pe' not described in 'amdgpu_vm_update_func' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1353: warning: Function parameter or member 'addr' not described in 'amdgpu_vm_update_func' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1353: warning: Function parameter or member 'count' not described in 'amdgpu_vm_update_func' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1353: warning: Function parameter or member 'incr' not described in 'amdgpu_vm_update_func' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1353: warning: Function parameter or member 'flags' not described in 'amdgpu_vm_update_func' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1520: warning: Function parameter or member 'params' not described in 'amdgpu_vm_update_huge' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1520: warning: Function parameter or member 'bo' not described in 'amdgpu_vm_update_huge' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1520: warning: Function parameter or member 'level' not described in 'amdgpu_vm_update_huge' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1520: warning: Function parameter or member 'pe' not described in 'amdgpu_vm_update_huge' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1520: warning: Function parameter or member 'addr' not described in 'amdgpu_vm_update_huge' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1520: warning: Function parameter or member 'count' not described in 'amdgpu_vm_update_huge' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1520: warning: Function parameter or member 'incr' not described in 'amdgpu_vm_update_huge' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:1520: warning: Function parameter or member 'flags' not described in 'amdgpu_vm_update_huge' drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c:3096: warning: Function parameter or member 'pasid' not described in 'amdgpu_vm_make_compute' drivers/gpu/drm/amd/display/amdgpu_dm/amdgpu_dm.h:128: warning: Incorrect use of kernel-doc format: Documentation Makefile include scripts source @atomic_obj drivers/gpu/drm/amd/display/amdgpu_dm/amdgpu_dm.h:203: warning: Function parameter or member 'atomic_obj' not described in 'amdgpu_display_manager' drivers/gpu/drm/amd/display/amdgpu_dm/amdgpu_dm.h:203: warning: Function parameter or member 'atomic_obj_lock' not described in 'amdgpu_display_manager' drivers/gpu/drm/amd/display/amdgpu_dm/amdgpu_dm.h:203: warning: Function parameter or member 'backlight_link' not described in 'amdgpu_display_manager' drivers/gpu/drm/amd/display/amdgpu_dm/amdgpu_dm.h:203: warning: Function parameter or member 'backlight_caps' not described in 'amdgpu_display_manager' drivers/gpu/drm/amd/display/amdgpu_dm/amdgpu_dm.h:203: warning: Function parameter or member 'freesync_module' not described in 'amdgpu_display_manager' drivers/gpu/drm/amd/display/amdgpu_dm/amdgpu_dm.h:203: warning: Function parameter or member 'fw_dmcu' not described in 'amdgpu_display_manager' drivers/gpu/drm/amd/display/amdgpu_dm/amdgpu_dm.h:203: warning: Function parameter or member 'dmcu_fw_version' not described in 'amdgpu_display_manager' drivers/gpu/drm/amd/display/amdgpu_dm/amdgpu_dm.c:1: warning: no structured comments found include/drm/drm_drv.h:618: warning: Function parameter or member 'gem_prime_pin' not described in 'drm_driver' include/drm/drm_drv.h:618: warning: Function parameter or member 'gem_prime_unpin' not described in 'drm_driver' include/drm/drm_drv.h:618: warning: Function parameter or member 'gem_prime_res_obj' not described in 'drm_driver' include/drm/drm_drv.h:618: warning: Function parameter or member 'gem_prime_get_sg_table' not described in 'drm_driver' include/drm/drm_drv.h:618: warning: Function parameter or member 'gem_prime_import_sg_table' not described in 'drm_driver' include/drm/drm_drv.h:618: warning: Function parameter or member 'gem_prime_vmap' not described in 'drm_driver' include/drm/drm_drv.h:618: warning: Function parameter or member 'gem_prime_vunmap' not described in 'drm_driver' include/drm/drm_drv.h:618: warning: Function parameter or member 'gem_prime_mmap' not described in 'drm_driver' include/drm/drm_atomic_state_helper.h:1: warning: no structured comments found drivers/gpu/drm/drm_dp_helper.c:1362: warning: Function parameter or member 'dsc_dpcd' not described in 'drm_dp_dsc_sink_max_slice_count' drivers/gpu/drm/drm_dp_helper.c:1362: warning: Function parameter or member 'is_edp' not described in 'drm_dp_dsc_sink_max_slice_count' drivers/gpu/drm/i915/i915_vma.h:49: warning: cannot understand function prototype: 'struct i915_vma ' drivers/gpu/drm/i915/i915_vma.h:1: warning: no structured comments found drivers/gpu/drm/i915/intel_guc_fwif.h:536: warning: cannot understand function prototype: 'struct guc_log_buffer_state ' drivers/gpu/drm/i915/i915_trace.h:1: warning: no structured comments found include/linux/skbuff.h:876: warning: Function parameter or member 'dev_scratch' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'list' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'ip_defrag_offset' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'skb_mstamp_ns' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member '__cloned_offset' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'head_frag' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member '__pkt_type_offset' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'encapsulation' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'encap_hdr_csum' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'csum_valid' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member '__pkt_vlan_present_offset' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'vlan_present' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'csum_complete_sw' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'csum_level' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'inner_protocol_type' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'remcsum_offload' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'sender_cpu' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'reserved_tailroom' not described in 'sk_buff' include/linux/skbuff.h:876: warning: Function parameter or member 'inner_ipproto' not described in 'sk_buff' include/net/sock.h:238: warning: Function parameter or member 'skc_addrpair' not described in 'sock_common' include/net/sock.h:238: warning: Function parameter or member 'skc_portpair' not described in 'sock_common' include/net/sock.h:238: warning: Function parameter or member 'skc_ipv6only' not described in 'sock_common' vim +886 sound/core/info.c 868 869 /** 870 * snd_card_rw_proc_new - Create a read/write text proc file entry for the card 871 * @card: the card instance 872 * @name: the file name 873 * @private_data: the arbitrary private data 874 * @read: the read callback 875 * 876 * This proc file entry will be registered via snd_card_register() call, and 877 * it will be removed automatically at the card removal, too. 878 */ 879 int snd_card_rw_proc_new(struct snd_card *card, const char *name, 880 void *private_data, 881 void (*read)(struct snd_info_entry *, 882 struct snd_info_buffer *), 883 void (*write)(struct snd_info_entry *entry, 884 struct snd_info_buffer *buffer)) 885 { > 886 struct snd_info_entry *entry; 887 888 entry = snd_info_create_card_entry(card, name, card->proc_root); 889 if (!entry) 890 return -ENOMEM; 891 snd_info_set_text_ops(entry, private_data, read); 892 if (write) { 893 entry->mode |= 0200; 894 entry->c.text.write = write; 895 } 896 return 0; 897 } 898 EXPORT_SYMBOL_GPL(snd_card_rw_proc_new); 899 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: compress: Fix stop handling on compressed capture streams
by Charles Keepax 05 Feb '19

05 Feb '19

It is normal user behaviour to start, stop, then start a stream again without closing it. Currently this works for compressed playback streams but not capture ones. The states on a compressed capture stream go directly from OPEN to PREPARED, unlike a playback stream which moves to SETUP and waits for a write of data before moving to PREPARED. Currently however, when a stop is sent the state is set to SETUP for both types of streams. This leaves a capture stream in the situation where a new start can't be sent as that requires the state to be PREPARED and a new set_params can't be sent as that requires the state to be OPEN. The only option being to close the stream, and then reopen. Correct this issues by allowing snd_compr_drain_notify to set the state depending on the stream direction, as we already do in set_params. Fixes: 49bb6402f1aa ("ALSA: compress_core: Add support for capture streams") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/compress_driver.h | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 0cdc3999ecfa8..c5188ff724d12 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -173,7 +173,11 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) if (snd_BUG_ON(!stream)) return; - stream->runtime->state = SNDRV_PCM_STATE_SETUP; + if (stream->direction == SND_COMPRESS_PLAYBACK) + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + else + stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + wake_up(&stream->runtime->sleep); } -- 2.11.0

2 1

[alsa-devel] [PATCH 0/4] ALSA: Yet more cleanups for procfs
by Takashi Iwai 05 Feb '19

05 Feb '19

Hi, this is the last part of my procfs cleanup patches. These are merely cleanups, no functional changes are expected. The whole patches are found in topic/proc-fixes in sound git tree. thanks, Takashi === Takashi Iwai (4): ALSA: info: Drop unused snd_info_entry.card field ALSA: info: Minor optimization ALSA: info: Move card id proc creation into info.c ALSA: cs46xx: Clean up proc file creations include/sound/core.h | 1 - include/sound/info.h | 1 - sound/core/info.c | 36 +++++++-------- sound/core/init.c | 31 ++----------- sound/pci/cs46xx/cs46xx_dsp_spos.h | 6 --- sound/pci/cs46xx/dsp_spos.c | 87 ++++++++++++------------------------- sound/pci/cs46xx/dsp_spos_scb_lib.c | 13 +++--- 7 files changed, 55 insertions(+), 120 deletions(-) -- 2.16.4

2 6

Re: [alsa-devel] [PATCH v6 0/3] ALSA: pcm: anonymous dup implementation
by Ricardo Biehl Pasquali 05 Feb '19

05 Feb '19

What about file sealing? See: Jonathan Corbet's article about sealed files (2014-04-09). <https://lwn.net/Articles/593918/>. Theodore Ts'o reply to file sealing patch suggesting an implementation on VFS (2014-03-20). Re: [PATCH 0/6] File Sealing & memfd_create() <https://lwn.net/Articles/593963/>. pasquali

1 0

[alsa-devel] quirks patch for new T+A USB DAC
by Udo Eberhardt 05 Feb '19

05 Feb '19

Hi all, This is a patch for sound/usb/quirks.c which enables DSD support for a new T+A DAC for which we have developed the USB firmware. Here is a patch description: This patch adds the T+A VID to the generic check in order to enable native DSD support for T+A devices. This works with the new T+A USB DAC model SD3100HV and will also work with future devices which support the XMOS/Thesycon style DSD format. I hope you can apply this patch. Let me know if you need more info. Thank you, Udo

2 4

[alsa-devel] [PATCH 00/11] ALSA: procfs cleanups
by Takashi Iwai 05 Feb '19

05 Feb '19

Hi, this is a patchset for cleaning up the procfs handling. The first three patches are to remove the superfluous snd_info_register() calls (which are done eventually at the card creation time), and the rest patches are for introducing new helpers and cleaning up with them. The latter has no functional changes but only reducing lots of codes. thanks, Takashi === Takashi Iwai (11): ALSA: pcm: Remove superfluous snd_info_register() calls ALSA: compress: Remove superfluous snd_info_register() calls ALSA: pci: Remove superfluous snd_info_register() calls ALSA: info: Add standard helpers for card proc file entries ALSA: drivers: Clean up with new procfs helpers ALSA: isa: Clean up with new procfs helpers ALSA: i2c: Clean up with new procfs helpers ALSA: pci: Clean up with new procfs helpers ALSA: pcmcia: Clean up with new procfs helpers ALSA: sparc: Clean up with new procfs helpers ALSA: usb: Clean up with new procfs helpers include/sound/info.h | 35 ++++++++++++++ sound/core/compress_offload.c | 11 +---- sound/core/info.c | 31 +++++++++++++ sound/core/pcm.c | 79 +++++++++++--------------------- sound/drivers/aloop.c | 10 +--- sound/drivers/dummy.c | 10 +--- sound/drivers/vx/vx_core.c | 5 +- sound/i2c/other/ak4113.c | 5 +- sound/i2c/other/ak4114.c | 5 +- sound/i2c/other/ak4xxx-adda.c | 8 +--- sound/isa/gus/gus_irq.c | 5 +- sound/isa/gus/gus_mem.c | 6 +-- sound/isa/opti9xx/miro.c | 5 +- sound/isa/sb/sb16_csp.c | 5 +- sound/pci/ac97/ac97_proc.c | 26 ++++------- sound/pci/ad1889.c | 6 +-- sound/pci/ak4531_codec.c | 5 +- sound/pci/ali5451/ali5451.c | 4 +- sound/pci/asihpi/asihpi.c | 6 +-- sound/pci/atiixp.c | 5 +- sound/pci/atiixp_modem.c | 6 +-- sound/pci/ca0106/ca0106_proc.c | 40 ++++++---------- sound/pci/cmipci.c | 5 +- sound/pci/cs4281.c | 3 +- sound/pci/cs46xx/dsp_spos.c | 16 ++----- sound/pci/emu10k1/emu10k1x.c | 12 ++--- sound/pci/emu10k1/emuproc.c | 81 +++++++++++++-------------------- sound/pci/ens1370.c | 6 +-- sound/pci/hda/hda_proc.c | 9 +--- sound/pci/ice1712/ice1712.c | 5 +- sound/pci/ice1712/ice1724.c | 5 +- sound/pci/ice1712/pontis.c | 12 ++--- sound/pci/ice1712/prodigy192.c | 5 +- sound/pci/ice1712/prodigy_hifi.c | 8 +--- sound/pci/ice1712/quartet.c | 4 +- sound/pci/intel8x0.c | 6 +-- sound/pci/intel8x0m.c | 6 +-- sound/pci/korg1212/korg1212.c | 6 +-- sound/pci/lola/lola_proc.c | 16 ++----- sound/pci/lx6464es/lx6464es.c | 8 +--- sound/pci/mixart/mixart.c | 6 +-- sound/pci/oxygen/oxygen_lib.c | 5 +- sound/pci/pcxhr/pcxhr.c | 21 +++------ sound/pci/riptide/riptide.c | 6 +-- sound/pci/rme32.c | 5 +- sound/pci/rme96.c | 5 +- sound/pci/rme9652/hdsp.c | 5 +- sound/pci/rme9652/hdspm.c | 55 ++++++++++------------ sound/pci/rme9652/rme9652.c | 6 +-- sound/pci/sonicvibes.c | 6 +-- sound/pci/trident/trident_main.c | 4 +- sound/pci/via82xx.c | 6 +-- sound/pci/via82xx_modem.c | 6 +-- sound/pci/ymfpci/ymfpci_main.c | 6 +-- sound/pcmcia/pdaudiocf/pdaudiocf_core.c | 5 +- sound/sparc/dbri.c | 10 +--- sound/usb/mixer.c | 7 ++- sound/usb/mixer_quirks.c | 6 +-- sound/usb/proc.c | 13 ++---- 59 files changed, 268 insertions(+), 446 deletions(-) -- 2.16.4

2 13

[alsa-devel] [PATCH v2] ALSA: hda/tegra: enable clock during probe
by Sameer Pujar 05 Feb '19

05 Feb '19

If CONFIG_PM is disabled or runtime PM calls are forbidden, the clocks will not be ON. This could cause issue during probe, where hda init setup is done. This patch enables clocks unconditionally during probe. Along with above, follwoing changes are done. * enable runtime PM before exiting from probe work. This helps to avoid usage of pm_runtime_get_sync/pm_runtime_put() in probe work. * hda_tegra_disable_clocks() is moved out of CONFIG_PM_SLEEP check. * runtime PM callbacks moved out of CONFIG_PM check Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Reviewed-by: Ravindra Lokhande <rlokhande(a)nvidia.com> Reviewed-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 26 +++++++++++++++++--------- 1 file changed, 17 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index c8d18dc..ba6175f 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -219,7 +219,6 @@ static int hda_tegra_enable_clocks(struct hda_tegra *data) return rc; } -#ifdef CONFIG_PM_SLEEP static void hda_tegra_disable_clocks(struct hda_tegra *data) { clk_disable_unprepare(data->hda2hdmi_clk); @@ -227,6 +226,7 @@ static void hda_tegra_disable_clocks(struct hda_tegra *data) clk_disable_unprepare(data->hda_clk); } +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -257,7 +257,6 @@ static int hda_tegra_resume(struct device *dev) } #endif /* CONFIG_PM_SLEEP */ -#ifdef CONFIG_PM static int hda_tegra_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -283,7 +282,7 @@ static int hda_tegra_runtime_resume(struct device *dev) int rc; rc = hda_tegra_enable_clocks(hda); - if (rc != 0) + if (rc) return rc; if (chip && chip->running) { hda_tegra_init(hda); @@ -292,7 +291,6 @@ static int hda_tegra_runtime_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops hda_tegra_pm = { SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) @@ -551,9 +549,9 @@ static int hda_tegra_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, card); - pm_runtime_enable(hda->dev); - if (!azx_has_pm_runtime(chip)) - pm_runtime_forbid(hda->dev); + err = hda_tegra_enable_clocks(hda); + if (err) + goto out_free; schedule_work(&hda->probe_work); @@ -571,7 +569,6 @@ static void hda_tegra_probe_work(struct work_struct *work) struct platform_device *pdev = to_platform_device(hda->dev); int err; - pm_runtime_get_sync(hda->dev); err = hda_tegra_first_init(chip, pdev); if (err < 0) goto out_free; @@ -592,8 +589,15 @@ static void hda_tegra_probe_work(struct work_struct *work) chip->running = 1; snd_hda_set_power_save(&chip->bus, power_save * 1000); + /* set device state as active */ + if (pm_runtime_set_active(hda->dev) < 0) + goto out_free; + /* enable runtime PM */ + pm_runtime_enable(hda->dev); + if (!azx_has_pm_runtime(chip)) + pm_runtime_forbid(hda->dev); + out_free: - pm_runtime_put(hda->dev); return; /* no error return from async probe */ } @@ -603,6 +607,10 @@ static int hda_tegra_remove(struct platform_device *pdev) ret = snd_card_free(dev_get_drvdata(&pdev->dev)); pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) { + hda_tegra_runtime_suspend(&pdev->dev); + pm_runtime_set_suspended(&pdev->dev); + } return ret; } -- 2.7.4

7 29

[alsa-devel] [PATCH] ASoC: wcd9335: remove some unnecessary NULL checks
by Dan Carpenter 05 Feb '19

05 Feb '19

These are arrays, not pointers, and they can't be NULL. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/wcd9335.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 3878187bb512..981f88a5f615 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2001,20 +2001,16 @@ static int wcd9335_set_channel_map(struct snd_soc_dai *dai, return -EINVAL; } - if (wcd->rx_chs) { - wcd->num_rx_port = rx_num; - for (i = 0; i < rx_num; i++) { - wcd->rx_chs[i].ch_num = rx_slot[i]; - INIT_LIST_HEAD(&wcd->rx_chs[i].list); - } + wcd->num_rx_port = rx_num; + for (i = 0; i < rx_num; i++) { + wcd->rx_chs[i].ch_num = rx_slot[i]; + INIT_LIST_HEAD(&wcd->rx_chs[i].list); } - if (wcd->tx_chs) { - wcd->num_tx_port = tx_num; - for (i = 0; i < tx_num; i++) { - wcd->tx_chs[i].ch_num = tx_slot[i]; - INIT_LIST_HEAD(&wcd->tx_chs[i].list); - } + wcd->num_tx_port = tx_num; + for (i = 0; i < tx_num; i++) { + wcd->tx_chs[i].ch_num = tx_slot[i]; + INIT_LIST_HEAD(&wcd->tx_chs[i].list); } return 0; -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: hda/ca0132: need CONFIG_PCI for compile
by Kuninori Morimoto 05 Feb '19

05 Feb '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit d99501b8575d ("ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap()") started to use pci_iounmap() on ca0132, but then, it should be depends on CONFIG_PCI. Otherwise, non CONFIG_PCI compile will be failed, like below linux/sound/pci/hda/patch_ca0132.c: In function 'ca0132_free': linux/sound/pci/hda/patch_ca0132.c:8455:3: error: implicit\ declaration of function 'pci_iounmap'; did you mean\ 'pcim_iounmap'? [-Werror=implicit-function-declaration] pci_iounmap(codec->bus->pci, spec->mem_base); ^~~~~~~~~~~ Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/pci/hda/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 4235907..03af159 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -171,6 +171,7 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_CA0132 tristate "Build Creative CA0132 codec support" + depends on CONFIG_PCI help Say Y or M here to include Creative CA0132 codec support in snd-hda-intel driver. -- 2.7.4

2 2

[alsa-devel] [PATCH v3 1/2] ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables
by Pierre-Louis Bossart 05 Feb '19

05 Feb '19

KASAN reports and additional traces point to out-of-bounds accesses to the dapm_up_seq and dapm_down_seq lookup tables. The indices used are larger than the array definition. Fix by adding missing entries for the new widget types in these two lookup tables, and align them with PGA values. Also the sequences for the following widgets were not defined. Since their values defaulted to zero, assign them explicitly snd_soc_dapm_input snd_soc_dapm_output snd_soc_dapm_vmid snd_soc_dapm_siggen snd_soc_dapm_sink Fixes: 8a70b4544ef4 ('ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.'). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5b74dffc9c11..2bf5bab579de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -70,12 +70,16 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, + [snd_soc_dapm_vmid] = 3, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai_in] = 4, [snd_soc_dapm_dai_out] = 4, [snd_soc_dapm_aif_in] = 4, [snd_soc_dapm_aif_out] = 4, [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_siggen] = 5, + [snd_soc_dapm_input] = 5, + [snd_soc_dapm_output] = 5, [snd_soc_dapm_mux] = 6, [snd_soc_dapm_demux] = 6, [snd_soc_dapm_dac] = 7, @@ -83,11 +87,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer] = 8, [snd_soc_dapm_mixer_named_ctl] = 8, [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_buffer] = 9, + [snd_soc_dapm_scheduler] = 9, + [snd_soc_dapm_effect] = 9, + [snd_soc_dapm_src] = 9, + [snd_soc_dapm_asrc] = 9, + [snd_soc_dapm_encoder] = 9, + [snd_soc_dapm_decoder] = 9, [snd_soc_dapm_adc] = 10, [snd_soc_dapm_out_drv] = 11, [snd_soc_dapm_hp] = 11, [snd_soc_dapm_spk] = 11, [snd_soc_dapm_line] = 11, + [snd_soc_dapm_sink] = 11, [snd_soc_dapm_kcontrol] = 12, [snd_soc_dapm_post] = 13, }; @@ -100,13 +112,25 @@ static int dapm_down_seq[] = { [snd_soc_dapm_spk] = 3, [snd_soc_dapm_line] = 3, [snd_soc_dapm_out_drv] = 3, + [snd_soc_dapm_sink] = 3, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_buffer] = 4, + [snd_soc_dapm_scheduler] = 4, + [snd_soc_dapm_effect] = 4, + [snd_soc_dapm_src] = 4, + [snd_soc_dapm_asrc] = 4, + [snd_soc_dapm_encoder] = 4, + [snd_soc_dapm_decoder] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_siggen] = 7, + [snd_soc_dapm_input] = 7, + [snd_soc_dapm_output] = 7, [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_vmid] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_demux] = 9, [snd_soc_dapm_aif_in] = 10, -- 2.17.1

2 5

[alsa-devel] [resend] [PATCH] ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check
by Kuninori Morimoto 05 Feb '19

05 Feb '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") added new rsnd_ssi_prepare() and moved rsnd_ssi_master_clk_start() to .prepare. But, ssi user count (= ssi->usrcnt) is incremented at .init (= rsnd_ssi_init()). Because of these timing exchange, ssi->usrcnt check at rsnd_ssi_master_clk_start() should be adjusted. Otherwise, 2nd master clock setup will be no check. This patch fixup this issue. Fixes: commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Reported-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Reported-by: Valentine Barshak <valentine.barshak(a)cogentembedded.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> --- Mark this is for v5.0 sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 45ef295..f5afab6 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -286,7 +286,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->usrcnt > 1) { + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; -- 2.7.4 _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] [asoc:for-5.1 100/129] sound/soc/codecs/wcd9335.c:2168:5-8: Unneeded variable: "ret". Return "0" on line 2182 (fwd)
by Julia Lawall 04 Feb '19

04 Feb '19

It's a small thing, but just returning 0 would be a little bit more clear than setting ret to 0 and then returning that. julia ---------- Forwarded message ---------- Date: Tue, 5 Feb 2019 05:57:05 +0800 From: kbuild test robot <lkp(a)intel.com> To: kbuild(a)01.org Cc: Julia Lawall <julia.lawall(a)lip6.fr> Subject: [asoc:for-5.1 100/129] sound/soc/codecs/wcd9335.c:2168:5-8: Unneeded variable: "ret". Return "0" on line 2182 CC: kbuild-all(a)01.org CC: alsa-devel(a)alsa-project.org TO: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> CC: Mark Brown <broonie(a)kernel.org> CC: Vinod Koul <vkoul(a)kernel.org> tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.1 head: 0f747bb273790f49be4660521d86d7cebd1bbe1e commit: 354461486f66e4311d9412c53205d773aac85b78 [100/129] ASoC: wcd9335: add playback dapm widgets :::::: branch date: 6 hours ago :::::: commit date: 7 days ago >> sound/soc/codecs/wcd9335.c:2168:5-8: Unneeded variable: "ret". Return "0" on line 2182 sound/soc/codecs/wcd9335.c:2498:5-8: Unneeded variable: "ret". Return "0" on line 2528 sound/soc/codecs/wcd9335.c:2302:5-8: Unneeded variable: "ret". Return "0" on line 2341 sound/soc/codecs/wcd9335.c:2453:5-8: Unneeded variable: "ret". Return "0" on line 2491 sound/soc/codecs/wcd9335.c:1583:5-8: Unneeded variable: "ret". Return "0" on line 1595 # https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/commit/?i… git remote add asoc https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git git remote update asoc git checkout 354461486f66e4311d9412c53205d773aac85b78 vim +2168 sound/soc/codecs/wcd9335.c 35446148 Srinivas Kandagatla 2019-01-28 2162 35446148 Srinivas Kandagatla 2019-01-28 2163 static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w, 35446148 Srinivas Kandagatla 2019-01-28 2164 struct snd_kcontrol *kc, int event) 35446148 Srinivas Kandagatla 2019-01-28 2165 { 35446148 Srinivas Kandagatla 2019-01-28 2166 struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); 35446148 Srinivas Kandagatla 2019-01-28 2167 struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); 35446148 Srinivas Kandagatla 2019-01-28 @2168 int ret = 0; 35446148 Srinivas Kandagatla 2019-01-28 2169 35446148 Srinivas Kandagatla 2019-01-28 2170 switch (event) { 35446148 Srinivas Kandagatla 2019-01-28 2171 case SND_SOC_DAPM_PRE_PMU: 35446148 Srinivas Kandagatla 2019-01-28 2172 wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_PRE_DAC, 35446148 Srinivas Kandagatla 2019-01-28 2173 WCD_CLSH_STATE_EAR, CLS_H_NORMAL); 35446148 Srinivas Kandagatla 2019-01-28 2174 35446148 Srinivas Kandagatla 2019-01-28 2175 break; 35446148 Srinivas Kandagatla 2019-01-28 2176 case SND_SOC_DAPM_POST_PMD: 35446148 Srinivas Kandagatla 2019-01-28 2177 wcd_clsh_ctrl_set_state(wcd->clsh_ctrl, WCD_CLSH_EVENT_POST_PA, 35446148 Srinivas Kandagatla 2019-01-28 2178 WCD_CLSH_STATE_EAR, CLS_H_NORMAL); 35446148 Srinivas Kandagatla 2019-01-28 2179 break; 35446148 Srinivas Kandagatla 2019-01-28 2180 }; 35446148 Srinivas Kandagatla 2019-01-28 2181 35446148 Srinivas Kandagatla 2019-01-28 @2182 return ret; 35446148 Srinivas Kandagatla 2019-01-28 2183 } 35446148 Srinivas Kandagatla 2019-01-28 2184 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 0/5] ALSA: Fix remaining NULL device with memory allocations
by Takashi Iwai 04 Feb '19

04 Feb '19

Hi, this patchset covers the still remaining drivers calling the memory allocation with NULL device pointer. Simply card->dev is passed there. Mark, two of them are trivial fixes for ASoC, and I'd like to merge them through my tree, since the last patch adds an explicit WARN_ON(). Give me Ack's if they are OK. thanks, Takashi === Takashi Iwai (5): ALSA: x86: Avoid passing NULL to memory allocators ALSA: arm: Avoid passing NULL to memory allocators ASoC: amd: Avoid passing NULL to memory allocators ASoC: sh: Avoid passing NULL to memory allocators ALSA: core: Don't allow NULL device for memory allocation sound/arm/aaci.c | 3 ++- sound/core/memalloc.c | 2 ++ sound/soc/amd/raven/acp3x-pcm-dma.c | 3 ++- sound/soc/sh/siu_pcm.c | 2 +- sound/x86/intel_hdmi_audio.c | 3 ++- 5 files changed, 9 insertions(+), 4 deletions(-) -- 2.16.4

3 8

[alsa-devel] [PATCH v2] ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables
by Pierre-Louis Bossart 04 Feb '19

04 Feb '19

KASAN reports and additional traces point to out-of-bounds accesses to the dapm_up_seq and dapm_down_seq lookup tables. The indices used are larger than the array definition. Fix by adding missing entries for the new widget types in these two lookup tables, and align them with PGA values. Also the sequences for the following widgets were not defined. Since their values defaulted to zero, assign them explicitly snd_soc_dapm_input snd_soc_dapm_output snd_soc_dapm_vmid snd_soc_dapm_siggen snd_soc_dapm_sink Fixes: 8a70b4544ef4 ('ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.'). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: fixed mistake with mic (was already handled) use values recommended by Mark sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5b74dffc9c11..2bf5bab579de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -70,12 +70,16 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, + [snd_soc_dapm_vmid] = 3, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai_in] = 4, [snd_soc_dapm_dai_out] = 4, [snd_soc_dapm_aif_in] = 4, [snd_soc_dapm_aif_out] = 4, [snd_soc_dapm_mic] = 5, + [snd_soc_dapm_siggen] = 5, + [snd_soc_dapm_input] = 5, + [snd_soc_dapm_output] = 5, [snd_soc_dapm_mux] = 6, [snd_soc_dapm_demux] = 6, [snd_soc_dapm_dac] = 7, @@ -83,11 +87,19 @@ static int dapm_up_seq[] = { [snd_soc_dapm_mixer] = 8, [snd_soc_dapm_mixer_named_ctl] = 8, [snd_soc_dapm_pga] = 9, + [snd_soc_dapm_buffer] = 9, + [snd_soc_dapm_scheduler] = 9, + [snd_soc_dapm_effect] = 9, + [snd_soc_dapm_src] = 9, + [snd_soc_dapm_asrc] = 9, + [snd_soc_dapm_encoder] = 9, + [snd_soc_dapm_decoder] = 9, [snd_soc_dapm_adc] = 10, [snd_soc_dapm_out_drv] = 11, [snd_soc_dapm_hp] = 11, [snd_soc_dapm_spk] = 11, [snd_soc_dapm_line] = 11, + [snd_soc_dapm_sink] = 11, [snd_soc_dapm_kcontrol] = 12, [snd_soc_dapm_post] = 13, }; @@ -100,13 +112,25 @@ static int dapm_down_seq[] = { [snd_soc_dapm_spk] = 3, [snd_soc_dapm_line] = 3, [snd_soc_dapm_out_drv] = 3, + [snd_soc_dapm_sink] = 3, [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_buffer] = 4, + [snd_soc_dapm_scheduler] = 4, + [snd_soc_dapm_effect] = 4, + [snd_soc_dapm_src] = 4, + [snd_soc_dapm_asrc] = 4, + [snd_soc_dapm_encoder] = 4, + [snd_soc_dapm_decoder] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, [snd_soc_dapm_mixer] = 5, [snd_soc_dapm_dac] = 6, [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_siggen] = 7, + [snd_soc_dapm_input] = 7, + [snd_soc_dapm_output] = 7, [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_vmid] = 8, [snd_soc_dapm_mux] = 9, [snd_soc_dapm_demux] = 9, [snd_soc_dapm_aif_in] = 10, -- 2.17.1

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[alsa-devel] [PATCH 0/3 v5] ALSA: pcm: anonymous dup implementation
by Jaroslav Kysela 04 Feb '19

04 Feb '19

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup The filtering might be refined. It depends on the real requirements. Perhaps, we may create more ioctl groups. Any comments are more than welcome. v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) v3 of the patches: - group integer declarations in snd_pcm_anonymous_dup() - replaced substream->pcm with pcm in snd_pcm_anonymous_dup() - added SNDRV_PCM_PERM_RW check for read/write/readv/writev syscalls v4 of the patches: - more simple restriction control (only two modes - full/buffer) - the tinyalsa implementation follows this change v5 of the patches: - merge pcm_..._mmap_allowed fcns to the snd_pcm_mmap_... fcns Cc: Phil Burk <philburk(a)google.com> Cc: Zach Riggle <riggle(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (3): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: merge pcm_..._mmap_allowed fcns to the snd_pcm_mmap_... fcns ALSA: pcm: implement the mmap buffer mode for the anonymous dup include/sound/pcm.h | 10 +-- include/uapi/sound/asound.h | 6 +- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 ++-- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 151 ++++++++++++++++++++++++++++++++------------ 6 files changed, 130 insertions(+), 53 deletions(-) -- 2.13.6

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[alsa-devel] [PATCH][next] ASoC: mediatek: btcvsd: fix spelling mistake "offest" -> "offset"
by Colin King 04 Feb '19

04 Feb '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_warn message. Fix this. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index 349a9120a0dd..e408c1b270ab 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1309,7 +1309,7 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) offset, ARRAY_SIZE(offset)); if (ret) { - dev_warn(dev, "%s(), get offest fail, ret %d\n", __func__, ret); + dev_warn(dev, "%s(), get offset fail, ret %d\n", __func__, ret); return ret; } btcvsd->infra_misc_offset = offset[0]; -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: ak4118: fix missing header
by Clément Péron 04 Feb '19

04 Feb '19

This driver use the gpio consumer interface. Add the header as it's needed. Signed-off-by: Clément Péron <peron.clem(a)gmail.com> --- This has been triggered by Kbuild-all for information: https://lists.01.org/pipermail/kbuild-all/2019-February/057379.html sound/soc/codecs/ak4118.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c index 238ab29f2bf4..ce419e8cf890 100644 --- a/sound/soc/codecs/ak4118.c +++ b/sound/soc/codecs/ak4118.c @@ -6,6 +6,7 @@ */ #include <linux/i2c.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of_device.h> #include <linux/of_gpio.h> -- 2.19.1

2 1

[alsa-devel] [PATCH v2] ASoC: topology: unload physical dai link in remove
by Pierre-Louis Bossart 04 Feb '19

04 Feb '19

From: Bard liao <yung-chuan.liao(a)linux.intel.com> soc_tplg_link_config() will find the physical dai link and call soc_tplg_dai_link_load() to load the BE dai link. Currently remove_link() is only used to remove the FE dai link which is created by the topology. The BE dai link cannot however be unloaded in snd_soc_tplg_component _remove(), which is problematic if anything needs to be released or reinitialized. This patch aligns the definitions of dynamic types with the existing UAPI and adds a new remove_backend_link() routine to unload the the BE dai link when snd_soc_tplg_component_remove() is invoked. Signed-off-by: Bard liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: reset the dobj.type to fix module load/unload issues include/sound/soc-topology.h | 1 + sound/soc/soc-topology.c | 32 ++++++++++++++++++++++++++++++++ 2 files changed, 33 insertions(+) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 8c43cfc240fa..5223896de26f 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -45,6 +45,7 @@ enum snd_soc_dobj_type { SND_SOC_DOBJ_DAI_LINK, SND_SOC_DOBJ_PCM, SND_SOC_DOBJ_CODEC_LINK, + SND_SOC_DOBJ_BACKEND_LINK, }; /* dynamic control object */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index bb7e5422a419..c5638e15a2dd 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -562,6 +562,25 @@ static void remove_link(struct snd_soc_component *comp, kfree(link); } +/* unload dai link */ +static void remove_backend_link(struct snd_soc_component *comp, + struct snd_soc_dobj *dobj, int pass) +{ + if (pass != SOC_TPLG_PASS_LINK) + return; + + if (dobj->ops && dobj->ops->link_unload) + dobj->ops->link_unload(comp, dobj); + + /* + * We don't free the link here as what remove_link() do since BE + * links are not allocated by topology. + * We however need to reset the dobj type to its initial values + */ + dobj->type = SND_SOC_DOBJ_NONE; + list_del(&dobj->list); +} + /* bind a kcontrol to it's IO handlers */ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, struct snd_kcontrol_new *k, @@ -2168,6 +2187,12 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, return ret; } + /* for unloading it in snd_soc_tplg_component_remove */ + link->dobj.index = tplg->index; + link->dobj.ops = tplg->ops; + link->dobj.type = SND_SOC_DOBJ_BACKEND_LINK; + list_add(&link->dobj.list, &tplg->comp->dobj_list); + return 0; } @@ -2654,6 +2679,13 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) case SND_SOC_DOBJ_DAI_LINK: remove_link(comp, dobj, pass); break; + case SND_SOC_DOBJ_BACKEND_LINK: + /* + * call link_unload ops if extra + * deinitialization is needed. + */ + remove_backend_link(comp, dobj, pass); + break; default: dev_err(comp->dev, "ASoC: invalid component type %d for removal\n", dobj->type); -- 2.17.1

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[alsa-devel] [PATCH] ASoC: soc-core: clear platform pointers on error
by Curtis Malainey 04 Feb '19

04 Feb '19

Originally snd_soc_init_platform was not cleaning up its pointers, this was fixed to always reallocate dynamic memory but created a memory leak when snd_soc_init_platform was called multiple times during the same probe attempt and also threw away any changes made to the struct between calls. In order to avoid reallocating memory that is still valid, the behaviour will be changed to clear the dynamically set pointers on a probe error and a unregister event and snd_soc_init_platform will go back to its original behaviour of only allocating null pointers so it will stop throwing away valid changes. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/soc-core.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d6b5edba8d2d..9431dd605103 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1053,7 +1053,7 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * soc.h :: struct snd_soc_dai_link */ /* convert Legacy platform link */ - if (!platform || dai_link->legacy_platform) { + if (!platform) { platform = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); @@ -1076,6 +1076,24 @@ static int snd_soc_init_platform(struct snd_soc_card *card, return 0; } +static void soc_cleanup_platform(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link; + int i; + /* + * FIXME + * + * this function should be removed with snd_soc_init_platform + */ + + for_each_card_prelinks(card, i, link) { + if (link->legacy_platform) { + link->legacy_platform = 0; + link->platforms = NULL; + } + } +} + static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -2020,6 +2038,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove and free each DAI */ soc_remove_dai_links(card); soc_remove_pcm_runtimes(card); + soc_cleanup_platform(card); /* remove auxiliary devices */ soc_remove_aux_devices(card); -- 2.20.1.495.gaa96b0ce6b-goog

2 1

[alsa-devel] Applied "ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe" to the asoc tree
by Mark Brown 04 Feb '19

04 Feb '19

The patch ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 11907e9d3533648615db08140e3045b829d2c141 Mon Sep 17 00:00:00 2001 From: wen yang <yellowriver2010(a)hotmail.com> Date: Sat, 2 Feb 2019 14:53:16 +0000 Subject: [PATCH] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 81f2fe2c6d23..60f87a0d99f4 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: rsnd: fixup MIX kctrl registration
by Kuninori Morimoto 02 Feb '19

02 Feb '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas sound device has many IPs and many situations. If platform/board uses MIXer, situation will be more complex. To avoid duplicate DVC kctrl registration when MIXer was used, it had original flags. But it was issue when sound card was re-binded, because no one can't cleanup this flags then. To solve this issue, commit 9c698e8481a15237a ("ASoC: rsnd: tidyup registering method for rsnd_kctrl_new()") checks registered card->controls, because if card was re-binded, these were cleanuped automatically. This patch could solve re-binding issue. But, it start to avoid MIX kctrl. To solve these issues, we need below. To avoid card re-binding issue: check registered card->controls To avoid duplicate DVC registration: check registered rsnd_kctrl_cfg To allow multiple MIX registration: check registered rsnd_kctrl_cfg This patch do it. Fixes: 9c698e8481a15237a ("ASoC: rsnd: tidyup registering method for rsnd_kctrl_new()") Reported-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-By: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 2c2c60a..9834474 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1524,14 +1524,14 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, int ret; /* - * 1) Avoid duplicate register (ex. MIXer case) - * 2) re-register if card was rebinded + * 1) Avoid duplicate register for DVC with MIX case + * 2) Allow duplicate register for MIX + * 3) re-register if card was rebinded */ list_for_each_entry(kctrl, &card->controls, list) { struct rsnd_kctrl_cfg *c = kctrl->private_data; - if (strcmp(kctrl->id.name, name) == 0 && - c->mod == mod) + if (c == cfg) return 0; } -- 2.7.4

2 1

[alsa-devel] [PATCH 0/2] ASoC: rsnd: fixup TDM Split mode judgment
by Kuninori Morimoto 02 Feb '19

02 Feb '19

Hi Mark These fixup Renesas sound TDM Split mode vs CTU issue. Kuninori Morimoto (2): ASoC: rsnd: synchronize connection check for simple-card/audio-graph ASoC: rsnd: fixup TDM Split mode check for CTU sound/soc/sh/rcar/core.c | 40 +++++++++++++++++++--------------------- 1 file changed, 19 insertions(+), 21 deletions(-) Best regards --- Kuninori Morimoto

2 4

[alsa-devel] [PATCH 1/2] ASoC: dapm: Only power up active channels from a DAI
by Charles Keepax 02 Feb '19

02 Feb '19

Currently all widgets attached to a DAI link will be powered up when the DAI is active, however this may include routes that are not actually in use if there are unused channels available on the DAI. The macros for creating AIF widgets already include an entry for slot, it is proposed to change that to channel. The effective difference here being respresenting the logical channel index rather than the physical slot index. The CODECs currently using the slot entry on the DAPM_AIF macros are using it in a manner consistent with this, the CODECs not using it just have the field set to zero. A variable is added to snd_soc_dapm_widget to represent this channel index and then for each AIF widget attached to a DAI this is compared against the number of channels on the stream. Enabling the links for those which will be in use. This has the nice property that the CODECs which haven't used the slot/channel entry in the macro will function exactly as before due to all the AIF widgets having a channel of zero and a stream by definition having at least one channel. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc-dapm.h | 22 +++++++++----- sound/soc/soc-dapm.c | 76 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-pcm.c | 4 +++ 3 files changed, 94 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 46f2ba3ffcb7c..79b4ddfb8e9e2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -214,21 +214,21 @@ struct device; .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} /* stream domain */ -#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \ +#define SND_SOC_DAPM_AIF_IN(wname, stname, wchan, wreg, wshift, winvert) \ { .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } -#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wslot, wreg, wshift, winvert, \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } +#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wchan, wreg, wshift, winvert, \ wevent, wflags) \ { .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .event = wevent, .event_flags = wflags } -#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \ +#define SND_SOC_DAPM_AIF_OUT(wname, stname, wchan, wreg, wshift, winvert) \ { .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } -#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wslot, wreg, wshift, winvert, \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } +#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wchan, wreg, wshift, winvert, \ wevent, wflags) \ { .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .event = wevent, .event_flags = wflags } #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, \ @@ -407,6 +407,10 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); +int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); + /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card); void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm); @@ -627,6 +631,8 @@ struct snd_soc_dapm_widget { int endpoints[2]; struct clk *clk; + + int channel; }; struct snd_soc_dapm_update { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c4c134195392..1714f612451e1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2525,6 +2525,78 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); +static int dapm_update_dai_chan(struct snd_soc_dapm_path *p, + struct snd_soc_dapm_widget *w, + int channels) +{ + switch (w->id) { + case snd_soc_dapm_aif_out: + case snd_soc_dapm_aif_in: + break; + default: + return 0; + } + + dev_dbg(w->dapm->dev, "%s DAI route %s -> %s\n", + w->channel < channels ? "Connecting" : "Disconnecting", + p->source->name, p->sink->name); + + if (w->channel < channels) + soc_dapm_connect_path(p, true, "dai update"); + else + soc_dapm_connect_path(p, false, "dai update"); + + return 0; +} + +static int dapm_update_dai_unlocked(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int dir = substream->stream; + int channels = params_channels(params); + struct snd_soc_dapm_path *p; + struct snd_soc_dapm_widget *w; + int ret; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) + w = dai->playback_widget; + else + w = dai->capture_widget; + + dev_dbg(dai->dev, "Update DAI routes for %s %s\n", dai->name, + dir == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); + + snd_soc_dapm_widget_for_each_sink_path(w, p) { + ret = dapm_update_dai_chan(p, p->sink, channels); + if (ret < 0) + return ret; + } + + snd_soc_dapm_widget_for_each_source_path(w, p) { + ret = dapm_update_dai_chan(p, p->source, channels); + if (ret < 0) + return ret; + } + + return 0; +} + +int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret; + + mutex_lock_nested(&rtd->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = dapm_update_dai_unlocked(substream, params, dai); + mutex_unlock(&rtd->card->dapm_mutex); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_update_dai); + /* * dapm_update_widget_flags() - Re-compute widget sink and source flags * @w: The widget for which to update the flags @@ -3689,6 +3761,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = soc_dai_hw_params(&substream, params, source); if (ret < 0) goto out; + + dapm_update_dai_unlocked(&substream, params, source); } substream.stream = SNDRV_PCM_STREAM_PLAYBACK; @@ -3709,6 +3783,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = soc_dai_hw_params(&substream, params, sink); if (ret < 0) goto out; + + dapm_update_dai_unlocked(&substream, params, sink); } break; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 03f36e534050f..a5b40e82dea4a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -969,6 +969,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_dai->channels = params_channels(&codec_params); codec_dai->sample_bits = snd_pcm_format_physical_width( params_format(&codec_params)); + + snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); } ret = soc_dai_hw_params(substream, params, cpu_dai); @@ -998,6 +1000,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); + snd_soc_dapm_update_dai(substream, params, cpu_dai); + ret = soc_pcm_params_symmetry(substream, params); if (ret) goto component_err; -- 2.11.0

2 3

[alsa-devel] Applied "ASoC: mediatek: add documents for btcvsd driver" to the asoc tree
by Mark Brown 02 Feb '19

02 Feb '19

The patch ASoC: mediatek: add documents for btcvsd driver has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 425da20a98e0ab6ba6b37ff00ab50519d7dd7740 Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Date: Wed, 30 Jan 2019 14:47:11 +0800 Subject: [PATCH] ASoC: mediatek: add documents for btcvsd driver document for btcvsd driver Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/mtk-btcvsd-snd.txt | 24 +++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt diff --git a/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt b/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt new file mode 100644 index 000000000000..679e44839b48 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt @@ -0,0 +1,24 @@ +Mediatek ALSA BT SCO CVSD/MSBC Driver + +Required properties: +- compatible = "mediatek,mtk-btcvsd-snd"; +- reg: register location and size of PKV and SRAM_BANK2 +- interrupts: should contain BTSCO interrupt +- mediatek,infracfg: the phandles of INFRASYS +- mediatek,offset: Array contains of register offset and mask + infra_misc_offset, + infra_conn_bt_cvsd_mask, + cvsd_mcu_read_offset, + cvsd_mcu_write_offset, + cvsd_packet_indicator_offset + +Example: + + mtk-btcvsd-snd@18000000 { + compatible = "mediatek,mtk-btcvsd-snd"; + reg=<0 0x18000000 0 0x1000>, + <0 0x18080000 0 0x8000>; + interrupts = <GIC_SPI 286 IRQ_TYPE_LEVEL_LOW>; + mediatek,infracfg = <&infrasys>; + mediatek,offset = <0xf00 0x800 0xfd0 0xfd4 0xfd8>; + }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: mediatek: add btcvsd driver" to the asoc tree
by Mark Brown 02 Feb '19

02 Feb '19

The patch ASoC: mediatek: add btcvsd driver has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4bd8597dc36c376a2bb1ef2c72984615bdeb68de Mon Sep 17 00:00:00 2001 From: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Date: Wed, 30 Jan 2019 14:47:10 +0800 Subject: [PATCH] ASoC: mediatek: add btcvsd driver The driver function for transferring/receiving BT encoded data to/from BT firmware. Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/Kconfig | 9 + sound/soc/mediatek/common/Makefile | 2 + sound/soc/mediatek/common/mtk-btcvsd.c | 1364 ++++++++++++++++++++++++ 3 files changed, 1375 insertions(+) create mode 100644 sound/soc/mediatek/common/mtk-btcvsd.c diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 8bb360ee7234..b35410e4020e 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -115,3 +115,12 @@ config SND_SOC_MT8183 that can be used with other codecs. Select Y if you have such device. If unsure select "N". + +config SND_SOC_MTK_BTCVSD + tristate "ALSA BT SCO CVSD/MSBC Driver" + help + This is for software BTCVSD. This enable + the function for transferring/receiving + BT encoded data to/from BT firmware. + Select Y if you have such device. + If unsure select "N". diff --git a/sound/soc/mediatek/common/Makefile b/sound/soc/mediatek/common/Makefile index cdadabc5fd16..9ab90433a8d7 100644 --- a/sound/soc/mediatek/common/Makefile +++ b/sound/soc/mediatek/common/Makefile @@ -2,3 +2,5 @@ # platform driver snd-soc-mtk-common-objs := mtk-afe-platform-driver.o mtk-afe-fe-dai.o obj-$(CONFIG_SND_SOC_MEDIATEK) += snd-soc-mtk-common.o + +obj-$(CONFIG_SND_SOC_MTK_BTCVSD) += mtk-btcvsd.o \ No newline at end of file diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c new file mode 100644 index 000000000000..349a9120a0dd --- /dev/null +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -0,0 +1,1364 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Mediatek ALSA BT SCO CVSD/MSBC Driver +// +// Copyright (c) 2019 MediaTek Inc. +// Author: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> + +#include <linux/mfd/syscon.h> +#include <linux/module.h> +#include <linux/of_address.h> +#include <linux/sched/clock.h> + +#include <sound/soc.h> + +#define BTCVSD_SND_NAME "mtk-btcvsd-snd" + +#define BT_CVSD_TX_NREADY BIT(21) +#define BT_CVSD_RX_READY BIT(22) +#define BT_CVSD_TX_UNDERFLOW BIT(23) +#define BT_CVSD_RX_OVERFLOW BIT(24) +#define BT_CVSD_INTERRUPT BIT(31) + +#define BT_CVSD_CLEAR \ + (BT_CVSD_TX_NREADY | BT_CVSD_RX_READY | BT_CVSD_TX_UNDERFLOW |\ + BT_CVSD_RX_OVERFLOW | BT_CVSD_INTERRUPT) + +/* TX */ +#define SCO_TX_ENCODE_SIZE (60) +/* 18 = 6 * 180 / SCO_TX_ENCODE_SIZE */ +#define SCO_TX_PACKER_BUF_NUM (18) + +/* RX */ +#define SCO_RX_PLC_SIZE (30) +#define SCO_RX_PACKER_BUF_NUM (64) +#define SCO_RX_PACKET_MASK (0x3F) + +#define SCO_CVSD_PACKET_VALID_SIZE 2 + +#define SCO_PACKET_120 120 +#define SCO_PACKET_180 180 + +#define BTCVSD_RX_PACKET_SIZE (SCO_RX_PLC_SIZE + SCO_CVSD_PACKET_VALID_SIZE) +#define BTCVSD_TX_PACKET_SIZE (SCO_TX_ENCODE_SIZE) + +#define BTCVSD_RX_BUF_SIZE (BTCVSD_RX_PACKET_SIZE * SCO_RX_PACKER_BUF_NUM) +#define BTCVSD_TX_BUF_SIZE (BTCVSD_TX_PACKET_SIZE * SCO_TX_PACKER_BUF_NUM) + +enum bt_sco_state { + BT_SCO_STATE_IDLE, + BT_SCO_STATE_RUNNING, + BT_SCO_STATE_ENDING, +}; + +enum bt_sco_direct { + BT_SCO_DIRECT_BT2ARM, + BT_SCO_DIRECT_ARM2BT, +}; + +enum bt_sco_packet_len { + BT_SCO_CVSD_30 = 0, + BT_SCO_CVSD_60, + BT_SCO_CVSD_90, + BT_SCO_CVSD_120, + BT_SCO_CVSD_10, + BT_SCO_CVSD_20, + BT_SCO_CVSD_MAX, +}; + +enum BT_SCO_BAND { + BT_SCO_NB, + BT_SCO_WB, +}; + +struct mtk_btcvsd_snd_hw_info { + unsigned int num_valid_addr; + unsigned long bt_sram_addr[20]; + unsigned int packet_length; + unsigned int packet_num; +}; + +struct mtk_btcvsd_snd_stream { + struct snd_pcm_substream *substream; + int stream; + + enum bt_sco_state state; + + unsigned int packet_size; + unsigned int buf_size; + u8 temp_packet_buf[SCO_PACKET_180]; + + int packet_w; + int packet_r; + snd_pcm_uframes_t prev_frame; + int prev_packet_idx; + + unsigned int xrun:1; + unsigned int timeout:1; + unsigned int mute:1; + unsigned int trigger_start:1; + unsigned int wait_flag:1; + unsigned int rw_cnt; + + unsigned long long time_stamp; + unsigned long long buf_data_equivalent_time; + + struct mtk_btcvsd_snd_hw_info buffer_info; +}; + +struct mtk_btcvsd_snd { + struct device *dev; + int irq_id; + + struct regmap *infra; + void __iomem *bt_pkv_base; + void __iomem *bt_sram_bank2_base; + + unsigned int infra_misc_offset; + unsigned int conn_bt_cvsd_mask; + unsigned int cvsd_mcu_read_offset; + unsigned int cvsd_mcu_write_offset; + unsigned int cvsd_packet_indicator; + + u32 *bt_reg_pkt_r; + u32 *bt_reg_pkt_w; + u32 *bt_reg_ctl; + + unsigned int irq_disabled:1; + + spinlock_t tx_lock; /* spinlock for bt tx stream control */ + spinlock_t rx_lock; /* spinlock for bt rx stream control */ + wait_queue_head_t tx_wait; + wait_queue_head_t rx_wait; + + struct mtk_btcvsd_snd_stream *tx; + struct mtk_btcvsd_snd_stream *rx; + u8 tx_packet_buf[BTCVSD_TX_BUF_SIZE]; + u8 rx_packet_buf[BTCVSD_RX_BUF_SIZE]; + + enum BT_SCO_BAND band; +}; + +struct mtk_btcvsd_snd_time_buffer_info { + unsigned long long data_count_equi_time; + unsigned long long time_stamp_us; +}; + +static const unsigned int btsco_packet_valid_mask[BT_SCO_CVSD_MAX][6] = { + {0x1, 0x1 << 1, 0x1 << 2, 0x1 << 3, 0x1 << 4, 0x1 << 5}, + {0x1, 0x1, 0x2, 0x2, 0x4, 0x4}, + {0x1, 0x1, 0x1, 0x2, 0x2, 0x2}, + {0x1, 0x1, 0x1, 0x1, 0x0, 0x0}, + {0x7, 0x7 << 3, 0x7 << 6, 0x7 << 9, 0x7 << 12, 0x7 << 15}, + {0x3, 0x3 << 1, 0x3 << 3, 0x3 << 4, 0x3 << 6, 0x3 << 7}, +}; + +static const unsigned int btsco_packet_info[BT_SCO_CVSD_MAX][4] = { + {30, 6, SCO_PACKET_180 / SCO_TX_ENCODE_SIZE, + SCO_PACKET_180 / SCO_RX_PLC_SIZE}, + {60, 3, SCO_PACKET_180 / SCO_TX_ENCODE_SIZE, + SCO_PACKET_180 / SCO_RX_PLC_SIZE}, + {90, 2, SCO_PACKET_180 / SCO_TX_ENCODE_SIZE, + SCO_PACKET_180 / SCO_RX_PLC_SIZE}, + {120, 1, SCO_PACKET_120 / SCO_TX_ENCODE_SIZE, + SCO_PACKET_120 / SCO_RX_PLC_SIZE}, + {10, 18, SCO_PACKET_180 / SCO_TX_ENCODE_SIZE, + SCO_PACKET_180 / SCO_RX_PLC_SIZE}, + {20, 9, SCO_PACKET_180 / SCO_TX_ENCODE_SIZE, + SCO_PACKET_180 / SCO_RX_PLC_SIZE}, +}; + +static const u8 table_msbc_silence[SCO_PACKET_180] = { + 0x01, 0x38, 0xad, 0x00, 0x00, 0xc5, 0x00, 0x00, 0x00, 0x00, + 0x77, 0x6d, 0xb6, 0xdd, 0xdb, 0x6d, 0xb7, 0x76, 0xdb, 0x6d, + 0xdd, 0xb6, 0xdb, 0x77, 0x6d, 0xb6, 0xdd, 0xdb, 0x6d, 0xb7, + 0x76, 0xdb, 0x6d, 0xdd, 0xb6, 0xdb, 0x77, 0x6d, 0xb6, 0xdd, + 0xdb, 0x6d, 0xb7, 0x76, 0xdb, 0x6d, 0xdd, 0xb6, 0xdb, 0x77, + 0x6d, 0xb6, 0xdd, 0xdb, 0x6d, 0xb7, 0x76, 0xdb, 0x6c, 0x00, + 0x01, 0xc8, 0xad, 0x00, 0x00, 0xc5, 0x00, 0x00, 0x00, 0x00, + 0x77, 0x6d, 0xb6, 0xdd, 0xdb, 0x6d, 0xb7, 0x76, 0xdb, 0x6d, + 0xdd, 0xb6, 0xdb, 0x77, 0x6d, 0xb6, 0xdd, 0xdb, 0x6d, 0xb7, + 0x76, 0xdb, 0x6d, 0xdd, 0xb6, 0xdb, 0x77, 0x6d, 0xb6, 0xdd, + 0xdb, 0x6d, 0xb7, 0x76, 0xdb, 0x6d, 0xdd, 0xb6, 0xdb, 0x77, + 0x6d, 0xb6, 0xdd, 0xdb, 0x6d, 0xb7, 0x76, 0xdb, 0x6c, 0x00, + 0x01, 0xf8, 0xad, 0x00, 0x00, 0xc5, 0x00, 0x00, 0x00, 0x00, + 0x77, 0x6d, 0xb6, 0xdd, 0xdb, 0x6d, 0xb7, 0x76, 0xdb, 0x6d, + 0xdd, 0xb6, 0xdb, 0x77, 0x6d, 0xb6, 0xdd, 0xdb, 0x6d, 0xb7, + 0x76, 0xdb, 0x6d, 0xdd, 0xb6, 0xdb, 0x77, 0x6d, 0xb6, 0xdd, + 0xdb, 0x6d, 0xb7, 0x76, 0xdb, 0x6d, 0xdd, 0xb6, 0xdb, 0x77, + 0x6d, 0xb6, 0xdd, 0xdb, 0x6d, 0xb7, 0x76, 0xdb, 0x6c, 0x00 +}; + +static void mtk_btcvsd_snd_irq_enable(struct mtk_btcvsd_snd *bt) +{ + regmap_update_bits(bt->infra, bt->infra_misc_offset, + bt->conn_bt_cvsd_mask, bt->conn_bt_cvsd_mask); +} + +static void mtk_btcvsd_snd_irq_disable(struct mtk_btcvsd_snd *bt) +{ + regmap_update_bits(bt->infra, bt->infra_misc_offset, + bt->conn_bt_cvsd_mask, 0); +} + +static void mtk_btcvsd_snd_set_state(struct mtk_btcvsd_snd *bt, + struct mtk_btcvsd_snd_stream *bt_stream, + int state) +{ + dev_dbg(bt->dev, "%s(), stream %d, state %d, tx->state %d, rx->state %d, irq_disabled %d\n", + __func__, + bt_stream->stream, state, + bt->tx->state, bt->rx->state, bt->irq_disabled); + + bt_stream->state = state; + + if (bt->tx->state == BT_SCO_STATE_IDLE && + bt->rx->state == BT_SCO_STATE_IDLE) { + if (!bt->irq_disabled) { + disable_irq(bt->irq_id); + mtk_btcvsd_snd_irq_disable(bt); + bt->irq_disabled = 1; + } + } else { + if (bt->irq_disabled) { + enable_irq(bt->irq_id); + mtk_btcvsd_snd_irq_enable(bt); + bt->irq_disabled = 0; + } + } +} + +static int mtk_btcvsd_snd_tx_init(struct mtk_btcvsd_snd *bt) +{ + memset(bt->tx, 0, sizeof(*bt->tx)); + memset(bt->tx_packet_buf, 0, sizeof(bt->tx_packet_buf)); + + bt->tx->packet_size = BTCVSD_TX_PACKET_SIZE; + bt->tx->buf_size = BTCVSD_TX_BUF_SIZE; + bt->tx->timeout = 0; + bt->tx->rw_cnt = 0; + bt->tx->stream = SNDRV_PCM_STREAM_PLAYBACK; + return 0; +} + +static int mtk_btcvsd_snd_rx_init(struct mtk_btcvsd_snd *bt) +{ + memset(bt->rx, 0, sizeof(*bt->rx)); + memset(bt->rx_packet_buf, 0, sizeof(bt->rx_packet_buf)); + + bt->rx->packet_size = BTCVSD_RX_PACKET_SIZE; + bt->rx->buf_size = BTCVSD_RX_BUF_SIZE; + bt->rx->timeout = 0; + bt->rx->rw_cnt = 0; + bt->tx->stream = SNDRV_PCM_STREAM_CAPTURE; + return 0; +} + +static void get_tx_time_stamp(struct mtk_btcvsd_snd *bt, + struct mtk_btcvsd_snd_time_buffer_info *ts) +{ + ts->time_stamp_us = bt->tx->time_stamp; + ts->data_count_equi_time = bt->tx->buf_data_equivalent_time; +} + +static void get_rx_time_stamp(struct mtk_btcvsd_snd *bt, + struct mtk_btcvsd_snd_time_buffer_info *ts) +{ + ts->time_stamp_us = bt->rx->time_stamp; + ts->data_count_equi_time = bt->rx->buf_data_equivalent_time; +} + +static int btcvsd_bytes_to_frame(struct snd_pcm_substream *substream, + int bytes) +{ + int count = bytes; + struct snd_pcm_runtime *runtime = substream->runtime; + + if (runtime->format == SNDRV_PCM_FORMAT_S32_LE || + runtime->format == SNDRV_PCM_FORMAT_U32_LE) + count = count >> 2; + else + count = count >> 1; + + count = count / runtime->channels; + return count; +} + +static void mtk_btcvsd_snd_data_transfer(enum bt_sco_direct dir, + u8 *src, u8 *dst, + unsigned int blk_size, + unsigned int blk_num) +{ + unsigned int i, j; + + if (blk_size == 60 || blk_size == 120 || blk_size == 20) { + u32 *src_32 = (u32 *)src; + u32 *dst_32 = (u32 *)dst; + + for (i = 0; i < (blk_size * blk_num / 4); i++) + *dst_32++ = *src_32++; + } else { + u16 *src_16 = (u16 *)src; + u16 *dst_16 = (u16 *)dst; + + for (j = 0; j < blk_num; j++) { + for (i = 0; i < (blk_size / 2); i++) + *dst_16++ = *src_16++; + + if (dir == BT_SCO_DIRECT_BT2ARM) + src_16++; + else + dst_16++; + } + } +} + +/* write encoded mute data to bt sram */ +static int btcvsd_tx_clean_buffer(struct mtk_btcvsd_snd *bt) +{ + unsigned int i; + unsigned int num_valid_addr; + unsigned long flags; + enum BT_SCO_BAND band = bt->band; + + /* prepare encoded mute data */ + if (band == BT_SCO_NB) + memset(bt->tx->temp_packet_buf, 170, SCO_PACKET_180); + else + memcpy(bt->tx->temp_packet_buf, + table_msbc_silence, SCO_PACKET_180); + + /* write mute data to bt tx sram buffer */ + spin_lock_irqsave(&bt->tx_lock, flags); + num_valid_addr = bt->tx->buffer_info.num_valid_addr; + + dev_info(bt->dev, "%s(), band %d, num_valid_addr %u\n", + __func__, band, num_valid_addr); + + for (i = 0; i < num_valid_addr; i++) { + void *dst; + + dev_info(bt->dev, "%s(), clean addr 0x%lx\n", __func__, + bt->tx->buffer_info.bt_sram_addr[i]); + + dst = (void *)bt->tx->buffer_info.bt_sram_addr[i]; + + mtk_btcvsd_snd_data_transfer(BT_SCO_DIRECT_ARM2BT, + bt->tx->temp_packet_buf, dst, + bt->tx->buffer_info.packet_length, + bt->tx->buffer_info.packet_num); + } + spin_unlock_irqrestore(&bt->tx_lock, flags); + + return 0; +} + +static int mtk_btcvsd_read_from_bt(struct mtk_btcvsd_snd *bt, + enum bt_sco_packet_len packet_type, + unsigned int packet_length, + unsigned int packet_num, + unsigned int blk_size, + unsigned int control) +{ + unsigned int i; + int pv; + u8 *src; + unsigned int packet_buf_ofs; + unsigned long flags; + unsigned long connsys_addr_rx, ap_addr_rx; + + connsys_addr_rx = *bt->bt_reg_pkt_r; + ap_addr_rx = (unsigned long)bt->bt_sram_bank2_base + + (connsys_addr_rx & 0xFFFF); + + if (connsys_addr_rx == 0xdeadfeed) { + /* bt return 0xdeadfeed if read register during bt sleep */ + dev_warn(bt->dev, "%s(), connsys_addr_rx == 0xdeadfeed", + __func__); + return -EIO; + } + + src = (u8 *)ap_addr_rx; + + mtk_btcvsd_snd_data_transfer(BT_SCO_DIRECT_BT2ARM, src, + bt->rx->temp_packet_buf, packet_length, + packet_num); + + spin_lock_irqsave(&bt->rx_lock, flags); + for (i = 0; i < blk_size; i++) { + packet_buf_ofs = (bt->rx->packet_w & SCO_RX_PACKET_MASK) * + bt->rx->packet_size; + memcpy(bt->rx_packet_buf + packet_buf_ofs, + bt->rx->temp_packet_buf + (SCO_RX_PLC_SIZE * i), + SCO_RX_PLC_SIZE); + if ((control & btsco_packet_valid_mask[packet_type][i]) == + btsco_packet_valid_mask[packet_type][i]) + pv = 1; + else + pv = 0; + + packet_buf_ofs += SCO_RX_PLC_SIZE; + memcpy(bt->rx_packet_buf + packet_buf_ofs, (void *)&pv, + SCO_CVSD_PACKET_VALID_SIZE); + bt->rx->packet_w++; + } + spin_unlock_irqrestore(&bt->rx_lock, flags); + return 0; +} + +int mtk_btcvsd_write_to_bt(struct mtk_btcvsd_snd *bt, + enum bt_sco_packet_len packet_type, + unsigned int packet_length, + unsigned int packet_num, + unsigned int blk_size) +{ + unsigned int i; + unsigned long flags; + u8 *dst; + unsigned long connsys_addr_tx, ap_addr_tx; + bool new_ap_addr_tx = true; + + connsys_addr_tx = *bt->bt_reg_pkt_w; + ap_addr_tx = (unsigned long)bt->bt_sram_bank2_base + + (connsys_addr_tx & 0xFFFF); + + if (connsys_addr_tx == 0xdeadfeed) { + /* bt return 0xdeadfeed if read register during bt sleep */ + dev_warn(bt->dev, "%s(), connsys_addr_tx == 0xdeadfeed\n", + __func__); + return -EIO; + } + + spin_lock_irqsave(&bt->tx_lock, flags); + for (i = 0; i < blk_size; i++) { + memcpy(bt->tx->temp_packet_buf + (bt->tx->packet_size * i), + (bt->tx_packet_buf + + (bt->tx->packet_r % SCO_TX_PACKER_BUF_NUM) * + bt->tx->packet_size), + bt->tx->packet_size); + + bt->tx->packet_r++; + } + spin_unlock_irqrestore(&bt->tx_lock, flags); + + dst = (u8 *)ap_addr_tx; + + if (!bt->tx->mute) { + mtk_btcvsd_snd_data_transfer(BT_SCO_DIRECT_ARM2BT, + bt->tx->temp_packet_buf, dst, + packet_length, packet_num); + } + + /* store bt tx buffer sram info */ + bt->tx->buffer_info.packet_length = packet_length; + bt->tx->buffer_info.packet_num = packet_num; + for (i = 0; i < bt->tx->buffer_info.num_valid_addr; i++) { + if (bt->tx->buffer_info.bt_sram_addr[i] == ap_addr_tx) { + new_ap_addr_tx = false; + break; + } + } + if (new_ap_addr_tx) { + unsigned int next_idx; + + spin_lock_irqsave(&bt->tx_lock, flags); + bt->tx->buffer_info.num_valid_addr++; + next_idx = bt->tx->buffer_info.num_valid_addr - 1; + bt->tx->buffer_info.bt_sram_addr[next_idx] = ap_addr_tx; + spin_unlock_irqrestore(&bt->tx_lock, flags); + dev_info(bt->dev, "%s(), new ap_addr_tx = 0x%lx, num_valid_addr %d\n", + __func__, ap_addr_tx, + bt->tx->buffer_info.num_valid_addr); + } + + if (bt->tx->mute) + btcvsd_tx_clean_buffer(bt); + + return 0; +} + +static irqreturn_t mtk_btcvsd_snd_irq_handler(int irq_id, void *dev) +{ + struct mtk_btcvsd_snd *bt = dev; + unsigned int packet_type, packet_num, packet_length; + unsigned int buf_cnt_tx, buf_cnt_rx, control; + + if (bt->rx->state != BT_SCO_STATE_RUNNING && + bt->rx->state != BT_SCO_STATE_ENDING && + bt->tx->state != BT_SCO_STATE_RUNNING && + bt->tx->state != BT_SCO_STATE_ENDING) { + dev_warn(bt->dev, "%s(), in idle state: rx->state: %d, tx->state: %d\n", + __func__, bt->rx->state, bt->tx->state); + goto irq_handler_exit; + } + + control = *bt->bt_reg_ctl; + packet_type = (control >> 18) & 0x7; + + if (((control >> 31) & 1) == 0) { + dev_warn(bt->dev, "%s(), ((control >> 31) & 1) == 0, control 0x%x\n", + __func__, control); + goto irq_handler_exit; + } + + if (packet_type >= BT_SCO_CVSD_MAX) { + dev_warn(bt->dev, "%s(), invalid packet_type %u, exit\n", + __func__, packet_type); + goto irq_handler_exit; + } + + packet_length = btsco_packet_info[packet_type][0]; + packet_num = btsco_packet_info[packet_type][1]; + buf_cnt_tx = btsco_packet_info[packet_type][2]; + buf_cnt_rx = btsco_packet_info[packet_type][3]; + + if (bt->rx->state == BT_SCO_STATE_RUNNING || + bt->rx->state == BT_SCO_STATE_ENDING) { + if (bt->rx->xrun) { + if (bt->rx->packet_w - bt->rx->packet_r <= + SCO_RX_PACKER_BUF_NUM - 2 * buf_cnt_rx) { + /* + * free space is larger then + * twice interrupt rx data size + */ + bt->rx->xrun = 0; + dev_warn(bt->dev, "%s(), rx->xrun 0!\n", + __func__); + } + } + + if (!bt->rx->xrun && + (bt->rx->packet_w - bt->rx->packet_r <= + SCO_RX_PACKER_BUF_NUM - buf_cnt_rx)) { + mtk_btcvsd_read_from_bt(bt, + packet_type, + packet_length, + packet_num, + buf_cnt_rx, + control); + bt->rx->rw_cnt++; + } else { + bt->rx->xrun = 1; + dev_warn(bt->dev, "%s(), rx->xrun 1\n", __func__); + } + } + + /* tx */ + bt->tx->timeout = 0; + if ((bt->tx->state == BT_SCO_STATE_RUNNING || + bt->tx->state == BT_SCO_STATE_ENDING) && + bt->tx->trigger_start) { + if (bt->tx->xrun) { + /* prepared data is larger then twice + * interrupt tx data size + */ + if (bt->tx->packet_w - bt->tx->packet_r >= + 2 * buf_cnt_tx) { + bt->tx->xrun = 0; + dev_warn(bt->dev, "%s(), tx->xrun 0\n", + __func__); + } + } + + if ((!bt->tx->xrun && + (bt->tx->packet_w - bt->tx->packet_r >= buf_cnt_tx)) || + bt->tx->state == BT_SCO_STATE_ENDING) { + mtk_btcvsd_write_to_bt(bt, + packet_type, + packet_length, + packet_num, + buf_cnt_tx); + bt->tx->rw_cnt++; + } else { + bt->tx->xrun = 1; + dev_warn(bt->dev, "%s(), tx->xrun 1\n", __func__); + } + } + + *bt->bt_reg_ctl &= ~BT_CVSD_CLEAR; + + if (bt->rx->state == BT_SCO_STATE_RUNNING || + bt->rx->state == BT_SCO_STATE_ENDING) { + bt->rx->wait_flag = 1; + wake_up_interruptible(&bt->rx_wait); + snd_pcm_period_elapsed(bt->rx->substream); + } + if (bt->tx->state == BT_SCO_STATE_RUNNING || + bt->tx->state == BT_SCO_STATE_ENDING) { + bt->tx->wait_flag = 1; + wake_up_interruptible(&bt->tx_wait); + snd_pcm_period_elapsed(bt->tx->substream); + } + + return IRQ_HANDLED; +irq_handler_exit: + *bt->bt_reg_ctl &= ~BT_CVSD_CLEAR; + return IRQ_HANDLED; +} + +static int wait_for_bt_irq(struct mtk_btcvsd_snd *bt, + struct mtk_btcvsd_snd_stream *bt_stream) +{ + unsigned long long t1, t2; + /* one interrupt period = 22.5ms */ + unsigned long long timeout_limit = 22500000; + int max_timeout_trial = 2; + int ret; + + bt_stream->wait_flag = 0; + + while (max_timeout_trial && !bt_stream->wait_flag) { + t1 = sched_clock(); + if (bt_stream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = wait_event_interruptible_timeout(bt->tx_wait, + bt_stream->wait_flag, + nsecs_to_jiffies(timeout_limit)); + } else { + ret = wait_event_interruptible_timeout(bt->rx_wait, + bt_stream->wait_flag, + nsecs_to_jiffies(timeout_limit)); + } + + t2 = sched_clock(); + t2 = t2 - t1; /* in ns (10^9) */ + + if (t2 > timeout_limit) { + dev_warn(bt->dev, "%s(), stream %d, timeout %llu, limit %llu, ret %d, flag %d\n", + __func__, bt_stream->stream, + t2, timeout_limit, ret, + bt_stream->wait_flag); + } + + if (ret < 0) { + /* + * error, -ERESTARTSYS if it was interrupted by + * a signal + */ + dev_warn(bt->dev, "%s(), stream %d, error, trial left %d\n", + __func__, + bt_stream->stream, max_timeout_trial); + + bt_stream->timeout = 1; + return ret; + } else if (ret == 0) { + /* conidtion is false after timeout */ + max_timeout_trial--; + dev_warn(bt->dev, "%s(), stream %d, error, timeout, condition is false, trial left %d\n", + __func__, + bt_stream->stream, max_timeout_trial); + + if (max_timeout_trial <= 0) { + bt_stream->timeout = 1; + return -ETIME; + } + } + } + + return 0; +} + +ssize_t mtk_btcvsd_snd_read(struct mtk_btcvsd_snd *bt, + char __user *buf, + size_t count) +{ + ssize_t read_size = 0, read_count = 0, cur_read_idx, cont; + unsigned int cur_buf_ofs = 0; + unsigned long avail; + unsigned long flags; + unsigned int packet_size = bt->rx->packet_size; + + while (count) { + spin_lock_irqsave(&bt->rx_lock, flags); + /* available data in RX packet buffer */ + avail = (bt->rx->packet_w - bt->rx->packet_r) * packet_size; + + cur_read_idx = (bt->rx->packet_r & SCO_RX_PACKET_MASK) * + packet_size; + spin_unlock_irqrestore(&bt->rx_lock, flags); + + if (!avail) { + int ret = wait_for_bt_irq(bt, bt->rx); + + if (ret) + return read_count; + + continue; + } + + /* count must be multiple of packet_size */ + if (count % packet_size != 0 || + avail % packet_size != 0) { + dev_warn(bt->dev, "%s(), count %zu or d %lu is not multiple of packet_size %dd\n", + __func__, count, avail, packet_size); + + count -= count % packet_size; + avail -= avail % packet_size; + } + + if (count > avail) + read_size = avail; + else + read_size = count; + + /* calculate continue space */ + cont = bt->rx->buf_size - cur_read_idx; + if (read_size > cont) + read_size = cont; + + if (copy_to_user(buf + cur_buf_ofs, + bt->rx_packet_buf + cur_read_idx, + read_size)) { + dev_warn(bt->dev, "%s(), copy_to_user fail\n", + __func__); + return -EFAULT; + } + + spin_lock_irqsave(&bt->rx_lock, flags); + bt->rx->packet_r += read_size / packet_size; + spin_unlock_irqrestore(&bt->rx_lock, flags); + + read_count += read_size; + cur_buf_ofs += read_size; + count -= read_size; + } + + /* + * save current timestamp & buffer time in times_tamp and + * buf_data_equivalent_time + */ + bt->rx->time_stamp = sched_clock(); + bt->rx->buf_data_equivalent_time = + (unsigned long long)(bt->rx->packet_w - bt->rx->packet_r) * + SCO_RX_PLC_SIZE * 16 * 1000 / 2 / 64; + bt->rx->buf_data_equivalent_time += read_count * SCO_RX_PLC_SIZE * + 16 * 1000 / packet_size / 2 / 64; + /* return equivalent time(us) to data count */ + bt->rx->buf_data_equivalent_time *= 1000; + + return read_count; +} + +ssize_t mtk_btcvsd_snd_write(struct mtk_btcvsd_snd *bt, + char __user *buf, + size_t count) +{ + int written_size = count, avail = 0, cur_write_idx, write_size, cont; + unsigned int cur_buf_ofs = 0; + unsigned long flags; + unsigned int packet_size = bt->tx->packet_size; + + /* + * save current timestamp & buffer time in time_stamp and + * buf_data_equivalent_time + */ + bt->tx->time_stamp = sched_clock(); + bt->tx->buf_data_equivalent_time = + (unsigned long long)(bt->tx->packet_w - bt->tx->packet_r) * + packet_size * 16 * 1000 / 2 / 64; + + /* return equivalent time(us) to data count */ + bt->tx->buf_data_equivalent_time *= 1000; + + while (count) { + spin_lock_irqsave(&bt->tx_lock, flags); + /* free space of TX packet buffer */ + avail = bt->tx->buf_size - + (bt->tx->packet_w - bt->tx->packet_r) * packet_size; + + cur_write_idx = (bt->tx->packet_w % SCO_TX_PACKER_BUF_NUM) * + packet_size; + spin_unlock_irqrestore(&bt->tx_lock, flags); + + if (!avail) { + int ret = wait_for_bt_irq(bt, bt->rx); + + if (ret) + return written_size; + + continue; + } + + /* count must be multiple of bt->tx->packet_size */ + if (count % packet_size != 0 || + avail % packet_size != 0) { + dev_warn(bt->dev, "%s(), count %zu or avail %d is not multiple of packet_size %d\n", + __func__, count, avail, packet_size); + count -= count % packet_size; + avail -= avail % packet_size; + } + + if (count > avail) + write_size = avail; + else + write_size = count; + + /* calculate continue space */ + cont = bt->tx->buf_size - cur_write_idx; + if (write_size > cont) + write_size = cont; + + if (copy_from_user(bt->tx_packet_buf + + cur_write_idx, + buf + cur_buf_ofs, + write_size)) { + dev_warn(bt->dev, "%s(), copy_from_user fail\n", + __func__); + return -EFAULT; + } + + spin_lock_irqsave(&bt->tx_lock, flags); + bt->tx->packet_w += write_size / packet_size; + spin_unlock_irqrestore(&bt->tx_lock, flags); + cur_buf_ofs += write_size; + count -= write_size; + } + + return written_size; +} + +static struct mtk_btcvsd_snd_stream *get_bt_stream + (struct mtk_btcvsd_snd *bt, struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return bt->tx; + else + return bt->rx; +} + +/* pcm ops */ +static const struct snd_pcm_hardware mtk_btcvsd_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_RESUME), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .buffer_bytes_max = 24 * 1024, + .period_bytes_max = 24 * 1024, + .periods_min = 2, + .periods_max = 16, + .fifo_size = 0, +}; + +static int mtk_pcm_btcvsd_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, BTCVSD_SND_NAME); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component); + int ret; + + dev_dbg(bt->dev, "%s(), stream %d, substream %p\n", + __func__, substream->stream, substream); + + snd_soc_set_runtime_hwparams(substream, &mtk_btcvsd_hardware); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = mtk_btcvsd_snd_tx_init(bt); + bt->tx->substream = substream; + } else { + ret = mtk_btcvsd_snd_rx_init(bt); + bt->rx->substream = substream; + } + + return ret; +} + +static int mtk_pcm_btcvsd_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, BTCVSD_SND_NAME); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component); + struct mtk_btcvsd_snd_stream *bt_stream = get_bt_stream(bt, substream); + + dev_dbg(bt->dev, "%s(), stream %d\n", __func__, substream->stream); + + mtk_btcvsd_snd_set_state(bt, bt_stream, BT_SCO_STATE_IDLE); + bt_stream->substream = NULL; + return 0; +} + +static int mtk_pcm_btcvsd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, BTCVSD_SND_NAME); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + params_buffer_bytes(hw_params) % bt->tx->packet_size != 0) { + dev_warn(bt->dev, "%s(), error, buffer size %d not valid\n", + __func__, + params_buffer_bytes(hw_params)); + return -EINVAL; + } + + substream->runtime->dma_bytes = params_buffer_bytes(hw_params); + return 0; +} + +static int mtk_pcm_btcvsd_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, BTCVSD_SND_NAME); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + btcvsd_tx_clean_buffer(bt); + + return 0; +} + +static int mtk_pcm_btcvsd_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, BTCVSD_SND_NAME); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component); + struct mtk_btcvsd_snd_stream *bt_stream = get_bt_stream(bt, substream); + + dev_dbg(bt->dev, "%s(), stream %d\n", __func__, substream->stream); + + mtk_btcvsd_snd_set_state(bt, bt_stream, BT_SCO_STATE_RUNNING); + return 0; +} + +static int mtk_pcm_btcvsd_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, BTCVSD_SND_NAME); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component); + struct mtk_btcvsd_snd_stream *bt_stream = get_bt_stream(bt, substream); + int stream = substream->stream; + int hw_packet_ptr; + + dev_dbg(bt->dev, "%s(), stream %d, cmd %d\n", + __func__, substream->stream, cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + hw_packet_ptr = stream == SNDRV_PCM_STREAM_PLAYBACK ? + bt_stream->packet_r : bt_stream->packet_w; + bt_stream->prev_packet_idx = hw_packet_ptr; + bt_stream->prev_frame = 0; + bt_stream->trigger_start = 1; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + bt_stream->trigger_start = 0; + mtk_btcvsd_snd_set_state(bt, bt_stream, BT_SCO_STATE_ENDING); + return 0; + default: + return -EINVAL; + } +} + +static snd_pcm_uframes_t mtk_pcm_btcvsd_pointer + (struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, BTCVSD_SND_NAME); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component); + struct mtk_btcvsd_snd_stream *bt_stream; + snd_pcm_uframes_t frame = 0; + int byte = 0; + int hw_packet_ptr; + int packet_diff; + spinlock_t *lock; /* spinlock for bt stream control */ + unsigned long flags; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + lock = &bt->tx_lock; + bt_stream = bt->tx; + } else { + lock = &bt->rx_lock; + bt_stream = bt->rx; + } + + spin_lock_irqsave(lock, flags); + hw_packet_ptr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + bt->tx->packet_r : bt->rx->packet_w; + + /* get packet diff from last time */ + if (hw_packet_ptr >= bt_stream->prev_packet_idx) { + packet_diff = hw_packet_ptr - bt_stream->prev_packet_idx; + } else { + /* integer overflow */ + packet_diff = (INT_MAX - bt_stream->prev_packet_idx) + + (hw_packet_ptr - INT_MIN) + 1; + } + bt_stream->prev_packet_idx = hw_packet_ptr; + + /* increased bytes */ + byte = packet_diff * bt_stream->packet_size; + + frame = btcvsd_bytes_to_frame(substream, byte); + frame += bt_stream->prev_frame; + frame %= substream->runtime->buffer_size; + + bt_stream->prev_frame = frame; + + spin_unlock_irqrestore(lock, flags); + + return frame; +} + +static int mtk_pcm_btcvsd_copy(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + void __user *buf, unsigned long count) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, BTCVSD_SND_NAME); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mtk_btcvsd_snd_write(bt, buf, count); + else + mtk_btcvsd_snd_read(bt, buf, count); + + return 0; +} + +static struct snd_pcm_ops mtk_btcvsd_ops = { + .open = mtk_pcm_btcvsd_open, + .close = mtk_pcm_btcvsd_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = mtk_pcm_btcvsd_hw_params, + .hw_free = mtk_pcm_btcvsd_hw_free, + .prepare = mtk_pcm_btcvsd_prepare, + .trigger = mtk_pcm_btcvsd_trigger, + .pointer = mtk_pcm_btcvsd_pointer, + .copy_user = mtk_pcm_btcvsd_copy, +}; + +/* kcontrol */ +static const char *const btsco_band_str[] = {"NB", "WB"}; + +static const struct soc_enum btcvsd_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(btsco_band_str), btsco_band_str), +}; + +static int btcvsd_band_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + + ucontrol->value.integer.value[0] = bt->band; + return 0; +} + +static int btcvsd_band_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + if (ucontrol->value.enumerated.item[0] >= e->items) + return -EINVAL; + + bt->band = ucontrol->value.integer.value[0]; + dev_dbg(bt->dev, "%s(), band %d\n", __func__, bt->band); + return 0; +} + +static int btcvsd_tx_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + + if (!bt->tx) { + ucontrol->value.integer.value[0] = 0; + return 0; + } + + ucontrol->value.integer.value[0] = bt->tx->mute; + return 0; +} + +static int btcvsd_tx_mute_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + + if (!bt->tx) + return 0; + + bt->tx->mute = ucontrol->value.integer.value[0]; + return 0; +} + +static int btcvsd_rx_irq_received_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + + if (!bt->rx) + return 0; + + ucontrol->value.integer.value[0] = bt->rx->rw_cnt ? 1 : 0; + return 0; +} + +static int btcvsd_rx_timeout_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + + if (!bt->rx) + return 0; + + ucontrol->value.integer.value[0] = bt->rx->timeout; + bt->rx->timeout = 0; + return 0; +} + +static int btcvsd_rx_timestamp_get(struct snd_kcontrol *kcontrol, + unsigned int __user *data, unsigned int size) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + int ret = 0; + struct mtk_btcvsd_snd_time_buffer_info time_buffer_info_rx; + + if (size > sizeof(struct mtk_btcvsd_snd_time_buffer_info)) + return -EINVAL; + + get_rx_time_stamp(bt, &time_buffer_info_rx); + + dev_dbg(bt->dev, "%s(), time_stamp_us %llu, data_count_equi_time %llu", + __func__, + time_buffer_info_rx.time_stamp_us, + time_buffer_info_rx.data_count_equi_time); + + if (copy_to_user(data, &time_buffer_info_rx, + sizeof(struct mtk_btcvsd_snd_time_buffer_info))) { + dev_warn(bt->dev, "%s(), copy_to_user fail", __func__); + ret = -EFAULT; + } + + return ret; +} + +static int btcvsd_tx_irq_received_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + + if (!bt->tx) + return 0; + + ucontrol->value.integer.value[0] = bt->tx->rw_cnt ? 1 : 0; + return 0; +} + +static int btcvsd_tx_timeout_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + + ucontrol->value.integer.value[0] = bt->tx->timeout; + return 0; +} + +static int btcvsd_tx_timestamp_get(struct snd_kcontrol *kcontrol, + unsigned int __user *data, unsigned int size) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + int ret = 0; + struct mtk_btcvsd_snd_time_buffer_info time_buffer_info_tx; + + if (size > sizeof(struct mtk_btcvsd_snd_time_buffer_info)) + return -EINVAL; + + get_tx_time_stamp(bt, &time_buffer_info_tx); + + dev_dbg(bt->dev, "%s(), time_stamp_us %llu, data_count_equi_time %llu", + __func__, + time_buffer_info_tx.time_stamp_us, + time_buffer_info_tx.data_count_equi_time); + + if (copy_to_user(data, &time_buffer_info_tx, + sizeof(struct mtk_btcvsd_snd_time_buffer_info))) { + dev_warn(bt->dev, "%s(), copy_to_user fail", __func__); + ret = -EFAULT; + } + + return ret; +} + +static const struct snd_kcontrol_new mtk_btcvsd_snd_controls[] = { + SOC_ENUM_EXT("BTCVSD Band", btcvsd_enum[0], + btcvsd_band_get, btcvsd_band_set), + SOC_SINGLE_BOOL_EXT("BTCVSD Tx Mute Switch", 0, + btcvsd_tx_mute_get, btcvsd_tx_mute_set), + SOC_SINGLE_BOOL_EXT("BTCVSD Tx Irq Received Switch", 0, + btcvsd_tx_irq_received_get, NULL), + SOC_SINGLE_BOOL_EXT("BTCVSD Tx Timeout Switch", 0, + btcvsd_tx_timeout_get, NULL), + SOC_SINGLE_BOOL_EXT("BTCVSD Rx Irq Received Switch", 0, + btcvsd_rx_irq_received_get, NULL), + SOC_SINGLE_BOOL_EXT("BTCVSD Rx Timeout Switch", 0, + btcvsd_rx_timeout_get, NULL), + SND_SOC_BYTES_TLV("BTCVSD Rx Timestamp", + sizeof(struct mtk_btcvsd_snd_time_buffer_info), + btcvsd_rx_timestamp_get, NULL), + SND_SOC_BYTES_TLV("BTCVSD Tx Timestamp", + sizeof(struct mtk_btcvsd_snd_time_buffer_info), + btcvsd_tx_timestamp_get, NULL), +}; + +static int mtk_btcvsd_snd_component_probe(struct snd_soc_component *component) +{ + return snd_soc_add_component_controls(component, + mtk_btcvsd_snd_controls, + ARRAY_SIZE(mtk_btcvsd_snd_controls)); +} + +static const struct snd_soc_component_driver mtk_btcvsd_snd_platform = { + .name = BTCVSD_SND_NAME, + .ops = &mtk_btcvsd_ops, + .probe = mtk_btcvsd_snd_component_probe, +}; + +static int mtk_btcvsd_snd_probe(struct platform_device *pdev) +{ + int ret = 0; + int irq_id; + u32 offset[5] = {0, 0, 0, 0, 0}; + struct mtk_btcvsd_snd *btcvsd; + struct device *dev = &pdev->dev; + + /* init btcvsd private data */ + btcvsd = devm_kzalloc(dev, sizeof(*btcvsd), GFP_KERNEL); + if (!btcvsd) + return -ENOMEM; + platform_set_drvdata(pdev, btcvsd); + btcvsd->dev = dev; + + /* init tx/rx */ + btcvsd->rx = devm_kzalloc(btcvsd->dev, sizeof(*btcvsd->rx), GFP_KERNEL); + if (!btcvsd->rx) + return -ENOMEM; + + btcvsd->tx = devm_kzalloc(btcvsd->dev, sizeof(*btcvsd->tx), GFP_KERNEL); + if (!btcvsd->tx) + return -ENOMEM; + + spin_lock_init(&btcvsd->tx_lock); + spin_lock_init(&btcvsd->rx_lock); + + init_waitqueue_head(&btcvsd->tx_wait); + init_waitqueue_head(&btcvsd->rx_wait); + + mtk_btcvsd_snd_tx_init(btcvsd); + mtk_btcvsd_snd_rx_init(btcvsd); + + /* irq */ + irq_id = platform_get_irq(pdev, 0); + if (irq_id <= 0) { + dev_err(dev, "%s no irq found\n", dev->of_node->name); + return irq_id < 0 ? irq_id : -ENXIO; + } + + ret = devm_request_irq(dev, irq_id, mtk_btcvsd_snd_irq_handler, + IRQF_TRIGGER_LOW, "BTCVSD_ISR_Handle", + (void *)btcvsd); + if (ret) { + dev_err(dev, "could not request_irq for BTCVSD_ISR_Handle\n"); + return ret; + } + + btcvsd->irq_id = irq_id; + + /* iomap */ + btcvsd->bt_pkv_base = of_iomap(dev->of_node, 0); + if (!btcvsd->bt_pkv_base) { + dev_err(dev, "iomap bt_pkv_base fail\n"); + return -EIO; + } + + btcvsd->bt_sram_bank2_base = of_iomap(dev->of_node, 1); + if (!btcvsd->bt_sram_bank2_base) { + dev_err(dev, "iomap bt_sram_bank2_base fail\n"); + return -EIO; + } + + btcvsd->infra = syscon_regmap_lookup_by_phandle(dev->of_node, + "mediatek,infracfg"); + if (IS_ERR(btcvsd->infra)) { + dev_err(dev, "cannot find infra controller: %ld\n", + PTR_ERR(btcvsd->infra)); + return PTR_ERR(btcvsd->infra); + } + + /* get offset */ + ret = of_property_read_u32_array(dev->of_node, "mediatek,offset", + offset, + ARRAY_SIZE(offset)); + if (ret) { + dev_warn(dev, "%s(), get offest fail, ret %d\n", __func__, ret); + return ret; + } + btcvsd->infra_misc_offset = offset[0]; + btcvsd->conn_bt_cvsd_mask = offset[1]; + btcvsd->cvsd_mcu_read_offset = offset[2]; + btcvsd->cvsd_mcu_write_offset = offset[3]; + btcvsd->cvsd_packet_indicator = offset[4]; + + btcvsd->bt_reg_pkt_r = btcvsd->bt_pkv_base + + btcvsd->cvsd_mcu_read_offset; + btcvsd->bt_reg_pkt_w = btcvsd->bt_pkv_base + + btcvsd->cvsd_mcu_write_offset; + btcvsd->bt_reg_ctl = btcvsd->bt_pkv_base + + btcvsd->cvsd_packet_indicator; + + /* init state */ + mtk_btcvsd_snd_set_state(btcvsd, btcvsd->tx, BT_SCO_STATE_IDLE); + mtk_btcvsd_snd_set_state(btcvsd, btcvsd->rx, BT_SCO_STATE_IDLE); + + return devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform, + NULL, 0); +} + +static int mtk_btcvsd_snd_remove(struct platform_device *pdev) +{ + struct mtk_btcvsd_snd *btcvsd = dev_get_drvdata(&pdev->dev); + + iounmap(btcvsd->bt_pkv_base); + iounmap(btcvsd->bt_sram_bank2_base); + return 0; +} + +static const struct of_device_id mtk_btcvsd_snd_dt_match[] = { + { .compatible = "mediatek,mtk-btcvsd-snd", }, + {}, +}; +MODULE_DEVICE_TABLE(of, mtk_btcvsd_snd_dt_match); + +static struct platform_driver mtk_btcvsd_snd_driver = { + .driver = { + .name = "mtk-btcvsd-snd", + .of_match_table = mtk_btcvsd_snd_dt_match, + }, + .probe = mtk_btcvsd_snd_probe, + .remove = mtk_btcvsd_snd_remove, +}; + +module_platform_driver(mtk_btcvsd_snd_driver); + +MODULE_DESCRIPTION("Mediatek ALSA BT SCO CVSD/MSBC Driver"); +MODULE_AUTHOR("KaiChieh Chuang <kaichieh.chuang(a)mediatek.com>"); +MODULE_LICENSE("GPL v2"); -- 2.20.1

1 0

[alsa-devel] [PATCH 0/3] ASoC: DAPM/topology oops/use-after-free fixes
by Pierre-Louis Bossart 02 Feb '19

02 Feb '19

KASAN checks and more SOF stress-tests expose bugs that have apparently been there for a very long time. Guennadi Liakhovetski (1): ASoC: topology: fix oops/use-after-free case with dai driver Pierre-Louis Bossart (2): ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables ASoC: dapm: fix use-after-free issue with dailink sname sound/soc/soc-core.c | 2 +- sound/soc/soc-dapm.c | 45 +++++++++++++++++++++++++++++++++++++++- sound/soc/soc-topology.c | 5 +++++ 3 files changed, 50 insertions(+), 2 deletions(-) -- 2.17.1

2 6

[alsa-devel] [PATCH] ASoC: dapm: Add warnings for widget overwrite when adding route
by Zhiwei Jiang 02 Feb '19

02 Feb '19

Currently, in some complex cases, more than one widgets have same name and registed from differnt dapm context, and route add from another context too. When snd_soc_dapm_add_route, the previous registered widget will overwritten by the latest same name widget, will cause unexpect error. For Asoc framework we cant avoid this situation and we cant decide which widget that wanted with route. At least we can give users a notice. Signed-off-by: Zhiwei Jiang <qq282012236(a)gmail.com> --- sound/soc/soc-dapm.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..af45ffcf712d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2741,6 +2741,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, char prefixed_sink[80]; char prefixed_source[80]; const char *prefix; + unsigned int sink_ref = 0; + unsigned int source_ref = 0; int ret; prefix = soc_dapm_prefix(dapm); @@ -2774,6 +2776,11 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, if (wsource) break; } + sink_ref++; + if (sink_ref > 1) + dev_warn(dapm->dev, + "ASoC: sink widget %s overwritten\n", + w->name); continue; } if (!wsource && !(strcmp(w->name, source))) { @@ -2783,6 +2790,11 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, if (wsink) break; } + source_ref++; + if (source_ref > 1) + dev_warn(dapm->dev, + "ASoC: source widget %s overwritten\n", + w->name); } } /* use widget from another DAPM context if not found from this */ -- 2.19.0

2 1

[alsa-devel] [PATCH 2/2] ASoC: cs35l36: Add device tree documentation for CS35L36
by James Schulman 02 Feb '19

02 Feb '19

Add device tree documentation for Cirrus Logic CS35L36 speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- .../devicetree/bindings/sound/cs35l36.txt | 168 +++++++++++++++++++++ 1 file changed, 168 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l36.txt diff --git a/Documentation/devicetree/bindings/sound/cs35l36.txt b/Documentation/devicetree/bindings/sound/cs35l36.txt new file mode 100644 index 0000000..ccfeaf6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs35l36.txt @@ -0,0 +1,168 @@ +CS35L36 Speaker Amplifier + +Required properties: + + - compatible : "cirrus,cs35l36" + + - reg : the I2C address of the device for I2C + + - VA-supply, VP-supply : power supplies for the device, + as covered in + Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + - cirrus,multi-amp-mode : Boolean to determine if there are more than + one amplifier in the system. If more than one it is best to Hi-Z the ASP + port to prevent bus contention on the output signal + + - cirrus,boost-ctl-millivolt : Boost Voltage Value. Configures the boost + converter's output voltage in mV. The range is from 2550mV to 12000mV with + increments of 50mV. + (Default) VP + + - cirrus,boost-ctl-select : Boost conerter control source selection. + Selects the source of the BST_CTL target VBST voltage for the boost + converter to generate. + 0x00 - Control Port Value + 0x01 - Class H Tracking (Default) + 0x10 - MultiDevice Sync Value + + - cirrus,boost-peak-milliamp : Boost-converter peak current limit in mA. + Configures the peak current by monitoring the current through the boost FET. + Range starts at 1600mA and goes to a maximum of 4500mA with increments of + 50mA. + (Default) 4.50 Amps + + - cirrus,boost-ind-nanohenry : Inductor estimation LBST reference value. + Seeds the digital boost converter's inductor estimation block with the initial + inductance value to reference. + + 1000 = 1uH (Default) + 1200 = 1.2uH + + - cirrus,amp-pcm-inv : Boolean to determine Amplifier will invert incoming + PCM data + + - cirrus,imon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound IMON feedback data + + - cirrus,vmon-pol-inv : Boolean to determine Amplifier will invert the + polarity of outbound VMON feedback data + + - cirrus,dcm-mode-enable : Boost converter automatic DCM Mode enable. + This enables the digital boost converter to operate in a low power + (Discontinuous Conduction) mode during low loading conditions. + + - cirrus,weak-fet-disable : Boolean : The strength of the output drivers is + reduced when operating in a Weak-FET Drive Mode and must not be used to drive + a large load. + + - cirrus,classh-wk-fet-delay : Weak-FET entry delay. Controls the delay + (in ms) before the Class H algorithm switches to the weak-FET voltage + (after the audio falls and remains below the value specified in WKFET_AMP_THLD). + + 0 = 0ms + 1 = 5ms + 2 = 10ms + 3 = 50ms + 4 = 100ms (Default) + 5 = 200ms + 6 = 500ms + 7 = 1000ms + + - cirrus,classh-weak-fet-thld-millivolt : Weak-FET amplifier drive threshold. + Configures the signal threshold at which the PWM output stage enters + weak-FET operation. The range is 50mV to 700mV in 50mV increments. + + - cirrus,temp-warn-threshold : Amplifier overtemperature warning threshold. + Configures the threshold at which the overtemperature warning condition occurs. + When the threshold is met, the overtemperature warning attenuation is applied + and the TEMP_WARN_EINT interrupt status bit is set. + If TEMP_WARN_MASK = 0, INTb is asserted. + + 0 = 105C + 1 = 115C + 2 = 125C (Default) + 3 = 135C + + - cirrus,irq-drive-select : Selects the driver type of the selected interrupt + output. + + 0 = Open-drain + 1 = Push-pull (Default) + + - cirrus,irq-gpio-select : Selects the pin to serve as the programmable + interrupt output. + + 0 = PDM_DATA / SWIRE_SD / INT (Default) + 1 = GPIO + +Optional properties for the "cirrus,vpbr-config" Sub-node + + - cirrus,vpbr-en : VBST brownout prevention enable. Configures whether the + VBST brownout prevention algorithm is enabled or disabled. + + 0 = VBST brownout prevention disabled (default) + 1 = VBST brownout prevention enabled + + See Section 7.31.1 VPBR Config for configuration options & further details + + - cirrus,vpbr-thld : Initial VPBR threshold. Configures the VP brownout + threshold voltage + + - cirrus,cirrus,vpbr-atk-rate : Attenuation attack step rate. Configures the + amount delay between consecutive volume attenuation steps when a brownout + condition is present and the VP brownout condition is in an attacking state. + + - cirrus,vpbr-atk-vol : VP brownout prevention step size. Configures the VP + brownout prevention attacking attenuation step size when operating in either + digital volume or analog gain modes. + + - cirrus,vpbr-max-attn : Maximum attenuation that the VP brownout prevention + can apply to the audio signal. + + - cirrus,vpbr-wait : Configures the delay time between a brownout condition + no longer being present and the VP brownout prevention entering an attenuation + release state. + + - cirrus,vpbr-rel-rate : Attenuation release step rate. Configures the delay + between consecutive volume attenuation release steps when a brownout condition + is not longer present and the VP brownout is in an attenuation release state. + + - cirrus,vpbr-mute-en : During the attack state, if the vpbr-max-attn value + is reached, the error condition still remains, and this bit is set, the audio + is muted. + +Example: + +cs35l36: cs35l36@40 { + compatible = "cirrus,cs35l36"; + reg = <0x40>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio0 54 0>; + interrupt-parent = <&gpio8>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-ctl-millivolt = <10000>; + cirrus,boost-ctl-select = <0x00>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,weak-fet-delay = <0x04>; + cirrus,weak-fet-thld = <0x01>; + cirrus,temp-warn-threshold = <0x01>; + cirrus,multi-amp-mode; + cirrus,irq-drive-select = <0x01>; + cirrus,irq-gpio-select = <0x01>; + + cirrus,vpbr-config { + cirrus,vpbr-en = <0x00>; + cirrus,vpbr-thld = <0x05>; + cirrus,vpbr-atk-rate = <0x02>; + cirrus,vpbr-atk-vol = <0x01>; + cirrus,vpbr-max-attn = <0x09>; + cirrus,vpbr-wait = <0x01>; + cirrus,vpbr-rel-rate = <0x05>; + cirrus,vpbr-mute-en = <0x00>; + }; +}; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier
by James Schulman 02 Feb '19

02 Feb '19

Add driver support for Cirrus Logic CS35L36 boosted speaker amplifier Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs35l36.h | 44 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l36.c | 1959 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l36.h | 446 ++++++++++ 5 files changed, 2456 insertions(+) create mode 100644 include/sound/cs35l36.h create mode 100644 sound/soc/codecs/cs35l36.c create mode 100644 sound/soc/codecs/cs35l36.h diff --git a/include/sound/cs35l36.h b/include/sound/cs35l36.h new file mode 100644 index 0000000..1ee7527 --- /dev/null +++ b/include/sound/cs35l36.h @@ -0,0 +1,44 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * linux/sound/cs35l36.h -- Platform data for CS35L36 + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H +#define __CS35L36_H + +struct cs35l36_vpbr_cfg { + bool is_present; + bool vpbr_en; + int vpbr_thld; + int vpbr_atk_rate; + int vpbr_atk_vol; + int vpbr_max_attn; + int vpbr_wait; + int vpbr_rel_rate; + int vpbr_mute_en; +}; + +struct cs35l36_platform_data { + bool multi_amp_mode; + bool dcm_mode; + int ldm_mode_sel; + bool amp_pcm_inv; + bool imon_pol_inv; + bool vmon_pol_inv; + int boost_ind; + int bst_vctl; + int bst_vctl_sel; + int bst_ipk; + bool extern_boost; + int temp_warn_thld; + int irq_drv_sel; + int irq_gpio_sel; + struct cs35l36_vpbr_cfg vpbr_config; +}; + +#endif /* __CS35L36_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a15710c..3946222 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,6 +55,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS35L33 if I2C select SND_SOC_CS35L34 if I2C select SND_SOC_CS35L35 if I2C + select SND_SOC_CS35L36 if I2C select SND_SOC_CS42L42 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT @@ -483,6 +484,10 @@ config SND_SOC_CS35L35 tristate "Cirrus Logic CS35L35 CODEC" depends on I2C +config SND_SOC_CS35L36 + tristate "Cirrus Logic CS35L36 CODEC" + depends on I2C + config SND_SOC_CS42L42 tristate "Cirrus Logic CS42L42 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d7a597..427b180 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -47,6 +47,7 @@ snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o snd-soc-cs35l35-objs := cs35l35.o +snd-soc-cs35l36-objs := cs35l36.o snd-soc-cs42l42-objs := cs42l42.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o @@ -318,6 +319,7 @@ obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o obj-$(CONFIG_SND_SOC_CS35L35) += snd-soc-cs35l35.o +obj-$(CONFIG_SND_SOC_CS35L36) += snd-soc-cs35l36.o obj-$(CONFIG_SND_SOC_CS42L42) += snd-soc-cs42l42.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c new file mode 100644 index 0000000..955b90e --- /dev/null +++ b/sound/soc/codecs/cs35l36.c @@ -0,0 +1,1959 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.c -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/cs35l36.h> +#include <linux/of_irq.h> +#include <linux/completion.h> + +#include "cs35l36.h" + +/* + * Some fields take zero as a valid value so use a high bit flag that won't + * get written to the device to mark those. + */ +#define CS35L36_VALID_PDATA 0x80000000 + +static const char * const cs35l36_supplies[] = { + "VA", + "VP", +}; + +struct cs35l36_private { + struct device *dev; + struct cs35l36_platform_data pdata; + struct regmap *regmap; + struct regulator_bulk_data supplies[2]; + int num_supplies; + int clksrc; + int chip_version; + int rev_id; + struct gpio_desc *reset_gpio; +}; + +struct cs35l36_pll_config { + int freq; + int clk_cfg; + int fll_igain; +}; + +static const struct cs35l36_pll_config cs35l36_pll_sysclk[] = { + {32768, 0x00, 0x05}, + {8000, 0x01, 0x03}, + {11025, 0x02, 0x03}, + {12000, 0x03, 0x03}, + {16000, 0x04, 0x04}, + {22050, 0x05, 0x04}, + {24000, 0x06, 0x04}, + {32000, 0x07, 0x05}, + {44100, 0x08, 0x05}, + {48000, 0x09, 0x05}, + {88200, 0x0A, 0x06}, + {96000, 0x0B, 0x06}, + {128000, 0x0C, 0x07}, + {176400, 0x0D, 0x07}, + {192000, 0x0E, 0x07}, + {256000, 0x0F, 0x08}, + {352800, 0x10, 0x08}, + {384000, 0x11, 0x08}, + {512000, 0x12, 0x09}, + {705600, 0x13, 0x09}, + {750000, 0x14, 0x09}, + {768000, 0x15, 0x09}, + {1000000, 0x16, 0x0A}, + {1024000, 0x17, 0x0A}, + {1200000, 0x18, 0x0A}, + {1411200, 0x19, 0x0A}, + {1500000, 0x1A, 0x0A}, + {1536000, 0x1B, 0x0A}, + {2000000, 0x1C, 0x0A}, + {2048000, 0x1D, 0x0A}, + {2400000, 0x1E, 0x0A}, + {2822400, 0x1F, 0x0A}, + {3000000, 0x20, 0x0A}, + {3072000, 0x21, 0x0A}, + {3200000, 0x22, 0x0A}, + {4000000, 0x23, 0x0A}, + {4096000, 0x24, 0x0A}, + {4800000, 0x25, 0x0A}, + {5644800, 0x26, 0x0A}, + {6000000, 0x27, 0x0A}, + {6144000, 0x28, 0x0A}, + {6250000, 0x29, 0x08}, + {6400000, 0x2A, 0x0A}, + {6500000, 0x2B, 0x08}, + {6750000, 0x2C, 0x09}, + {7526400, 0x2D, 0x0A}, + {8000000, 0x2E, 0x0A}, + {8192000, 0x2F, 0x0A}, + {9600000, 0x30, 0x0A}, + {11289600, 0x31, 0x0A}, + {12000000, 0x32, 0x0A}, + {12288000, 0x33, 0x0A}, + {12500000, 0x34, 0x08}, + {12800000, 0x35, 0x0A}, + {13000000, 0x36, 0x0A}, + {13500000, 0x37, 0x0A}, + {19200000, 0x38, 0x0A}, + {22579200, 0x39, 0x0A}, + {24000000, 0x3A, 0x0A}, + {24576000, 0x3B, 0x0A}, + {25000000, 0x3C, 0x0A}, + {25600000, 0x3D, 0x0A}, + {26000000, 0x3E, 0x0A}, + {27000000, 0x3F, 0x0A}, +}; + +struct reg_default cs35l36_reg[] = { + {CS35L36_TESTKEY_CTRL, 0x00000000}, + {CS35L36_USERKEY_CTL, 0x00000000}, + {CS35L36_OTP_CTRL1, 0x00002460}, + {CS35L36_OTP_CTRL2, 0x00000000}, + {CS35L36_OTP_CTRL3, 0x00000000}, + {CS35L36_OTP_CTRL4, 0x00000000}, + {CS35L36_OTP_CTRL5, 0x00000000}, + {CS35L36_PAC_CTL1, 0x00000004}, + {CS35L36_PAC_CTL2, 0x00000000}, + {CS35L36_PAC_CTL3, 0x00000000}, + {CS35L36_PWR_CTRL1, 0x00000000}, + {CS35L36_PWR_CTRL2, 0x00003321}, + {CS35L36_PWR_CTRL3, 0x01000010}, + {CS35L36_CTRL_OVRRIDE, 0x00000002}, + {CS35L36_AMP_OUT_MUTE, 0x00000000}, + {CS35L36_OTP_TRIM_STATUS, 0x00000000}, + {CS35L36_DISCH_FILT, 0x00000000}, + {CS35L36_PROTECT_REL_ERR, 0x00000000}, + {CS35L36_PAD_INTERFACE, 0x00000038}, + {CS35L36_PLL_CLK_CTRL, 0x00000010}, + {CS35L36_GLOBAL_CLK_CTRL, 0x00000003}, + {CS35L36_ADC_CLK_CTRL, 0x00000000}, + {CS35L36_SWIRE_CLK_CTRL, 0x00000000}, + {CS35L36_SP_SCLK_CLK_CTRL, 0x00000000}, + {CS35L36_MDSYNC_EN, 0x00000000}, + {CS35L36_MDSYNC_TX_ID, 0x00000000}, + {CS35L36_MDSYNC_PWR_CTRL, 0x00000000}, + {CS35L36_MDSYNC_DATA_TX, 0x00000000}, + {CS35L36_MDSYNC_TX_STATUS, 0x00000002}, + {CS35L36_MDSYNC_RX_STATUS, 0x00000000}, + {CS35L36_MDSYNC_ERR_STATUS, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL1, 0x00000000}, + {CS35L36_BSTCVRT_VCTRL2, 0x00000001}, + {CS35L36_BSTCVRT_PEAK_CUR, 0x0000004A}, + {CS35L36_BSTCVRT_SFT_RAMP, 0x00000003}, + {CS35L36_BSTCVRT_COEFF, 0x00002424}, + {CS35L36_BSTCVRT_SLOPE_LBST, 0x00005800}, + {CS35L36_BSTCVRT_SW_FREQ, 0x00010000}, + {CS35L36_BSTCVRT_DCM_CTRL, 0x00002001}, + {CS35L36_BSTCVRT_DCM_MODE_FORCE, 0x00000000}, + {CS35L36_BSTCVRT_OVERVOLT_CTRL, 0x00000130}, + {CS35L36_VPI_LIMIT_MODE, 0x00000000}, + {CS35L36_VPI_LIMIT_MINMAX, 0x00003000}, + {CS35L36_VPI_VP_THLD, 0x00101010}, + {CS35L36_VPI_TRACK_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_MODE_CTRL, 0x00000000}, + {CS35L36_VPI_TRIG_STEPS, 0x00000000}, + {CS35L36_VI_SPKMON_FILT, 0x00000003}, + {CS35L36_VI_SPKMON_GAIN, 0x00000909}, + {CS35L36_VI_SPKMON_IP_SEL, 0x00000000}, + {CS35L36_DTEMP_WARN_THLD, 0x00000002}, + {CS35L36_DTEMP_STATUS, 0x00000000}, + {CS35L36_VPVBST_FS_SEL, 0x00000001}, + {CS35L36_VPVBST_VP_CTRL, 0x000001C0}, + {CS35L36_VPVBST_VBST_CTRL, 0x000001C0}, + {CS35L36_ASP_TX_PIN_CTRL, 0x00000028}, + {CS35L36_ASP_RATE_CTRL, 0x00090000}, + {CS35L36_ASP_FORMAT, 0x00000002}, + {CS35L36_ASP_FRAME_CTRL, 0x00180018}, + {CS35L36_ASP_TX1_TX2_SLOT, 0x00010000}, + {CS35L36_ASP_TX3_TX4_SLOT, 0x00030002}, + {CS35L36_ASP_TX5_TX6_SLOT, 0x00050004}, + {CS35L36_ASP_TX7_TX8_SLOT, 0x00070006}, + {CS35L36_ASP_RX1_SLOT, 0x00000000}, + {CS35L36_ASP_RX_TX_EN, 0x00000000}, + {CS35L36_ASP_RX1_SEL, 0x00000008}, + {CS35L36_ASP_TX1_SEL, 0x00000018}, + {CS35L36_ASP_TX2_SEL, 0x00000019}, + {CS35L36_ASP_TX3_SEL, 0x00000028}, + {CS35L36_ASP_TX4_SEL, 0x00000029}, + {CS35L36_ASP_TX5_SEL, 0x00000020}, + {CS35L36_ASP_TX6_SEL, 0x00000000}, + {CS35L36_SWIRE_P1_TX1_SEL, 0x00000018}, + {CS35L36_SWIRE_P1_TX2_SEL, 0x00000019}, + {CS35L36_SWIRE_P2_TX1_SEL, 0x00000028}, + {CS35L36_SWIRE_P2_TX2_SEL, 0x00000029}, + {CS35L36_SWIRE_P2_TX3_SEL, 0x00000020}, + {CS35L36_SWIRE_DP1_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP2_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_DP3_FIFO_CFG, 0x0000001B}, + {CS35L36_SWIRE_PCM_RX_DATA, 0x00000000}, + {CS35L36_SWIRE_FS_SEL, 0x00000001}, + {CS35L36_AMP_DIG_VOL_CTRL, 0x00008000}, + {CS35L36_VPBR_CFG, 0x02AA1905}, + {CS35L36_VBBR_CFG, 0x02AA1905}, + {CS35L36_VPBR_STATUS, 0x00000000}, + {CS35L36_VBBR_STATUS, 0x00000000}, + {CS35L36_OVERTEMP_CFG, 0x00000001}, + {CS35L36_AMP_ERR_VOL, 0x00000000}, + {CS35L36_CLASSH_CFG, 0x000B0405}, + {CS35L36_CLASSH_FET_DRV_CFG, 0x00000111}, + {CS35L36_NG_CFG, 0x00000033}, + {CS35L36_AMP_GAIN_CTRL, 0x00000273}, + {CS35L36_PWM_MOD_IO_CTRL, 0x00000000}, + {CS35L36_PWM_MOD_STATUS, 0x00000000}, + {CS35L36_DAC_MSM_CFG, 0x00000000}, + {CS35L36_AMP_SLOPE_CTRL, 0x00000B00}, + {CS35L36_AMP_PDM_VOLUME, 0x00000000}, + {CS35L36_AMP_PDM_RATE_CTRL, 0x00000000}, + {CS35L36_PDM_CH_SEL, 0x00000000}, + {CS35L36_AMP_NG_CTRL, 0x0000212F}, + {CS35L36_PDM_HIGHFILT_CTRL, 0x00000000}, + {CS35L36_PAC_INT0_CTRL, 0x00000001}, + {CS35L36_PAC_INT1_CTRL, 0x00000001}, + {CS35L36_PAC_INT2_CTRL, 0x00000001}, + {CS35L36_PAC_INT3_CTRL, 0x00000001}, + {CS35L36_PAC_INT4_CTRL, 0x00000001}, + {CS35L36_PAC_INT5_CTRL, 0x00000001}, + {CS35L36_PAC_INT6_CTRL, 0x00000001}, + {CS35L36_PAC_INT7_CTRL, 0x00000001}, +}; + +bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_OTP_MEM30: + case CS35L36_OTP_CTRL1: + case CS35L36_OTP_CTRL2: + case CS35L36_OTP_CTRL3: + case CS35L36_OTP_CTRL4: + case CS35L36_OTP_CTRL5: + case CS35L36_PAC_CTL1: + case CS35L36_PAC_CTL2: + case CS35L36_PAC_CTL3: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_PWR_CTRL1: + case CS35L36_PWR_CTRL2: + case CS35L36_PWR_CTRL3: + case CS35L36_CTRL_OVRRIDE: + case CS35L36_AMP_OUT_MUTE: + case CS35L36_OTP_TRIM_STATUS: + case CS35L36_DISCH_FILT: + case CS35L36_PROTECT_REL_ERR: + case CS35L36_PAD_INTERFACE: + case CS35L36_PLL_CLK_CTRL: + case CS35L36_GLOBAL_CLK_CTRL: + case CS35L36_ADC_CLK_CTRL: + case CS35L36_SWIRE_CLK_CTRL: + case CS35L36_SP_SCLK_CLK_CTRL: + case CS35L36_TST_FS_MON0: + case CS35L36_MDSYNC_EN: + case CS35L36_MDSYNC_TX_ID: + case CS35L36_MDSYNC_PWR_CTRL: + case CS35L36_MDSYNC_DATA_TX: + case CS35L36_MDSYNC_TX_STATUS: + case CS35L36_MDSYNC_RX_STATUS: + case CS35L36_MDSYNC_ERR_STATUS: + case CS35L36_BSTCVRT_VCTRL1: + case CS35L36_BSTCVRT_VCTRL2: + case CS35L36_BSTCVRT_PEAK_CUR: + case CS35L36_BSTCVRT_SFT_RAMP: + case CS35L36_BSTCVRT_COEFF: + case CS35L36_BSTCVRT_SLOPE_LBST: + case CS35L36_BSTCVRT_SW_FREQ: + case CS35L36_BSTCVRT_DCM_CTRL: + case CS35L36_BSTCVRT_DCM_MODE_FORCE: + case CS35L36_BSTCVRT_OVERVOLT_CTRL: + case CS35L36_BST_TST_MANUAL: + case CS35L36_BST_ANA2_TEST: + case CS35L36_VPI_LIMIT_MODE: + case CS35L36_VPI_LIMIT_MINMAX: + case CS35L36_VPI_VP_THLD: + case CS35L36_VPI_TRACK_CTRL: + case CS35L36_VPI_TRIG_MODE_CTRL: + case CS35L36_VPI_TRIG_STEPS: + case CS35L36_VI_SPKMON_FILT: + case CS35L36_VI_SPKMON_GAIN: + case CS35L36_VI_SPKMON_IP_SEL: + case CS35L36_DTEMP_WARN_THLD: + case CS35L36_DTEMP_STATUS: + case CS35L36_VPVBST_FS_SEL: + case CS35L36_VPVBST_VP_CTRL: + case CS35L36_VPVBST_VBST_CTRL: + case CS35L36_ASP_TX_PIN_CTRL: + case CS35L36_ASP_RATE_CTRL: + case CS35L36_ASP_FORMAT: + case CS35L36_ASP_FRAME_CTRL: + case CS35L36_ASP_TX1_TX2_SLOT: + case CS35L36_ASP_TX3_TX4_SLOT: + case CS35L36_ASP_TX5_TX6_SLOT: + case CS35L36_ASP_TX7_TX8_SLOT: + case CS35L36_ASP_RX1_SLOT: + case CS35L36_ASP_RX_TX_EN: + case CS35L36_ASP_RX1_SEL: + case CS35L36_ASP_TX1_SEL: + case CS35L36_ASP_TX2_SEL: + case CS35L36_ASP_TX3_SEL: + case CS35L36_ASP_TX4_SEL: + case CS35L36_ASP_TX5_SEL: + case CS35L36_ASP_TX6_SEL: + case CS35L36_SWIRE_P1_TX1_SEL: + case CS35L36_SWIRE_P1_TX2_SEL: + case CS35L36_SWIRE_P2_TX1_SEL: + case CS35L36_SWIRE_P2_TX2_SEL: + case CS35L36_SWIRE_P2_TX3_SEL: + case CS35L36_SWIRE_DP1_FIFO_CFG: + case CS35L36_SWIRE_DP2_FIFO_CFG: + case CS35L36_SWIRE_DP3_FIFO_CFG: + case CS35L36_SWIRE_PCM_RX_DATA: + case CS35L36_SWIRE_FS_SEL: + case CS35L36_AMP_DIG_VOL_CTRL: + case CS35L36_VPBR_CFG: + case CS35L36_VBBR_CFG: + case CS35L36_VPBR_STATUS: + case CS35L36_VBBR_STATUS: + case CS35L36_OVERTEMP_CFG: + case CS35L36_AMP_ERR_VOL: + case CS35L36_CLASSH_CFG: + case CS35L36_CLASSH_FET_DRV_CFG: + case CS35L36_NG_CFG: + case CS35L36_AMP_GAIN_CTRL: + case CS35L36_PWM_MOD_IO_CTRL: + case CS35L36_PWM_MOD_STATUS: + case CS35L36_DAC_MSM_CFG: + case CS35L36_AMP_SLOPE_CTRL: + case CS35L36_AMP_PDM_VOLUME: + case CS35L36_AMP_PDM_RATE_CTRL: + case CS35L36_PDM_CH_SEL: + case CS35L36_AMP_NG_CTRL: + case CS35L36_PDM_HIGHFILT_CTRL: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + case CS35L36_PAC_INT0_CTRL: + case CS35L36_PAC_INT1_CTRL: + case CS35L36_PAC_INT2_CTRL: + case CS35L36_PAC_INT3_CTRL: + case CS35L36_PAC_INT4_CTRL: + case CS35L36_PAC_INT5_CTRL: + case CS35L36_PAC_INT6_CTRL: + case CS35L36_PAC_INT7_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +bool cs35l36_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_TST_FS_MON0: + return true; + default: + return false; + } +} + +bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L36_SW_RESET: + case CS35L36_SW_REV: + case CS35L36_HW_REV: + case CS35L36_TESTKEY_CTRL: + case CS35L36_USERKEY_CTL: + case CS35L36_DEVICE_ID: + case CS35L36_FAB_ID: + case CS35L36_REV_ID: + case CS35L36_INT1_STATUS: + case CS35L36_INT2_STATUS: + case CS35L36_INT3_STATUS: + case CS35L36_INT4_STATUS: + case CS35L36_INT1_RAW_STATUS: + case CS35L36_INT2_RAW_STATUS: + case CS35L36_INT3_RAW_STATUS: + case CS35L36_INT4_RAW_STATUS: + case CS35L36_INT1_MASK: + case CS35L36_INT2_MASK: + case CS35L36_INT3_MASK: + case CS35L36_INT4_MASK: + case CS35L36_INT1_EDGE_LVL_CTRL: + case CS35L36_INT3_EDGE_LVL_CTRL: + case CS35L36_PAC_INT_STATUS: + case CS35L36_PAC_INT_RAW_STATUS: + case CS35L36_PAC_INT_FLUSH_CTRL: + return true; + default: + if (reg >= CS35L36_PAC_PMEM_WORD0 && + reg <= CS35L36_PAC_PMEM_WORD1023) + return true; + else + return false; + } +} + +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); + +static const char * const cs35l36_pcm_sftramp_text[] = { + "Off", ".5ms", "1ms", "2ms", "4ms", "8ms", "15ms", "30ms"}; + +static SOC_ENUM_SINGLE_DECL(pcm_sft_ramp, CS35L36_AMP_DIG_VOL_CTRL, 0, + cs35l36_pcm_sftramp_text); + +static int cs35l36_ldm_sel_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = cs35l36->pdata.ldm_mode_sel; + + return 0; +} + +static int cs35l36_ldm_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int val = (ucontrol->value.integer.value[0]) ? CS35L36_NG_AMP_EN_MASK : + 0; + + cs35l36->pdata.ldm_mode_sel = val; + + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_AMP_EN_MASK, val); + + return 0; +} + +static const struct snd_kcontrol_new cs35l36_aud_controls[] = { + SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, + 3, 0x4D0, 0x390, dig_vol_tlv), + SOC_SINGLE_TLV("Analog PCM Volume", CS35L36_AMP_GAIN_CTRL, 5, 0x13, 0, + amp_gain_tlv), + SOC_ENUM("PCM Soft Ramp", pcm_sft_ramp), + SOC_SINGLE("Amp Gain Zero-Cross", CS35L36_AMP_GAIN_CTRL, + CS35L36_AMP_ZC_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Enter Ramp", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_ENTER_SHIFT, 1, 0), + SOC_SINGLE("PDM LDM Exit Ramp", CS35L36_DAC_MSM_CFG, + CS35L36_PDM_LDM_EXIT_SHIFT, 1, 0), + SOC_SINGLE_BOOL_EXT("LDM Select", 0, cs35l36_ldm_sel_get, + cs35l36_ldm_sel_put), +}; + +static int cs35l36_main_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + u32 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 1 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + + regmap_read(cs35l36->regmap, CS35L36_INT4_RAW_STATUS, &reg); + + if (WARN_ON_ONCE(reg & CS35L36_PLL_UNLOCK_MASK)) + dev_crit(cs35l36->dev, "PLL Unlocked\n"); + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_PCM); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 0 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RX1_SEL, + CS35L36_PCM_RX_SEL_MASK, + CS35L36_PCM_RX_SEL_ZERO); + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_MASK, + 1 << CS35L36_AMP_MUTE_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL1, + CS35L36_GLOBAL_EN_MASK, + 0 << CS35L36_GLOBAL_EN_SHIFT); + + usleep_range(2000, 2100); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_boost_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_EN << + CS35L36_BST_EN_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + if (!cs35l36->pdata.extern_boost) + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL2, + CS35L36_BST_EN_MASK, + CS35L36_BST_DIS_VP << + CS35L36_BST_EN_SHIFT); + break; + default: + dev_dbg(component->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const char * const cs35l36_chan_text[] = { + "RX1", + "RX2", +}; + +static SOC_ENUM_SINGLE_DECL(chansel_enum, CS35L36_ASP_RX1_SLOT, 0, + cs35l36_chan_text); + +static const struct snd_kcontrol_new cs35l36_chan_mux = + SOC_DAPM_ENUM("Input Mux", chansel_enum); + +static const struct snd_kcontrol_new amp_enable_ctrl = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", CS35L36_AMP_OUT_MUTE, + CS35L36_AMP_MUTE_SHIFT, 1, 1); + +static const struct snd_kcontrol_new boost_ctrl = + SOC_DAPM_SINGLE_VIRT("Switch", 1); + +static const char * const asp_tx_src_text[] = { + "Zero Fill", "ASPRX1", "VMON", "IMON", "ERRVOL", "VPMON", "VBSTMON" +}; + +static const unsigned int asp_tx_src_values[] = { + 0x00, 0x08, 0x18, 0x19, 0x20, 0x28, 0x29 +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx1_src_enum, CS35L36_ASP_TX1_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx1_src = + SOC_DAPM_ENUM("ASPTX1SRC", asp_tx1_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx2_src_enum, CS35L36_ASP_TX2_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx2_src = + SOC_DAPM_ENUM("ASPTX2SRC", asp_tx2_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx3_src_enum, CS35L36_ASP_TX3_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx3_src = + SOC_DAPM_ENUM("ASPTX3SRC", asp_tx3_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx4_src_enum, CS35L36_ASP_TX4_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx4_src = + SOC_DAPM_ENUM("ASPTX4SRC", asp_tx4_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx5_src_enum, CS35L36_ASP_TX5_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx5_src = + SOC_DAPM_ENUM("ASPTX5SRC", asp_tx5_src_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL(asp_tx6_src_enum, CS35L36_ASP_TX6_SEL, 0, + CS35L36_APS_TX_SEL_MASK, asp_tx_src_text, + asp_tx_src_values); + +static const struct snd_kcontrol_new asp_tx6_src = + SOC_DAPM_ENUM("ASPTX6SRC", asp_tx6_src_enum); + +static const struct snd_soc_dapm_widget cs35l36_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Channel Mux", SND_SOC_NOPM, 0, 0, &cs35l36_chan_mux), + SND_SOC_DAPM_AIF_IN("SDIN", NULL, 0, CS35L36_ASP_RX_TX_EN, 16, 0), + + SND_SOC_DAPM_OUT_DRV_E("Main AMP", CS35L36_PWR_CTRL2, 0, 0, NULL, 0, + cs35l36_main_amp_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 1, &amp_enable_ctrl), + SND_SOC_DAPM_MIXER("CLASS H", CS35L36_PWR_CTRL3, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH_E("BOOST Enable", SND_SOC_NOPM, 0, 0, &boost_ctrl, + cs35l36_boost_event, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_AIF_OUT("ASPTX1", NULL, 0, CS35L36_ASP_RX_TX_EN, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX2", NULL, 1, CS35L36_ASP_RX_TX_EN, 1, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 2, CS35L36_ASP_RX_TX_EN, 2, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 3, CS35L36_ASP_RX_TX_EN, 3, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX5", NULL, 4, CS35L36_ASP_RX_TX_EN, 4, 0), + SND_SOC_DAPM_AIF_OUT("ASPTX6", NULL, 5, CS35L36_ASP_RX_TX_EN, 5, 0), + + SND_SOC_DAPM_MUX("ASPTX1SRC", SND_SOC_NOPM, 0, 0, &asp_tx1_src), + SND_SOC_DAPM_MUX("ASPTX2SRC", SND_SOC_NOPM, 0, 0, &asp_tx2_src), + SND_SOC_DAPM_MUX("ASPTX3SRC", SND_SOC_NOPM, 0, 0, &asp_tx3_src), + SND_SOC_DAPM_MUX("ASPTX4SRC", SND_SOC_NOPM, 0, 0, &asp_tx4_src), + SND_SOC_DAPM_MUX("ASPTX5SRC", SND_SOC_NOPM, 0, 0, &asp_tx5_src), + SND_SOC_DAPM_MUX("ASPTX6SRC", SND_SOC_NOPM, 0, 0, &asp_tx6_src), + + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L36_PWR_CTRL2, 12, 0), + SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L36_PWR_CTRL2, 13, 0), + SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L36_PWR_CTRL2, 8, 0), + SND_SOC_DAPM_ADC("VBSTMON ADC", NULL, CS35L36_PWR_CTRL2, 9, 0), + + SND_SOC_DAPM_INPUT("VP"), + SND_SOC_DAPM_INPUT("VBST"), + SND_SOC_DAPM_INPUT("VSENSE"), +}; + +static const struct snd_soc_dapm_route cs35l36_audio_map[] = { + {"VPMON ADC", NULL, "VP"}, + {"VBSTMON ADC", NULL, "VBST"}, + {"IMON ADC", NULL, "VSENSE"}, + {"VMON ADC", NULL, "VSENSE"}, + + {"ASPTX1SRC", "IMON", "IMON ADC"}, + {"ASPTX1SRC", "VMON", "VMON ADC"}, + {"ASPTX1SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX1SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX2SRC", "IMON", "IMON ADC"}, + {"ASPTX2SRC", "VMON", "VMON ADC"}, + {"ASPTX2SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX2SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX3SRC", "IMON", "IMON ADC"}, + {"ASPTX3SRC", "VMON", "VMON ADC"}, + {"ASPTX3SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX3SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX4SRC", "IMON", "IMON ADC"}, + {"ASPTX4SRC", "VMON", "VMON ADC"}, + {"ASPTX4SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX4SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX5SRC", "IMON", "IMON ADC"}, + {"ASPTX5SRC", "VMON", "VMON ADC"}, + {"ASPTX5SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX5SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX6SRC", "IMON", "IMON ADC"}, + {"ASPTX6SRC", "VMON", "VMON ADC"}, + {"ASPTX6SRC", "VBSTMON", "VBSTMON ADC"}, + {"ASPTX6SRC", "VPMON", "VPMON ADC"}, + + {"ASPTX1", NULL, "ASPTX1SRC"}, + {"ASPTX2", NULL, "ASPTX2SRC"}, + {"ASPTX3", NULL, "ASPTX3SRC"}, + {"ASPTX4", NULL, "ASPTX4SRC"}, + {"ASPTX5", NULL, "ASPTX5SRC"}, + {"ASPTX6", NULL, "ASPTX6SRC"}, + + {"AMP Capture", NULL, "ASPTX1"}, + {"AMP Capture", NULL, "ASPTX2"}, + {"AMP Capture", NULL, "ASPTX3"}, + {"AMP Capture", NULL, "ASPTX4"}, + {"AMP Capture", NULL, "ASPTX5"}, + {"AMP Capture", NULL, "ASPTX6"}, + + {"AMP Enable", "Switch", "AMP Playback"}, + {"SDIN", NULL, "AMP Enable"}, + {"Channel Mux", "RX1", "SDIN"}, + {"Channel Mux", "RX2", "SDIN"}, + {"BOOST Enable", "Switch", "Channel Mux"}, + {"CLASS H", NULL, "BOOST Enable"}, + {"Main AMP", NULL, "Channel Mux"}, + {"Main AMP", NULL, "CLASS H"}, + {"SPK", NULL, "Main AMP"}, +}; + +static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, + unsigned int fmt) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component_dai->component); + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode, clk_frc; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + slave_mode = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_MSTR_MASK, + slave_mode << CS35L36_SCLK_MSTR_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_MSTR_MASK, + slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + clk_frc = 1; + break; + case SND_SOC_DAIFMT_GATED: + clk_frc = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_FRC_MASK, clk_frc << + CS35L36_SCLK_FRC_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_FRC_MASK, clk_frc << + CS35L36_LRCLK_FRC_SHIFT); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + asp_fmt = 0; + break; + case SND_SOC_DAIFMT_I2S: + asp_fmt = 2; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + lrclk_fmt = 1; + sclk_fmt = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + lrclk_fmt = 0; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + lrclk_fmt = 1; + sclk_fmt = 1; + break; + case SND_SOC_DAIFMT_NB_NF: + lrclk_fmt = 0; + sclk_fmt = 0; + break; + default: + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, + CS35L36_LRCLK_INV_MASK, + lrclk_fmt << CS35L36_LRCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_SCLK_INV_MASK, + sclk_fmt << CS35L36_SCLK_INV_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FORMAT, + CS35L36_ASP_FMT_MASK, asp_fmt); + + return 0; +} + +struct cs35l36_global_fs_config { + int rate; + int fs_cfg; +}; + +static const struct cs35l36_global_fs_config cs35l36_fs_rates[] = { + {12000, 0x01}, + {24000, 0x02}, + {48000, 0x03}, + {96000, 0x04}, + {192000, 0x05}, + {384000, 0x06}, + {11025, 0x09}, + {22050, 0x0A}, + {44100, 0x0B}, + {88200, 0x0C}, + {176400, 0x0D}, + {8000, 0x11}, + {16000, 0x12}, + {32000, 0x13}, +}; + +static int cs35l36_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(dai->component); + unsigned int asp_width, global_fs = params_rate(params); + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_fs_rates); i++) { + if (global_fs == cs35l36_fs_rates[i].rate) + regmap_update_bits(cs35l36->regmap, + CS35L36_GLOBAL_CLK_CTRL, + CS35L36_GLOBAL_FS_MASK, + cs35l36_fs_rates[i].fs_cfg << + CS35L36_GLOBAL_FS_SHIFT); + } + + switch (params_width(params)) { + case 16: + asp_width = CS35L36_ASP_WIDTH_16; + break; + case 24: + asp_width = CS35L36_ASP_WIDTH_24; + break; + case 32: + asp_width = CS35L36_ASP_WIDTH_32; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_RX_WIDTH_MASK, + asp_width << CS35L36_ASP_RX_WIDTH_SHIFT); + } else { + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, + CS35L36_ASP_TX_WIDTH_MASK, + asp_width << CS35L36_ASP_TX_WIDTH_SHIFT); + } + + return 0; +} + +static int cs35l36_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int fs1, fs2; + + if (freq > CS35L36_FS_NOM_6MHZ) { + fs1 = CS35L36_FS1_DEFAULT_VAL; + fs2 = CS35L36_FS2_DEFAULT_VAL; + } else { + fs1 = 3 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + fs2 = 5 * ((CS35L36_FS_NOM_6MHZ * 4 + freq - 1) / freq) + 4; + } + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_TST_FS_MON0, + CS35L36_FS1_WINDOW_MASK | CS35L36_FS2_WINDOW_MASK, + fs1 | (fs2 << CS35L36_FS2_WINDOW_SHIFT)); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + return 0; +} + +static const struct cs35l36_pll_config *cs35l36_get_clk_config( + struct cs35l36_private *cs35l36, int freq) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l36_pll_sysclk); i++) { + if (cs35l36_pll_sysclk[i].freq == freq) + return &cs35l36_pll_sysclk[i]; + } + + return NULL; +} + +static const unsigned int cs35l36_src_rates[] = { + 8000, 12000, 11025, 16000, 22050, 24000, 32000, + 44100, 48000, 88200, 96000, 176400, 192000, 384000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { + .count = ARRAY_SIZE(cs35l36_src_rates), + .list = cs35l36_src_rates, +}; + +static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l36_ops = { + .startup = cs35l36_pcm_startup, + .set_fmt = cs35l36_set_dai_fmt, + .hw_params = cs35l36_pcm_hw_params, + .set_sysclk = cs35l36_dai_set_sysclk, +}; + +static struct snd_soc_dai_driver cs35l36_dai[] = { + { + .name = "cs35l36-pcm", + .id = 0, + .playback = { + .stream_name = "AMP Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_RX_FORMATS, + }, + .capture = { + .stream_name = "AMP Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = CS35L36_TX_FORMATS, + }, + .ops = &cs35l36_ops, + .symmetric_rates = 1, + }, +}; + +static int cs35l36_component_set_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, + int dir) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + const struct cs35l36_pll_config *clk_cfg; + int prev_clksrc; + bool pdm_switch; + + prev_clksrc = cs35l36->clksrc; + + switch (clk_id) { + case 0: + cs35l36->clksrc = CS35L36_PLLSRC_SCLK; + break; + case 1: + cs35l36->clksrc = CS35L36_PLLSRC_LRCLK; + break; + case 2: + cs35l36->clksrc = CS35L36_PLLSRC_PDMCLK; + break; + case 3: + cs35l36->clksrc = CS35L36_PLLSRC_SELF; + break; + case 4: + cs35l36->clksrc = CS35L36_PLLSRC_MCLK; + break; + default: + return -EINVAL; + } + + clk_cfg = cs35l36_get_clk_config(cs35l36, freq); + if (clk_cfg == NULL) { + dev_err(component->dev, "Invalid CLK Config Freq: %d\n", freq); + return -EINVAL; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 1 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_REFCLK_FREQ_MASK, + clk_cfg->clk_cfg << CS35L36_REFCLK_FREQ_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 0 << CS35L36_PLL_REFCLK_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_CLK_SEL_MASK, + cs35l36->clksrc); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_OPENLOOP_MASK, + 0 << CS35L36_PLL_OPENLOOP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_CLK_CTRL, + CS35L36_PLL_REFCLK_EN_MASK, + 1 << CS35L36_PLL_REFCLK_EN_SHIFT); + + if (cs35l36->rev_id == CS35L36_REV_A0) { + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_write(cs35l36->regmap, CS35L36_DCO_CTRL, 0x00036DA8); + regmap_write(cs35l36->regmap, CS35L36_MISC_CTRL, 0x0100EE0E); + + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_IGAIN_MASK, + CS35L36_PLL_IGAIN << + CS35L36_PLL_IGAIN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_PLL_LOOP_PARAMS, + CS35L36_PLL_FFL_IGAIN_MASK, + clk_cfg->fll_igain); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->clksrc == CS35L36_PLLSRC_PDMCLK) { + pdm_switch = cs35l36->pdata.ldm_mode_sel && + (prev_clksrc != CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 1 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } else { + pdm_switch = cs35l36->pdata.ldm_mode_sel && + (prev_clksrc == CS35L36_PLLSRC_PDMCLK); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 0 << CS35L36_NG_DELAY_SHIFT); + + regmap_update_bits(cs35l36->regmap, CS35L36_DAC_MSM_CFG, + CS35L36_PDM_MODE_MASK, + 0 << CS35L36_PDM_MODE_SHIFT); + + if (pdm_switch) + regmap_update_bits(cs35l36->regmap, CS35L36_NG_CFG, + CS35L36_NG_DELAY_MASK, + 3 << CS35L36_NG_DELAY_SHIFT); + } + + return 0; +} + +static int cs35l36_boost_inductor(struct cs35l36_private *cs35l36, int inductor) +{ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K1_MASK, 0x3C); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_COEFF, + CS35L36_BSTCVRT_K2_MASK, + 0x3C << CS35L36_BSTCVRT_K2_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SW_FREQ, + CS35L36_BSTCVRT_CCMFREQ_MASK, 0x00); + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x75 << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x00); + break; + case 1200: /* 1.2 uH */ + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_SLOPE_MASK, + 0x6B << CS35L36_BSTCVRT_SLOPE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_SLOPE_LBST, + CS35L36_BSTCVRT_LBSTVAL_MASK, 0x01); + break; + default: + dev_err(cs35l36->dev, "%s Invalid Inductor Value %d uH\n", + __func__, inductor); + return -EINVAL; + } + + return 0; +} + +static int cs35l36_component_probe(struct snd_soc_component *component) +{ + struct cs35l36_private *cs35l36 = + snd_soc_component_get_drvdata(component); + int ret = 0; + + if ((cs35l36->rev_id == CS35L36_REV_A0) && cs35l36->pdata.dcm_mode) { + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_DCM_CTRL, + CS35L36_DCM_AUTO_MASK, + CS35L36_DCM_AUTO_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_MASK, + 0 << CS35L36_BST_MAN_IPKCOMP_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BST_TST_MANUAL, + CS35L36_BST_MAN_IPKCOMP_EN_MASK, + CS35L36_BST_MAN_IPKCOMP_EN_MASK); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + if (cs35l36->pdata.amp_pcm_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_AMP_DIG_VOL_CTRL, + CS35L36_AMP_PCM_INV_MASK, + CS35L36_AMP_PCM_INV_MASK); + + if (cs35l36->pdata.multi_amp_mode) + regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, + CS35L36_ASP_TX_HIZ_MASK, + CS35L36_ASP_TX_HIZ_MASK); + + if (cs35l36->pdata.imon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_IMON_POL_MASK, 0); + + if (cs35l36->pdata.vmon_pol_inv) + regmap_update_bits(cs35l36->regmap, CS35L36_VI_SPKMON_FILT, + CS35L36_VMON_POL_MASK, 0); + + if (cs35l36->pdata.bst_vctl) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + cs35l36->pdata.bst_vctl); + + if (cs35l36->pdata.bst_vctl_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L35_BSTCVRT_CTL_SEL_MASK, + cs35l36->pdata.bst_vctl_sel); + + if (cs35l36->pdata.bst_ipk) + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_PEAK_CUR, + CS35L36_BST_IPK_MASK, + cs35l36->pdata.bst_ipk); + + if (cs35l36->pdata.boost_ind) { + ret = cs35l36_boost_inductor(cs35l36, cs35l36->pdata.boost_ind); + if (ret < 0) { + dev_err(cs35l36->dev, + "Boost inductor config failed(%d)\n", ret); + return ret; + } + } + + if (cs35l36->pdata.temp_warn_thld) + regmap_update_bits(cs35l36->regmap, CS35L36_DTEMP_WARN_THLD, + CS35L36_TEMP_THLD_MASK, + cs35l36->pdata.temp_warn_thld); + + if (cs35l36->pdata.irq_drv_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_DRV_SEL_MASK, + cs35l36->pdata.irq_drv_sel << + CS35L36_INT_DRV_SEL_SHIFT); + + if (cs35l36->pdata.irq_gpio_sel) + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_GPIO_SEL_MASK, + cs35l36->pdata.irq_gpio_sel << + CS35L36_INT_GPIO_SEL_SHIFT); + + /* + * Rev B0 has 2 versions + * L36 is 10V + * L37 is 12V + * If L36 we need to clamp some values for safety + * after probe has setup dt values. We want to make + * sure we dont miss any values set in probe + */ + if (cs35l36->chip_version == CS35L36_10V_L36) { + regmap_update_bits(cs35l36->regmap, + CS35L36_BSTCVRT_OVERVOLT_CTRL, + CS35L36_BST_OVP_THLD_MASK, + CS35L36_BST_OVP_THLD_11V); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + regmap_update_bits(cs35l36->regmap, CS35L36_BST_ANA2_TEST, + CS35L36_BST_OVP_TRIM_MASK, + CS35L36_BST_OVP_TRIM_11V << + CS35L36_BST_OVP_TRIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL2, + CS35L36_BST_CTRL_LIM_MASK, + 1 << CS35L36_BST_CTRL_LIM_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_BSTCVRT_VCTRL1, + CS35L35_BSTCVRT_CTL_MASK, + CS35L36_BST_CTRL_10V_CLAMP); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + } + + /* + * RevA and B require the disabling of + * SYNC_GLOBAL_OVR when GLOBAL_EN = 0. + * Just turn it off from default + */ + regmap_update_bits(cs35l36->regmap, CS35L36_CTRL_OVRRIDE, + CS35L36_SYNC_GLOBAL_OVR_MASK, + 0 << CS35L36_SYNC_GLOBAL_OVR_SHIFT); + + return 0; +} + +static const struct snd_soc_component_driver soc_component_dev_cs35l36 = { + .probe = &cs35l36_component_probe, + .set_sysclk = cs35l36_component_set_sysclk, + .dapm_widgets = cs35l36_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l36_dapm_widgets), + .dapm_routes = cs35l36_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l36_audio_map), + .controls = cs35l36_aud_controls, + .num_controls = ARRAY_SIZE(cs35l36_aud_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct regmap_config cs35l36_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = CS35L36_PAC_PMEM_WORD1023, + .reg_defaults = cs35l36_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l36_reg), + .precious_reg = cs35l36_precious_reg, + .volatile_reg = cs35l36_volatile_reg, + .readable_reg = cs35l36_readable_reg, + .cache_type = REGCACHE_RBTREE, +}; + +static irqreturn_t cs35l36_irq(int irq, void *data) +{ + struct cs35l36_private *cs35l36 = data; + unsigned int status[4]; + unsigned int masks[4]; + int ret = IRQ_NONE; + + /* ack the irq by reading all status registers */ + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_STATUS, status, + ARRAY_SIZE(status)); + + regmap_bulk_read(cs35l36->regmap, CS35L36_INT1_MASK, masks, + ARRAY_SIZE(masks)); + + /* Check to see if unmasked bits are active */ + if (!(status[0] & ~masks[0]) && !(status[1] & ~masks[1]) && + !(status[2] & ~masks[2]) && !(status[3] & ~masks[3])) { + return IRQ_NONE; + } + + /* + * The following interrupts require a + * protection release cycle to get the + * speaker out of Safe-Mode. + */ + if (status[2] & CS35L36_AMP_SHORT_ERR) { + dev_crit(cs35l36->dev, "Amp short error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, + CS35L36_AMP_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_AMP_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT3_STATUS, + CS35L36_AMP_SHORT_ERR, + CS35L36_AMP_SHORT_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_WARN) { + dev_crit(cs35l36->dev, "Over temperature warning\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, + CS35L36_TEMP_WARN_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_WARN_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_WARN, CS35L36_TEMP_WARN); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_TEMP_ERR) { + dev_crit(cs35l36->dev, "Over temperature error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_TEMP_ERR, CS35L36_TEMP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_OVP_ERR) { + dev_crit(cs35l36->dev, "VBST Over Voltage error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, CS35L36_TEMP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_TEMP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_OVP_ERR, CS35L36_BST_OVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_DCM_UVP_ERR) { + dev_crit(cs35l36->dev, "DCM VBST Under Voltage Error\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, + CS35L36_BST_UVP_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_UVP_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_DCM_UVP_ERR, + CS35L36_BST_DCM_UVP_ERR); + ret = IRQ_HANDLED; + } + + if (status[0] & CS35L36_BST_SHORT_ERR) { + dev_crit(cs35l36->dev, "LBST SHORT error!\n"); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, + CS35L36_BST_SHORT_ERR_RLS); + regmap_update_bits(cs35l36->regmap, CS35L36_PROTECT_REL_ERR, + CS35L36_BST_SHORT_ERR_RLS, 0); + regmap_update_bits(cs35l36->regmap, CS35L36_INT1_STATUS, + CS35L36_BST_SHORT_ERR, + CS35L36_BST_SHORT_ERR); + ret = IRQ_HANDLED; + } + + return ret; +} + +static int cs35l36_handle_of_data(struct i2c_client *i2c_client, + struct cs35l36_platform_data *pdata) +{ + struct device_node *np = i2c_client->dev.of_node; + struct cs35l36_vpbr_cfg *vpbr_config = &pdata->vpbr_config; + struct device_node *vpbr_node; + unsigned int val; + int ret; + + if (!np) + return 0; + + ret = of_property_read_u32(np, "cirrus,boost-ctl-millivolt", &val); + if (!ret) { + if (val < 2550 || val > 12000) { + dev_err(&i2c_client->dev, + "Invalid Boost Voltage %d mV\n", val); + return -EINVAL; + } + pdata->bst_vctl = (((val - 2550) / 100) + 1) << 1; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-ctl-millivolt'"); + return -EINVAL; + } + + ret = of_property_read_u32(np, "cirrus,boost-ctl-select", &val); + if (!ret) + pdata->bst_vctl_sel = val | CS35L36_VALID_PDATA; + + ret = of_property_read_u32(np, "cirrus,boost-peak-milliamp", &val); + if (!ret) { + if (val < 1600 || val > 4500) { + dev_err(&i2c_client->dev, + "Invalid Boost Peak Current %u mA\n", val); + return -EINVAL; + } + + pdata->bst_ipk = (val - 1600) / 50; + } else { + dev_err(&i2c_client->dev, + "Unable to find required parameter 'cirrus,boost-peak-milliamp'"); + return -EINVAL; + } + + pdata->multi_amp_mode = of_property_read_bool(np, + "cirrus,multi-amp-mode"); + + pdata->dcm_mode = of_property_read_bool(np, + "cirrus,dcm-mode-enable"); + + pdata->amp_pcm_inv = of_property_read_bool(np, + "cirrus,amp-pcm-inv"); + + pdata->imon_pol_inv = of_property_read_bool(np, + "cirrus,imon-pol-inv"); + + pdata->vmon_pol_inv = of_property_read_bool(np, + "cirrus,vmon-pol-inv"); + + if (of_property_read_u32(np, "cirrus,temp-warn-threshold", &val) >= 0) + pdata->temp_warn_thld = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val) >= 0) { + pdata->boost_ind = val; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + + if (of_property_read_u32(np, "cirrus,irq-drive-select", &val) >= 0) + pdata->irq_drv_sel = val | CS35L36_VALID_PDATA; + + if (of_property_read_u32(np, "cirrus,irq-gpio-select", &val) >= 0) + pdata->irq_gpio_sel = val | CS35L36_VALID_PDATA; + + /* VPBR Config */ + vpbr_node = of_get_child_by_name(np, "cirrus,vpbr-config"); + vpbr_config->is_present = vpbr_node ? true : false; + if (vpbr_config->is_present) { + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-en", + &val) >= 0) + vpbr_config->vpbr_en = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-thld", + &val) >= 0) + vpbr_config->vpbr_thld = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-rate", + &val) >= 0) + vpbr_config->vpbr_atk_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-atk-vol", + &val) >= 0) + vpbr_config->vpbr_atk_vol = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-max-attn", + &val) >= 0) + vpbr_config->vpbr_max_attn = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-wait", + &val) >= 0) + vpbr_config->vpbr_wait = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-rel-rate", + &val) >= 0) + vpbr_config->vpbr_rel_rate = val; + if (of_property_read_u32(vpbr_node, "cirrus,vpbr-mute-en", + &val) >= 0) + vpbr_config->vpbr_mute_en = val; + } + of_node_put(vpbr_node); + + return 0; +} + +static int cs35l36_pac(struct cs35l36_private *cs35l36) +{ + int ret, count; + unsigned int val; + + if (cs35l36->rev_id != CS35L36_REV_B0) + return 0; + + /* + * Magic code for internal PAC + */ + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_UNLOCK2); + + usleep_range(9500, 10500); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_RESET); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS); + regmap_write(cs35l36->regmap, CS35L36_PAC_PMEM_WORD0, + CS35L36_B0_PAC_PATCH); + + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL3, + CS35L36_PAC_MEM_ACCESS_CLR); + regmap_write(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK); + + usleep_range(9500, 10500); + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", ret); + return ret; + } + + count = 0; + while (!(val & CS35L36_MCU_CONFIG_CLR)) { + usleep_range(100, 200); + count++; + + ret = regmap_read(cs35l36->regmap, CS35L36_INT4_STATUS, + &val); + if (ret < 0) { + dev_err(cs35l36->dev, "Failed to read int4_status %d\n", + ret); + return ret; + } + + if (count >= 100) + return -EINVAL; + } + + regmap_write(cs35l36->regmap, CS35L36_INT4_STATUS, + CS35L36_MCU_CONFIG_CLR); + regmap_update_bits(cs35l36->regmap, CS35L36_PAC_CTL1, + CS35L36_PAC_ENABLE_MASK, 0); + + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK1); + regmap_write(cs35l36->regmap, CS35L36_TESTKEY_CTRL, + CS35L36_TEST_LOCK2); + + return 0; +} + +static void cs35l36_apply_vpbr_config(struct cs35l36_private *cs35l36) +{ + struct cs35l36_platform_data *pdata = &cs35l36->pdata; + struct cs35l36_vpbr_cfg *vpbr_config = &pdata->vpbr_config; + + regmap_update_bits(cs35l36->regmap, CS35L36_PWR_CTRL3, + CS35L36_VPBR_EN_MASK, + vpbr_config->vpbr_en << + CS35L36_VPBR_EN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_THLD_MASK, + vpbr_config->vpbr_thld << + CS35L36_VPBR_THLD_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MAX_ATTN_MASK, + vpbr_config->vpbr_max_attn << + CS35L36_VPBR_MAX_ATTN_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_VOL_MASK, + vpbr_config->vpbr_atk_vol << + CS35L36_VPBR_ATK_VOL_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_ATK_RATE_MASK, + vpbr_config->vpbr_atk_rate << + CS35L36_VPBR_ATK_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_WAIT_MASK, + vpbr_config->vpbr_wait << + CS35L36_VPBR_WAIT_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_REL_RATE_MASK, + vpbr_config->vpbr_rel_rate << + CS35L36_VPBR_REL_RATE_SHIFT); + regmap_update_bits(cs35l36->regmap, CS35L36_VPBR_CFG, + CS35L36_VPBR_MUTE_EN_MASK, + vpbr_config->vpbr_mute_en << + CS35L36_VPBR_MUTE_EN_SHIFT); +} + +static const struct reg_sequence cs35l36_reva0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + /* Errata Writes */ + { CS35L36_OTP_CTRL1, 0x00002060 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { CS35L36_OTP_CTRL1, 0x00002460 }, + { CS35L36_OTP_CTRL2, 0x00000001 }, + { 0x00002088, 0x012A1838 }, + { 0x00003014, 0x0100EE0E }, + { 0x00003008, 0x0008184A }, + { 0x00007418, 0x509001C8 }, + { 0x00007064, 0x0929A800 }, + { 0x00002D10, 0x0002C01C }, + { 0x0000410C, 0x00000A11 }, + { 0x00006E08, 0x8B19140C }, + { 0x00006454, 0x0300000A }, + { CS35L36_AMP_NG_CTRL, 0x000020EF }, + { 0x00007E34, 0x0000000E }, + { 0x0000410C, 0x00000A11 }, + { 0x00007410, 0x20514B00 }, + /* PAC Config */ + { CS35L36_CTRL_OVRRIDE, 0x00000000 }, + { CS35L36_PAC_INT0_CTRL, 0x00860001 }, + { CS35L36_PAC_INT1_CTRL, 0x00860001 }, + { CS35L36_PAC_INT2_CTRL, 0x00860001 }, + { CS35L36_PAC_INT3_CTRL, 0x00860001 }, + { CS35L36_PAC_INT4_CTRL, 0x00860001 }, + { CS35L36_PAC_INT5_CTRL, 0x00860001 }, + { CS35L36_PAC_INT6_CTRL, 0x00860001 }, + { CS35L36_PAC_INT7_CTRL, 0x00860001 }, + { CS35L36_PAC_INT_FLUSH_CTRL, 0x000000FF }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static const struct reg_sequence cs35l36_revb0_errata_patch[] = { + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_UNLOCK2 }, + { 0x00007064, 0x0929A800 }, + { 0x00007850, 0x00002FA9 }, + { 0x00007854, 0x0003F1D5 }, + { 0x00007858, 0x0003F5E3 }, + { 0x0000785C, 0x00001137 }, + { 0x00007860, 0x0001A7A5 }, + { 0x00007864, 0x0002F16A }, + { 0x00007868, 0x00003E21 }, + { 0x00007848, 0x00000001 }, + { 0x00003854, 0x05180240 }, + { 0x00007418, 0x509001C8 }, + { 0x0000394C, 0x028764BD }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK1 }, + { CS35L36_TESTKEY_CTRL, CS35L36_TEST_LOCK2 }, +}; + +static int cs35l36_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l36_private *cs35l36; + struct device *dev = &i2c_client->dev; + struct cs35l36_platform_data *pdata = dev_get_platdata(dev); + struct irq_data *irq_d; + int ret, irq_pol, chip_irq_pol, i; + u32 reg_id, reg_revid, l37_id_reg; + + cs35l36 = devm_kzalloc(dev, sizeof(struct cs35l36_private), GFP_KERNEL); + if (!cs35l36) + return -ENOMEM; + + cs35l36->dev = dev; + + i2c_set_clientdata(i2c_client, cs35l36); + cs35l36->regmap = devm_regmap_init_i2c(i2c_client, &cs35l36_regmap); + if (IS_ERR(cs35l36->regmap)) { + ret = PTR_ERR(cs35l36->regmap); + dev_err(dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); + for (i = 0; i < ARRAY_SIZE(cs35l36_supplies); i++) + cs35l36->supplies[i].supply = cs35l36_supplies[i]; + + ret = devm_regulator_bulk_get(dev, cs35l36->num_supplies, + cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to request core supplies: %d\n", ret); + return ret; + } + + if (pdata) { + cs35l36->pdata = *pdata; + } else { + pdata = devm_kzalloc(dev, sizeof(struct cs35l36_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + if (i2c_client->dev.of_node) { + ret = cs35l36_handle_of_data(i2c_client, pdata); + if (ret != 0) + return ret; + + } + + cs35l36->pdata = *pdata; + } + + ret = regulator_bulk_enable(cs35l36->num_supplies, cs35l36->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + /* returning NULL can be an option if in stereo mode */ + cs35l36->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs35l36->reset_gpio)) { + ret = PTR_ERR(cs35l36->reset_gpio); + cs35l36->reset_gpio = NULL; + if (ret == -EBUSY) { + dev_info(dev, "Reset line busy, assuming shared reset\n"); + } else { + dev_err(dev, "Failed to get reset GPIO: %d\n", ret); + goto err_disable_regs; + } + } + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 1); + + usleep_range(2000, 2100); + + /* initialize amplifier */ + ret = regmap_read(cs35l36->regmap, CS35L36_SW_RESET, &reg_id); + if (ret < 0) { + dev_err(dev, "Get Device ID failed %d\n", ret); + goto err; + } + + if (reg_id != CS35L36_CHIP_ID) { + dev_err(dev, "Device ID (%X). Expected ID %X\n", reg_id, + CS35L36_CHIP_ID); + ret = -ENODEV; + goto err; + } + + ret = regmap_read(cs35l36->regmap, CS35L36_REV_ID, &reg_revid); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed %d\n", ret); + goto err; + } + + cs35l36->rev_id = reg_revid >> 8; + + ret = regmap_read(cs35l36->regmap, CS35L36_OTP_MEM30, &l37_id_reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to read otp_id Register %d\n", + ret); + return ret; + } + + if ((l37_id_reg & CS35L36_OTP_REV_MASK) == CS35L36_OTP_REV_L37) + cs35l36->chip_version = CS35L36_12V_L37; + else + cs35l36->chip_version = CS35L36_10V_L36; + + switch (cs35l36->rev_id) { + case CS35L36_REV_A0: + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_reva0_errata_patch, + ARRAY_SIZE(cs35l36_reva0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply A0 errata patch %d\n", + ret); + goto err; + } + break; + case CS35L36_REV_B0: + ret = cs35l36_pac(cs35l36); + if (ret < 0) { + dev_err(dev, "Failed to Trim OTP %d\n", ret); + goto err; + } + + ret = regmap_register_patch(cs35l36->regmap, + cs35l36_revb0_errata_patch, + ARRAY_SIZE(cs35l36_revb0_errata_patch)); + if (ret < 0) { + dev_err(dev, "Failed to apply B0 errata patch %d\n", + ret); + goto err; + } + break; + } + + if (pdata->vpbr_config.is_present) + cs35l36_apply_vpbr_config(cs35l36); + + irq_d = irq_get_irq_data(i2c_client->irq); + if (IS_ERR(irq_d)) { + dev_err(&i2c_client->dev, "Invalid IRQ: %d\n", i2c_client->irq); + ret = PTR_ERR(irq_d); + goto err; + } + + irq_pol = irqd_get_trigger_type(irq_d); + + switch (irq_pol) { + case IRQF_TRIGGER_FALLING: + case IRQF_TRIGGER_LOW: + chip_irq_pol = 0; + break; + case IRQF_TRIGGER_RISING: + case IRQF_TRIGGER_HIGH: + chip_irq_pol = 1; + break; + default: + dev_err(cs35l36->dev, "Invalid IRQ polarity: %d\n", irq_pol); + ret = -EINVAL; + goto err; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_POL_SEL_MASK, + chip_irq_pol << CS35L36_INT_POL_SEL_SHIFT); + + ret = devm_request_threaded_irq(dev, i2c_client->irq, NULL, cs35l36_irq, + IRQF_ONESHOT | irq_pol, "cs35l36", + cs35l36); + if (ret != 0) { + dev_err(dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + + regmap_update_bits(cs35l36->regmap, CS35L36_PAD_INTERFACE, + CS35L36_INT_OUTPUT_EN_MASK, 1); + + /* Set interrupt masks for critical errors */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_DEFAULT); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_DEFAULT); + + dev_info(&i2c_client->dev, "Cirrus Logic CS35L%d, Revision: %02X\n", + cs35l36->chip_version, reg_revid >> 8); + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_cs35l36, + cs35l36_dai, + ARRAY_SIZE(cs35l36_dai)); + if (ret < 0) { + dev_err(dev, "%s: Register component failed %d\n", __func__, + ret); + goto err; + } + + return 0; + +err: + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + +err_disable_regs: + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + return ret; +} + +static int cs35l36_i2c_remove(struct i2c_client *client) +{ + struct cs35l36_private *cs35l36 = i2c_get_clientdata(client); + + /* Reset interrupt masks for device removal */ + regmap_write(cs35l36->regmap, CS35L36_INT1_MASK, + CS35L36_INT1_MASK_RESET); + regmap_write(cs35l36->regmap, CS35L36_INT3_MASK, + CS35L36_INT3_MASK_RESET); + + if (cs35l36->reset_gpio) + gpiod_set_value_cansleep(cs35l36->reset_gpio, 0); + + regulator_bulk_disable(cs35l36->num_supplies, cs35l36->supplies); + + return 0; +} +static const struct of_device_id cs35l36_of_match[] = { + {.compatible = "cirrus,cs35l36"}, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l36_of_match); + +static const struct i2c_device_id cs35l36_id[] = { + {"cs35l36", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l36_id); + +static struct i2c_driver cs35l36_i2c_driver = { + .driver = { + .name = "cs35l36", + .of_match_table = cs35l36_of_match, + }, + .id_table = cs35l36_id, + .probe = cs35l36_i2c_probe, + .remove = cs35l36_i2c_remove, +}; +module_i2c_driver(cs35l36_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L36 driver"); +MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l36.h b/sound/soc/codecs/cs35l36.h new file mode 100644 index 0000000..f6e38c6 --- /dev/null +++ b/sound/soc/codecs/cs35l36.h @@ -0,0 +1,446 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * cs35l36.h -- CS35L36 ALSA SoC audio driver + * + * Copyright 2018 Cirrus Logic, Inc. + * + * Author: James Schulman <james.schulman(a)cirrus.com> + * + */ + +#ifndef __CS35L36_H__ +#define __CS35L36_H__ + +#include <linux/regmap.h> + +#define CS35L36_FIRSTREG 0x00000000 +#define CS35L36_LASTREG 0x00E037FC +#define CS35L36_SW_RESET 0x00000000 +#define CS35L36_SW_REV 0x00000004 +#define CS35L36_HW_REV 0x00000008 +#define CS35L36_TESTKEY_CTRL 0x00000020 +#define CS35L36_USERKEY_CTL 0x00000024 +#define CS35L36_OTP_MEM30 0x00000478 +#define CS35L36_OTP_CTRL1 0x00000500 +#define CS35L36_OTP_CTRL2 0x00000504 +#define CS35L36_OTP_CTRL3 0x00000508 +#define CS35L36_OTP_CTRL4 0x0000050C +#define CS35L36_OTP_CTRL5 0x00000510 +#define CS35L36_PAC_CTL1 0x00000C00 +#define CS35L36_PAC_CTL2 0x00000C04 +#define CS35L36_PAC_CTL3 0x00000C08 +#define CS35L36_DEVICE_ID 0x00002004 +#define CS35L36_FAB_ID 0x00002008 +#define CS35L36_REV_ID 0x0000200C +#define CS35L36_PWR_CTRL1 0x00002014 +#define CS35L36_PWR_CTRL2 0x00002018 +#define CS35L36_PWR_CTRL3 0x0000201C +#define CS35L36_CTRL_OVRRIDE 0x00002020 +#define CS35L36_AMP_OUT_MUTE 0x00002024 +#define CS35L36_OTP_TRIM_STATUS 0x00002028 +#define CS35L36_DISCH_FILT 0x0000202C +#define CS35L36_OSC_TRIM 0x00002030 +#define CS35L36_PROTECT_REL_ERR 0x00002034 +#define CS35L36_PAD_INTERFACE 0x00002400 +#define CS35L36_PLL_CLK_CTRL 0x00002C04 +#define CS35L36_GLOBAL_CLK_CTRL 0x00002C0C +#define CS35L36_ADC_CLK_CTRL 0x00002C10 +#define CS35L36_SWIRE_CLK_CTRL 0x00002C14 +#define CS35L36_SP_SCLK_CLK_CTRL 0x00002D00 +#define CS35L36_TST_FS_MON0 0x00002D10 +#define CS35L36_PLL_LOOP_PARAMS 0x00003008 +#define CS35L36_DCO_CTRL 0x00003010 +#define CS35L36_MISC_CTRL 0x00003014 +#define CS35L36_MDSYNC_EN 0x00003404 +#define CS35L36_MDSYNC_TX_ID 0x00003408 +#define CS35L36_MDSYNC_PWR_CTRL 0x0000340C +#define CS35L36_MDSYNC_DATA_TX 0x00003410 +#define CS35L36_MDSYNC_TX_STATUS 0x0000341C +#define CS35L36_MDSYNC_RX_STATUS 0x00003420 +#define CS35L36_MDSYNC_ERR_STATUS 0x00003424 +#define CS35L36_BSTCVRT_VCTRL1 0x00003800 +#define CS35L36_BSTCVRT_VCTRL2 0x00003804 +#define CS35L36_BSTCVRT_PEAK_CUR 0x00003808 +#define CS35L36_BSTCVRT_SFT_RAMP 0x0000380C +#define CS35L36_BSTCVRT_COEFF 0x00003810 +#define CS35L36_BSTCVRT_SLOPE_LBST 0x00003814 +#define CS35L36_BSTCVRT_SW_FREQ 0x00003818 +#define CS35L36_BSTCVRT_DCM_CTRL 0x0000381C +#define CS35L36_BSTCVRT_DCM_MODE_FORCE 0x00003820 +#define CS35L36_BSTCVRT_OVERVOLT_CTRL 0x00003830 +#define CS35L36_BST_TST_MANUAL 0x0000393C +#define CS35L36_BST_ANA2_TEST 0x0000394C +#define CS35L36_VPI_LIMIT_MODE 0x00003C04 +#define CS35L36_VPI_LIMIT_MINMAX 0x00003C08 +#define CS35L36_VPI_VP_THLD 0x00003C0C +#define CS35L36_VPI_TRACK_CTRL 0x00003C10 +#define CS35L36_VPI_TRIG_MODE_CTRL 0x00003C14 +#define CS35L36_VPI_TRIG_STEPS 0x00003C18 +#define CS35L36_VI_SPKMON_FILT 0x00004004 +#define CS35L36_VI_SPKMON_GAIN 0x00004008 +#define CS35L36_VI_SPKMON_IP_SEL 0x00004100 +#define CS35L36_DTEMP_WARN_THLD 0x00004220 +#define CS35L36_DTEMP_STATUS 0x00004300 +#define CS35L36_VPVBST_FS_SEL 0x00004400 +#define CS35L36_VPVBST_VP_CTRL 0x00004440 +#define CS35L36_VPVBST_VBST_CTRL 0x00004444 +#define CS35L36_ASP_TX_PIN_CTRL 0x00004800 +#define CS35L36_ASP_RATE_CTRL 0x00004804 +#define CS35L36_ASP_FORMAT 0x00004808 +#define CS35L36_ASP_FRAME_CTRL 0x00004818 +#define CS35L36_ASP_TX1_TX2_SLOT 0x0000481C +#define CS35L36_ASP_TX3_TX4_SLOT 0x00004820 +#define CS35L36_ASP_TX5_TX6_SLOT 0x00004824 +#define CS35L36_ASP_TX7_TX8_SLOT 0x00004828 +#define CS35L36_ASP_RX1_SLOT 0x0000482C +#define CS35L36_ASP_RX_TX_EN 0x0000483C +#define CS35L36_ASP_RX1_SEL 0x00004C00 +#define CS35L36_ASP_TX1_SEL 0x00004C20 +#define CS35L36_ASP_TX2_SEL 0x00004C24 +#define CS35L36_ASP_TX3_SEL 0x00004C28 +#define CS35L36_ASP_TX4_SEL 0x00004C2C +#define CS35L36_ASP_TX5_SEL 0x00004C30 +#define CS35L36_ASP_TX6_SEL 0x00004C34 +#define CS35L36_SWIRE_P1_TX1_SEL 0x00004C40 +#define CS35L36_SWIRE_P1_TX2_SEL 0x00004C44 +#define CS35L36_SWIRE_P2_TX1_SEL 0x00004C60 +#define CS35L36_SWIRE_P2_TX2_SEL 0x00004C64 +#define CS35L36_SWIRE_P2_TX3_SEL 0x00004C68 +#define CS35L36_SWIRE_DP1_FIFO_CFG 0x00005000 +#define CS35L36_SWIRE_DP2_FIFO_CFG 0x00005004 +#define CS35L36_SWIRE_DP3_FIFO_CFG 0x00005008 +#define CS35L36_SWIRE_PCM_RX_DATA 0x0000500C +#define CS35L36_SWIRE_FS_SEL 0x00005010 +#define CS35L36_SPARE_CP_BITS 0x00005C00 +#define CS35L36_AMP_DIG_VOL_CTRL 0x00006000 +#define CS35L36_VPBR_CFG 0x00006404 +#define CS35L36_VBBR_CFG 0x00006408 +#define CS35L36_VPBR_STATUS 0x0000640C +#define CS35L36_VBBR_STATUS 0x00006410 +#define CS35L36_OVERTEMP_CFG 0x00006414 +#define CS35L36_AMP_ERR_VOL 0x00006418 +#define CS35L36_CLASSH_CFG 0x00006800 +#define CS35L36_CLASSH_FET_DRV_CFG 0x00006804 +#define CS35L36_NG_CFG 0x00006808 +#define CS35L36_AMP_GAIN_CTRL 0x00006C04 +#define CS35L36_PWM_MOD_IO_CTRL 0x0000706C +#define CS35L36_PWM_MOD_STATUS 0x00007070 +#define CS35L36_DAC_MSM_CFG 0x00007400 +#define CS35L36_AMP_SLOPE_CTRL 0x00007410 +#define CS35L36_AMP_PDM_VOLUME 0x00007E04 +#define CS35L36_AMP_PDM_RATE_CTRL 0x00007E08 +#define CS35L36_PDM_CH_SEL 0x00007E10 +#define CS35L36_AMP_NG_CTRL 0x00007E14 +#define CS35L36_PDM_HIGHFILT_CTRL 0x00007E3C +#define CS35L36_INT1_STATUS 0x00D00000 +#define CS35L36_INT2_STATUS 0x00D00004 +#define CS35L36_INT3_STATUS 0x00D00008 +#define CS35L36_INT4_STATUS 0x00D0000C +#define CS35L36_INT1_RAW_STATUS 0x00D00020 +#define CS35L36_INT2_RAW_STATUS 0x00D00024 +#define CS35L36_INT3_RAW_STATUS 0x00D00028 +#define CS35L36_INT4_RAW_STATUS 0x00D0002C +#define CS35L36_INT1_MASK 0x00D00040 +#define CS35L36_INT2_MASK 0x00D00044 +#define CS35L36_INT3_MASK 0x00D00048 +#define CS35L36_INT4_MASK 0x00D0004C +#define CS35L36_INT1_EDGE_LVL_CTRL 0x00D00060 +#define CS35L36_INT3_EDGE_LVL_CTRL 0x00D00068 +#define CS35L36_PAC_INT_STATUS 0x00D00200 +#define CS35L36_PAC_INT_RAW_STATUS 0x00D00210 +#define CS35L36_PAC_INT_FLUSH_CTRL 0x00D00218 +#define CS35L36_PAC_INT0_CTRL 0x00D00220 +#define CS35L36_PAC_INT1_CTRL 0x00D00224 +#define CS35L36_PAC_INT2_CTRL 0x00D00228 +#define CS35L36_PAC_INT3_CTRL 0x00D0022C +#define CS35L36_PAC_INT4_CTRL 0x00D00230 +#define CS35L36_PAC_INT5_CTRL 0x00D00234 +#define CS35L36_PAC_INT6_CTRL 0x00D00238 +#define CS35L36_PAC_INT7_CTRL 0x00D0023C +#define CS35L36_PAC_PMEM_WORD0 0x00E02800 +#define CS35L36_PAC_PMEM_WORD1 0x00E02804 +#define CS35L36_PAC_PMEM_WORD1023 0x00E037FC + +#define CS35L36_INTPAC_REG_COUNT 25 +#define CS35L36_CHIP_ID 0x00035A36 + +#define CS35L36_INT_OUTPUT_EN_MASK 0x01 +#define CS35L36_INT_GPIO_SEL_MASK 0x02 +#define CS35L36_INT_GPIO_SEL_SHIFT 1 +#define CS35L36_INT_POL_SEL_MASK 0x04 +#define CS35L36_INT_POL_SEL_SHIFT 2 +#define CS35L36_INT_DRV_SEL_MASK 0x20 +#define CS35L36_INT_DRV_SEL_SHIFT 5 +#define CS35L36_IRQ_SRC_MASK 0x08 +#define CS35L36_IRQ_SRC_SHIFT 3 + +#define CS35L36_SCLK_MSTR_MASK 0x40 +#define CS35L36_SCLK_MSTR_SHIFT 6 +#define CS35L36_LRCLK_MSTR_MASK 0x01 +#define CS35L36_LRCLK_MSTR_SHIFT 0 +#define CS35L36_SCLK_INV_MASK 0x100 +#define CS35L36_SCLK_INV_SHIFT 8 +#define CS35L36_LRCLK_INV_MASK 0x04 +#define CS35L36_LRCLK_INV_SHIFT 2 +#define CS35L36_SCLK_FRC_MASK 0x80 +#define CS35L36_SCLK_FRC_SHIFT 7 +#define CS35L36_LRCLK_FRC_MASK 0x02 +#define CS35L36_LRCLK_FRC_SHIFT 1 + +#define CS35L36_PDM_MODE_MASK 0x01 +#define CS35L36_PDM_MODE_SHIFT 0 + +#define CS35L36_ASP_FMT_MASK 0x07 +#define CS35L36_ASP_FMT_SHIFT 0 + +#define CS35L36_ASP_RX_WIDTH_MASK 0xFF0000 +#define CS35L36_ASP_RX_WIDTH_SHIFT 16 +#define CS35L36_ASP_TX_WIDTH_MASK 0xFF +#define CS35L36_ASP_TX_WIDTH_SHIFT 0 +#define CS35L36_ASP_WIDTH_16 0x10 +#define CS35L36_ASP_WIDTH_24 0x18 +#define CS35L36_ASP_WIDTH_32 0x20 + +#define CS35L36_ASP_RX1_SLOT_MASK 0x3F +#define CS35L36_ASP_RX1_EN_MASK 0x00010000 +#define CS35L36_ASP_RX1_EN_SHIFT 16 + +#define CS35L36_ASP_TX1_SLOT_MASK 0x3F +#define CS35L36_ASP_TX2_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX2_SLOT_SHIFT 16 +#define CS35L36_ASP_TX3_SLOT_MASK 0x3F +#define CS35L36_ASP_TX4_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX4_SLOT_SHIFT 16 +#define CS35L36_ASP_TX5_SLOT_MASK 0x3F +#define CS35L36_ASP_TX6_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX6_SLOT_SHIFT 16 +#define CS35L36_ASP_TX7_SLOT_MASK 0x3F +#define CS35L36_ASP_TX8_SLOT_MASK 0x3F0000 +#define CS35L36_ASP_TX8_SLOT_SHIFT 16 +#define CS35L36_ASP_TX_HIZ_MASK 0x200000 + +#define CS35L36_APS_TX_SEL_MASK 0x7F + +#define CS35L36_ASP_TX1_EN_MASK 0x01 +#define CS35L36_ASP_TX2_EN_MASK 0x02 +#define CS35L36_ASP_TX2_EN_SHIFT 1 +#define CS35L36_ASP_TX3_EN_MASK 0x04 +#define CS35L36_ASP_TX3_EN_SHIFT 2 +#define CS35L36_ASP_TX4_EN_MASK 0x08 +#define CS35L36_ASP_TX4_EN_SHIFT 3 +#define CS35L36_ASP_TX5_EN_MASK 0x10 +#define CS35L36_ASP_TX5_EN_SHIFT 4 +#define CS35L36_ASP_TX6_EN_MASK 0x20 +#define CS35L36_ASP_TX6_EN_SHIFT 5 +#define CS35L36_ASP_TX7_EN_MASK 0x40 +#define CS35L36_ASP_TX7_EN_SHIFT 6 +#define CS35L36_ASP_TX8_EN_MASK 0x80 +#define CS35L36_ASP_TX8_EN_SHIFT 7 + + +#define CS35L36_PLL_CLK_SEL_MASK 0x07 +#define CS35L36_PLL_CLK_SEL_SHIFT 0 +#define CS35L36_PLLSRC_SCLK 0 +#define CS35L36_PLLSRC_LRCLK 1 +#define CS35L36_PLLSRC_SELF 3 +#define CS35L36_PLLSRC_PDMCLK 4 +#define CS35L36_PLLSRC_MCLK 5 +#define CS35L36_PLLSRC_SWIRE 7 +#define CS35L36_REFCLK_FREQ_MASK 0x7E0 +#define CS35L36_REFCLK_FREQ_SHIFT 5 +#define CS35L36_PLL_OPENLOOP_MASK 0x800 +#define CS35L36_PLL_OPENLOOP_SHIFT 11 +#define CS35L36_PLL_REFCLK_EN_MASK 0x10 +#define CS35L36_PLL_REFCLK_EN_SHIFT 4 + + +#define CS35L36_GLOBAL_FS_MASK 0x1F +#define CS35L36_GLOBAL_FS_SHIFT 0 + +#define CS35L36_HPF_PCM_EN_MASK 0x800 +#define CS35L36_HPF_PCM_EN_SHIFT 15 +#define CS35L36_PCM_RX_SEL_MASK 0x7F +#define CS35L36_PCM_RX_SEL_SHIFT 0 + +#define CS35L36_PCM_RX_SEL_ZERO 0x00 +#define CS35L36_PCM_RX_SEL_PCM 0x08 +#define CS35L36_PCM_RX_SEL_SWIRE 0x10 +#define CS35L36_PCM_RX_SEL_DIAG 0x04 + +#define CS35L36_GLOBAL_EN_MASK 0x01 +#define CS35L36_GLOBAL_EN_SHIFT 0x00 + +#define CS35L36_AMP_PCM_INV_MASK 0x4000 +#define CS35L36_AMP_PCM_INV_SHIFT 14 + +#define CS35L36_AMP_VOL_PCM_MASK 0x3FF8 +#define CS35L36_AMP_VOL_PCM_SHIFT 3 +#define CS35L36_DIGITAL_MUTE 0x04CF + +#define CS35L36_AMP_RAMP_MASK 0x0007 +#define CS35L36_AMP_RAMP_SHIFT 0 + +#define CS35L36_AMP_MUTE_MASK 0x0010 +#define CS35L36_AMP_MUTE_SHIFT 4 + +#define CS35L36_GLOBAL_RESYNC_FS1_MASK 0x00000200 +#define CS35L36_GLOBAL_RESYNC_FS2_MASK 0x00000400 +#define CS35L36_SYNC_GLOBAL_OVR_MASK 0x00000002 +#define CS35L36_SYNC_GLOBAL_OVR_SHIFT 1 + +#define CS35L36_REFCLK_IN_MASK 0x00100000 +#define CS35L36_PLL_UNLOCK_MASK 0x00002000 + +#define CS35L36_ASP_RX_UDF_MASK 0x00000040 +#define CS35L36_ASP_RX_OVF_MASK 0x00000080 + +#define CS35L36_IMON_POL_MASK 0x02 +#define CS35L36_IMON_POL_SHIFT 1 + +#define CS35L36_VMON_POL_MASK 0x01 +#define CS35L36_VMON_POL_SHIFT 0 + +#define CS35L36_PDN_DONE 0x40 +#define CS35L36_PDN_DONE_SHIFT 6 +#define CS35L36_PUP_DONE 0x80 +#define CS35L36_PUP_DONE_SHIFT 7 +#define CS35L36_GLOBAL_EN_ASSRT 0x20 +#define CS35L36_PUP_DONE_IRQ_UNMASK 0x7F +#define CS35L36_PUP_DONE_IRQ_MASK 0xBF + +#define CS35L36_FS1_WINDOW_MASK 0x000007FF +#define CS35L36_FS2_WINDOW_MASK 0x00FFF800 +#define CS35L36_FS2_WINDOW_SHIFT 12 + +#define CS35L36_PLL_FFL_IGAIN_MASK 0x0F +#define CS35L36_PLL_IGAIN_MASK 0x3F0 +#define CS35L36_PLL_IGAIN_SHIFT 4 +#define CS35L36_PLL_IGAIN 0x04 + +#define CS35L36_BST_EN_MASK 0x30 +#define CS35L36_BST_EN 0x02 +#define CS35L36_BST_DIS_VP 0x01 +#define CS35L36_BST_DIS_EXTN 0x00 +#define CS35L36_BST_EN_SHIFT 4 +#define CS35L36_BST_MAN_IPKCOMP_MASK 0x200 +#define CS35L36_BST_MAN_IPKCOMP_SHIFT 9 + +#define CS35L36_BST_MAN_IPKCOMP_EN_MASK 0x100 +#define CS35L36_BST_MAN_IPKCOMP_EN_SHIFT 8 + +#define CS35L36_BST_IPK_MASK 0x7F +#define CS35L36_BST_OVP_THLD_MASK 0x3F +#define CS35L36_BST_OVP_THLD_11V 0x10 +#define CS35L36_BST_OVP_TRIM_MASK 0x00078000 +#define CS35L36_BST_OVP_TRIM_SHIFT 15 +#define CS35L36_BST_OVP_TRIM_11V 0x0C +#define CS35L36_BST_CTRL_LIM_MASK 0x04 +#define CS35L36_BST_CTRL_LIM_SHIFT 2 +#define CS35L36_BST_CTRL_10V_CLAMP 0x96 + +#define CS35L36_NG_AMP_EN_MASK 0x3F00 +#define CS35L36_NG_DELAY_MASK 0x70 +#define CS35L36_NG_DELAY_SHIFT 4 +#define CS35L36_AMP_ZC_SHIFT 10 +#define CS35L36_PDM_LDM_ENTER_SHIFT 3 +#define CS35L36_PDM_LDM_EXIT_SHIFT 4 + +#define CS35L36_BSTCVRT_K1_MASK 0xFF +#define CS35L36_BSTCVRT_K2_MASK 0xFF00 +#define CS35L36_BSTCVRT_K2_SHIFT 8 +#define CS35L36_BSTCVRT_SLOPE_MASK 0xFF00 +#define CS35L36_BSTCVRT_SLOPE_SHIFT 8 +#define CS35L36_BSTCVRT_CCMFREQ_MASK 0x0F +#define CS35L36_BSTCVRT_LBSTVAL_MASK 0x03 +#define CS35L35_BSTCVRT_CTL_MASK 0xFF +#define CS35L35_BSTCVRT_CTL_SEL_MASK 0x03 +#define CS35L36_DCM_AUTO_MASK 0x01 + +#define CS35L36_INT1_MASK_DEFAULT 0xF9BA7FFF +#define CS35L36_INT1_MASK_RESET 0xFFFFFFFF +#define CS35L36_INT3_MASK_DEFAULT 0xFFFFEFFF +#define CS35L36_INT3_MASK_RESET 0xFFFFFFFF + + +#define CS35L36_AMP_SHORT_ERR 0x1000 +#define CS35L36_BST_SHORT_ERR 0x40000 +#define CS35L36_TEMP_WARN 0x2000000 +#define CS35L36_TEMP_ERR 0x4000000 +#define CS35L36_BST_OVP_ERR 0x10000 +#define CS35L36_BST_DCM_UVP_ERR 0x20000 + +#define CS35L36_AMP_SHORT_ERR_RLS 0x02 +#define CS35L36_BST_SHORT_ERR_RLS 0x04 +#define CS35L36_BST_OVP_ERR_RLS 0x08 +#define CS35L36_BST_UVP_ERR_RLS 0x10 +#define CS35L36_TEMP_WARN_ERR_RLS 0x20 +#define CS35L36_TEMP_ERR_RLS 0x40 +#define CS35L36_TEMP_THLD_MASK 0x03 + +#define CS35L36_REV_B0 0xb0 +#define CS35L36_REV_A0 0xa0 +#define CS35L36_B0_PAC_PATCH 0x00DD0102 + +#define CS35L36_OTP_ECC_EN_MASK 0x400 +#define CS35L36_OTP_ECC_EN_SHIFT 10 +#define CS35L36_OTP_RUN_BOOT_MASK 0x01 +#define CS35L36_OTP_BOOT_DONE 0x2000000 +#define CS35L36_PAC_RESET_MASK 0x04 +#define CS35L36_PAC_RESET_SHIFT 2 +#define CS35L36_PAC_STALL_MASK 0x02 +#define CS35L36_PAC_STALL_SHIFT 1 +#define CS35L36_PAC_ENABLE_MASK 0x00000001 +#define CS35L36_PAC_MEM_ACCESS 0x01 +#define CS35L36_PAC_MEM_ACCESS_CLR 0 +#define CS35L36_SOFT_RESET 0x5AAA +#define CS35L36_MCU_BOOT_COMPLETE 0x02 +#define CS35L36_MCU_CONFIG_UNMASK 0x00FEFFFF +#define CS35L36_MCU_CONFIG_CLR 0x00010000 +#define CS35L36_MCU_CONFIG_MASK 0x00FFFFFF +#define CS35L36_GPIO_INT_SEL_MASK 0x0000003B +#define CS35L36_GPIO_INT_SEL_UNMASK 0x0000003A +#define CS35L36_PAC_RESET 0x00000000 +#define CS35L36_OTP_REV_MASK 0x00FF0000 +#define CS35L36_OTP_REV_L37 0x00CC0000 +#define CS35L36_12V_L37 37 +#define CS35L36_10V_L36 36 + +#define CS35L36_VPBR_EN_MASK 0x00001000 +#define CS35L36_VPBR_EN_SHIFT 12 + +#define CS35L36_VPBR_THLD_MASK 0x0000001F +#define CS35L36_VPBR_THLD_SHIFT 0 +#define CS35L36_VPBR_MAX_ATTN_MASK 0x00000F00 +#define CS35L36_VPBR_MAX_ATTN_SHIFT 8 +#define CS35L36_VPBR_ATK_VOL_MASK 0x0000F000 +#define CS35L36_VPBR_ATK_VOL_SHIFT 12 +#define CS35L36_VPBR_ATK_RATE_MASK 0x00070000 +#define CS35L36_VPBR_ATK_RATE_SHIFT 16 +#define CS35L36_VPBR_WAIT_MASK 0x00180000 +#define CS35L36_VPBR_WAIT_SHIFT 19 +#define CS35L36_VPBR_REL_RATE_MASK 0x00E00000 +#define CS35L36_VPBR_REL_RATE_SHIFT 21 +#define CS35L36_VPBR_MUTE_EN_MASK 0x01000000 +#define CS35L36_VPBR_MUTE_EN_SHIFT 24 + +#define CS35L36_OSC_FREQ_TRIM_MASK 0x070 +#define CS35L36_OSC_TRIM_DONE 0x08 + +#define CS35L36_FS1_DEFAULT_VAL 16 +#define CS35L36_FS2_DEFAULT_VAL 36 +#define CS35L36_FS_NOM_6MHZ 6000000 + +#define CS35L36_TEST_UNLOCK1 0x00005555 +#define CS35L36_TEST_UNLOCK2 0x0000AAAA +#define CS35L36_TEST_LOCK1 0x0000CCCC +#define CS35L36_TEST_LOCK2 0x00003333 + +#define CS35L36_PAC_PROG_MEM 512 + +#define CS35L36_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define CS35L36_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +extern const int cs35l36_a0_pac_patch[CS35L36_PAC_PROG_MEM]; + +#endif -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe
by wen yang 02 Feb '19

02 Feb '19

The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 81f2fe2..60f87a0 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.7.4

1 0

[alsa-devel] [PATCH 0/2] ALSA: pcm: implement the anonymous dup v3
by Jaroslav Kysela 01 Feb '19

01 Feb '19

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup The filtering might be refined. It depends on the real requirements. Perhaps, we may create more ioctl groups. Any comments are more than welcome. v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) v3 of the patches: - group integer declarations in snd_pcm_anonymous_dup() - replaced substream->pcm with pcm in snd_pcm_anonymous_dup() - added SNDRV_PCM_PERM_RW check for read/write/readv/writev syscalls Cc: Phil Burk <philburk(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (2): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: implement the ioctl/mmap filter for the anonymous dup include/sound/pcm.h | 9 +-- include/uapi/sound/asound.h | 12 +++- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 ++-- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 158 ++++++++++++++++++++++++++++++++++++++++++-- 6 files changed, 176 insertions(+), 19 deletions(-) -- 2.13.6

6 20

[alsa-devel] [PATCH 0/7] ASoC: topology: fixes and improvements
by Pierre-Louis Bossart 01 Feb '19

01 Feb '19

Module load/unload tests exposed a set of weaknesses in both the SOF and Skylake drivers, some of which are related to the topology framework. This patchset provides a combined set of fixes identified by the two teams. DAPM routes were also not properly handled, leading to confusing issues when the topology resources were released. All these patches were tested on top of the SOF tree, which tracks Mark's branch on a weekly basis. Amadeusz Sławiński (3): ASoC: topology: Reduce number of dereferences when accessing dobj ASoC: topology: Remove widgets from dobj list ASoC: topology: Fix memory leak from soc_tplg_denum_create_texts Bard liao (2): ASoC: topology: fix memory leak in soc_tplg_dapm_widget_create ASoC: topology: unload physical dai link in remove Ranjani Sridharan (2): ASoC: topology: add SND_SOC_DOBJ_GRAPH type for dapm routes ASoC: topology: modify dapm route loading routine and add dapm route unloading include/sound/soc-dapm.h | 2 + include/sound/soc-topology.h | 8 +- sound/soc/soc-topology.c | 165 +++++++++++++++++++++++++++++------ 3 files changed, 144 insertions(+), 31 deletions(-) -- 2.17.1

3 13

[alsa-devel] [PATCH] ALSA: hda - Serialize codec registrations
by Takashi Iwai 01 Feb '19

01 Feb '19

In the current code, the codec registration may happen both at the codec bind time and the end of the controller probe time. In a rare occasion, they race with each other, leading to Oops due to the still uninitialized card device. This patch introduces a simple flag to prevent the codec registration at the codec bind time as long as the controller probe is going on. The controller probe invokes snd_card_register() that does the whole registration task, and we don't need to register each piece beforehand. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_codec.h | 1 + sound/pci/hda/hda_bind.c | 3 ++- sound/pci/hda/hda_intel.c | 2 ++ 3 files changed, 5 insertions(+), 1 deletion(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 7fa48b100936..cc7c8d42d4fd 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -68,6 +68,7 @@ struct hda_bus { unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ + unsigned int bus_probing :1; /* during probing process */ int primary_dig_out_type; /* primary digital out PCM type */ unsigned int mixer_assigned; /* codec addr for mixer name */ diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 9174f1b3a987..1ec706ced75c 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -115,7 +115,8 @@ static int hda_codec_driver_probe(struct device *dev) err = snd_hda_codec_build_controls(codec); if (err < 0) goto error_module; - if (codec->card->registered) { + /* only register after the bus probe finished; otherwise it's racy */ + if (!codec->bus->bus_probing && codec->card->registered) { err = snd_card_register(codec->card); if (err < 0) goto error_module; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e784130ea4e0..e5c49003e75f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2185,6 +2185,7 @@ static int azx_probe_continue(struct azx *chip) int dev = chip->dev_index; int err; + to_hda_bus(bus)->bus_probing = 1; hda->probe_continued = 1; /* bind with i915 if needed */ @@ -2269,6 +2270,7 @@ static int azx_probe_continue(struct azx *chip) if (err < 0) hda->init_failed = 1; complete_all(&hda->probe_wait); + to_hda_bus(bus)->bus_probing = 0; return err; } -- 2.16.4

1 0

Re: [alsa-devel] auto_mute for early patch issue
by Takashi Iwai 01 Feb '19

01 Feb '19

On Fri, 01 Feb 2019 11:21:58 +0100, Kailang wrote: > > Hi Takashi, > > Got it. Thanks. > > Could i need to make a patch for moving spec->suppress_auto_mute check? > > Because Chrome OS need to back porting from upstream commit patches. > So, the better solution was moving spec->suppress_auto_mute check. Right? After checking the whole code, I think what we can do now even safely is to clean up a patch like below. This will reduce lots of code. With that applied, I don't think we need the change in hda_generic.c any longer. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda/realtek - Use a common helper for hp pin reference Replace the open-codes in many places with a new common helper for performing the same thing: referring to the primary headphone pin. This eventually fixes the potentially missing headphone pin on some weird devices, too. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 96 +++++++++++-------------------------------- 1 file changed, 24 insertions(+), 72 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5b1a71fb909c..6df758adff84 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -515,6 +515,15 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) } } +/* get a primary headphone pin if available */ +static hda_nid_t alc_get_hp_pin(struct alc_spec *spec) +{ + if (spec->gen.autocfg.hp_pins[0]) + return spec->gen.autocfg.hp_pins[0]; + if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) + return spec->gen.autocfg.line_out_pins[0]; + return 0; +} /* * Realtek SSID verification @@ -725,9 +734,7 @@ static int alc_subsystem_id(struct hda_codec *codec, const hda_nid_t *ports) * 15 : 1 --> enable the function "Mute internal speaker * when the external headphone out jack is plugged" */ - if (!spec->gen.autocfg.hp_pins[0] && - !(spec->gen.autocfg.line_out_pins[0] && - spec->gen.autocfg.line_out_type == AUTO_PIN_HP_OUT)) { + if (!alc_get_hp_pin(spec)) { hda_nid_t nid; tmp = (ass >> 11) & 0x3; /* HP to chassis */ nid = ports[tmp]; @@ -2959,15 +2966,10 @@ static void alc282_restore_default_value(struct hda_codec *codec) static void alc282_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; int coef78; - if (!spec->gen.autocfg.hp_outs && spec->gen.suppress_auto_mute) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - alc282_restore_default_value(codec); if (!hp_pin) @@ -3001,15 +3003,10 @@ static void alc282_init(struct hda_codec *codec) static void alc282_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; int coef78; - if (!spec->gen.autocfg.hp_outs && spec->gen.suppress_auto_mute) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) { alc269_shutup(codec); return; @@ -3084,14 +3081,9 @@ static void alc283_restore_default_value(struct hda_codec *codec) static void alc283_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!spec->gen.autocfg.hp_outs) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - alc283_restore_default_value(codec); if (!hp_pin) @@ -3125,14 +3117,9 @@ static void alc283_init(struct hda_codec *codec) static void alc283_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!spec->gen.autocfg.hp_outs) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) { alc269_shutup(codec); return; @@ -3166,14 +3153,9 @@ static void alc283_shutup(struct hda_codec *codec) static void alc256_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!spec->gen.autocfg.hp_outs && spec->gen.suppress_auto_mute) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) return; @@ -3207,14 +3189,9 @@ static void alc256_init(struct hda_codec *codec) static void alc256_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!spec->gen.autocfg.hp_outs && spec->gen.suppress_auto_mute) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) { alc269_shutup(codec); return; @@ -3248,14 +3225,9 @@ static void alc256_shutup(struct hda_codec *codec) static void alc225_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp1_pin_sense, hp2_pin_sense; - if (!spec->gen.autocfg.hp_outs && spec->gen.suppress_auto_mute) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) return; @@ -3296,14 +3268,9 @@ static void alc225_init(struct hda_codec *codec) static void alc225_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp1_pin_sense, hp2_pin_sense; - if (!spec->gen.autocfg.hp_outs && spec->gen.suppress_auto_mute) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) { alc269_shutup(codec); return; @@ -3345,14 +3312,9 @@ static void alc225_shutup(struct hda_codec *codec) static void alc_default_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!spec->gen.autocfg.hp_outs && spec->gen.suppress_auto_mute) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) return; @@ -3379,14 +3341,9 @@ static void alc_default_init(struct hda_codec *codec) static void alc_default_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); bool hp_pin_sense; - if (!spec->gen.autocfg.hp_outs && spec->gen.suppress_auto_mute) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) { alc269_shutup(codec); return; @@ -3416,14 +3373,9 @@ static void alc_default_shutup(struct hda_codec *codec) static void alc294_hp_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); int i, val; - if (!spec->gen.autocfg.hp_outs && spec->gen.suppress_auto_mute) { - if (spec->gen.autocfg.line_out_type == AC_JACK_HP_OUT) - hp_pin = spec->gen.autocfg.line_out_pins[0]; - } - if (!hp_pin) return; @@ -4825,7 +4777,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; - hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + hda_nid_t hp_pin = alc_get_hp_pin(spec); int new_headset_mode; @@ -5104,7 +5056,7 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int hp_pin = spec->gen.autocfg.hp_pins[0]; + int hp_pin = alc_get_hp_pin(spec); /* Prevent pop noises when headphones are plugged in */ snd_hda_codec_write(codec, hp_pin, 0, @@ -5197,7 +5149,7 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PROBE) { int mic_pin = find_ext_mic_pin(codec); - int hp_pin = spec->gen.autocfg.hp_pins[0]; + int hp_pin = alc_get_hp_pin(spec); if (snd_BUG_ON(!mic_pin || !hp_pin)) return; -- 2.16.4

1 0

Re: [alsa-devel] auto_mute for early patch issue
by Kailang 01 Feb '19

01 Feb '19

Hi Takashi, I move check line to below. It will show value for hp_pins[0]. As I know, search from verb table in function snd_hda_parse_pin_defcfg(). It will fill pins at auto_pin_cfg. Why it need to run below function to get cfg->hp_pins? static int check_auto_mute_availability(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; int present = 0; int i, err; - if (spec->suppress_auto_mute) - return 0; if (cfg->hp_pins[0]) present++; if (cfg->line_out_pins[0]) present++; if (cfg->speaker_pins[0]) present++; if (present < 2) /* need two different output types */ return 0; if (!cfg->speaker_pins[0] && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { memcpy(cfg->speaker_pins, cfg->line_out_pins, sizeof(cfg->speaker_pins)); cfg->speaker_outs = cfg->line_outs; } if (!cfg->hp_pins[0] && cfg->line_out_type == AUTO_PIN_HP_OUT) { memcpy(cfg->hp_pins, cfg->line_out_pins, sizeof(cfg->hp_pins)); cfg->hp_outs = cfg->line_outs; } + if (spec->suppress_auto_mute) + return 0; ...... ..... ...... } BR, Kailang -----Original Message----- From: Kailang Sent: Thursday, January 31, 2019 12:08 PM To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> Subject: auto_mute for early patch issue Hi Takashi, Chrome OS use early patch firmware file to load model and use auto_mute =no hint options. I find an issue for early patch issue about hint option auto_mute = no. This will cause all spec->init_hook can't get hp_pin values. For example: alc225_init() alc225_shutup() alc256_init() alc256_shutup() Early patch file hint auto_mute = no equal spec->suppress_auto_mute = 1 It will return function check_auto_mute_availability() in had_generic.c. hp_pin = spec->gen.autocfg.hp_pins[0]; ==> It will get value 0. I think this issue was for a long time. Please see the attach dmesg. One is kill hint auto_mute option. BR, Kailang

2 7

[alsa-devel] [PATCH 1/1] ASoC: rsnd: allow to register kctrl with different cfg
by Jiada Wang 01 Feb '19

01 Feb '19

Currently kctrl with same name can only be created once, but in "MIX Playback Volume" case, same kctrl is used to control different MIX volume for different dai-link. this patch by check kctrl's cfg, allow to create kctrl with same name but different cfg. Fixes: commit 9c698e8481a1 ("ASoC: rsnd: tidyup registering method for rsnd_kctrl_new()") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 79659f7b32b9..909ee0948da1 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1526,14 +1526,15 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, int ret; /* - * 1) Avoid duplicate register (ex. MIXer case) + * 1) Avoid duplicate register (ex. DVC case) * 2) re-register if card was rebinded */ list_for_each_entry(kctrl, &card->controls, list) { struct rsnd_kctrl_cfg *c = kctrl->private_data; if (strcmp(kctrl->id.name, name) == 0 && - c->mod == mod) + c->mod == mod && + &c->val == &cfg->val) return 0; } -- 2.19.2

2 1

[alsa-devel] [PATCH] Headset microphone support for System76 darp5
by Jeremy Soller 31 Jan '19

31 Jan '19

Headset microphone support for System76 darp5 Signed-off-by: Jeremy Soller <jeremy(a)system76.com> --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b4f472157ebd..15fedb0eeae1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5591,6 +5591,7 @@ enum { ALC294_FIXUP_ASUS_HEADSET_MIC, ALC294_FIXUP_ASUS_SPK, ALC225_FIXUP_HEADSET_JACK, + ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6537,6 +6538,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, }, + [ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6606,6 +6616,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY), + SND_PCI_QUIRK(0x1558, 0x1325, "System76", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), /* ALC282 */ SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),

2 4

[alsa-devel] [PATCH 0/3] sound: 3 cosmetic patches
by Guennadi Liakhovetski 31 Jan '19

31 Jan '19

From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> 3 trivial cosmetic patches. Guennadi Liakhovetski (3): sound: (cosmetic) remove a superfluous function declaration asoc: (cosmetic) simplify a condition asoc: (cosmetic) simplify a function sound/core/pcm_native.c | 2 -- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-topology.c | 6 +----- 3 files changed, 3 insertions(+), 9 deletions(-) -- 2.17.1

2 4

[alsa-devel] [PATCH] wcd9335: Remove unnecessary null checks in wcd9335_set_channel_map
by Nathan Chancellor 31 Jan '19

31 Jan '19

Clang warns: sound/soc/codecs/wcd9335.c:2004:11: warning: address of array 'wcd->rx_chs' will always evaluate to 'true' [-Wpointer-bool-conversion] if (wcd->rx_chs) { ~~ ~~~~~^~~~~~ sound/soc/codecs/wcd9335.c:2012:11: warning: address of array 'wcd->tx_chs' will always evaluate to 'true' [-Wpointer-bool-conversion] if (wcd->tx_chs) { ~~ ~~~~~^~~~~~ 2 warnings generated. Fixes: 20aedafdf492 ("ASoC: wcd9335: add support to wcd9335 codec") Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/soc/codecs/wcd9335.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 3878187bb512..981f88a5f615 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2001,20 +2001,16 @@ static int wcd9335_set_channel_map(struct snd_soc_dai *dai, return -EINVAL; } - if (wcd->rx_chs) { - wcd->num_rx_port = rx_num; - for (i = 0; i < rx_num; i++) { - wcd->rx_chs[i].ch_num = rx_slot[i]; - INIT_LIST_HEAD(&wcd->rx_chs[i].list); - } + wcd->num_rx_port = rx_num; + for (i = 0; i < rx_num; i++) { + wcd->rx_chs[i].ch_num = rx_slot[i]; + INIT_LIST_HEAD(&wcd->rx_chs[i].list); } - if (wcd->tx_chs) { - wcd->num_tx_port = tx_num; - for (i = 0; i < tx_num; i++) { - wcd->tx_chs[i].ch_num = tx_slot[i]; - INIT_LIST_HEAD(&wcd->tx_chs[i].list); - } + wcd->num_tx_port = tx_num; + for (i = 0; i < tx_num; i++) { + wcd->tx_chs[i].ch_num = tx_slot[i]; + INIT_LIST_HEAD(&wcd->tx_chs[i].list); } return 0; -- 2.20.1

2 1

[alsa-devel] auto_mute for early patch issue
by Kailang 31 Jan '19

31 Jan '19

Hi Takashi, Chrome OS use early patch firmware file to load model and use auto_mute =no hint options. I find an issue for early patch issue about hint option auto_mute = no. This will cause all spec->init_hook can't get hp_pin values. For example: alc225_init() alc225_shutup() alc256_init() alc256_shutup() Early patch file hint auto_mute = no equal spec->suppress_auto_mute = 1 It will return function check_auto_mute_availability() in had_generic.c. hp_pin = spec->gen.autocfg.hp_pins[0]; ==> It will get value 0. I think this issue was for a long time. Please see the attach dmesg. One is kill hint auto_mute option. BR, Kailang

1 0

Re: [alsa-devel] linux-next: Tree for Jan 30 (sound/soc/ti/davinci-mcasp.c)
by Randy Dunlap 30 Jan '19

30 Jan '19

On 1/29/19 11:39 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20190129: > on i386: ld: sound/soc/ti/davinci-mcasp.o: in function `davinci_mcasp_probe': davinci-mcasp.c:(.text+0x1d99): undefined reference to `edma_pcm_platform_register' Full randconfig file is attached. PS: Please fix this line in the source file also: * Steve Chen <schen@.mvista.com> (extra '.') -- ~Randy

1 0

[alsa-devel] [PATCH v3 00/14] Sound Open Firmware (SOF) core
by Pierre-Louis Bossart 30 Jan '19

30 Jan '19

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core and utilities. Support for Intel devices is provided as a follow-up series. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Main changes since v2 (end of August): There are honestly too many changes to be listed... The biggest changes are the introduction of an ABI check and runtime_pm, as well as countless improvements coming from product validation. We will provide a revision history starting with the next update. Precisions: Since I've been maintaining the SOF kernel parts for most of 2018, Liam and I mutually agreed that it would be more efficient for me to send the patches upstream to avoid any disconnects. The code in this patchset is directly squashed from the SOF development branch [1], which tracks Mark Brown's for-next branch on a weekly basis and the configurations used for testing are based on the defconfigs at [2] Full disclosure on known limitation and issues (full list at [3]) a) the interaction with ASoC is not always perfect, there are a couple of points where a better solution is desired (use of private data, support for link DMA, etc). The known points are explicitly documented in the code and will be updated. To be clearer, it's our belief that SOF does not cripple ASoC in any way, and that merging this SOF core does not harm others. It's just complicated to get things right, especially on a couple of cases where the topology and DPCM frameworks generate complex flows that very few people understand. b) The get/put methods for controls generate an IPC and possibly a wake-up. This is not optimal and is being fixed. c) The debugfs parts will be updated for runtime_pm parts d) The topology free, firmware release and Hdaudio exit sequences will be updated. It is our intention to fully support module load/unload and free all resources. e) runtime_pm is being hardened and the use of SMART_SUSPEND was suggested. Thank you for reviews and comments, we appreciate the time spent commenting on this large patchset. Thanks in particular to Alan Cox and Andy Shevchenko for their comments on an earlier version. Pierre [1] https://github.com/thesofproject/linux [2] https://github.com/thesofproject/kconfig [3] https://github.com/thesofproject/linux/issues Liam Girdwood (12): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware logger support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add PM support ASoC: SOF: Add Nocodec machine driver support Pierre-Louis Bossart (2): ASoC: SOF: Add xtensa support ASoC: SOF: Add utils include/sound/soc.h | 3 + include/sound/sof.h | 91 + include/sound/sof/control.h | 125 ++ include/sound/sof/dai-intel.h | 175 ++ include/sound/sof/dai.h | 75 + include/sound/sof/header.h | 158 ++ include/sound/sof/info.h | 118 ++ include/sound/sof/pm.h | 48 + include/sound/sof/stream.h | 149 ++ include/sound/sof/topology.h | 256 +++ include/sound/sof/trace.h | 66 + include/sound/sof/xtensa.h | 44 + include/uapi/sound/sof/abi.h | 63 + include/uapi/sound/sof/eq.h | 164 ++ include/uapi/sound/sof/fw.h | 67 + include/uapi/sound/sof/header.h | 27 + include/uapi/sound/sof/manifest.h | 188 ++ include/uapi/sound/sof/tokens.h | 95 + include/uapi/sound/sof/tone.h | 21 + include/uapi/sound/sof/trace.h | 93 + sound/soc/sof/control.c | 421 +++++ sound/soc/sof/core.c | 398 +++++ sound/soc/sof/debug.c | 176 ++ sound/soc/sof/ipc.c | 813 +++++++++ sound/soc/sof/loader.c | 322 ++++ sound/soc/sof/nocodec.c | 81 + sound/soc/sof/ops.c | 200 +++ sound/soc/sof/ops.h | 379 ++++ sound/soc/sof/pcm.c | 755 ++++++++ sound/soc/sof/pm.c | 403 +++++ sound/soc/sof/sof-priv.h | 574 ++++++ sound/soc/sof/topology.c | 2775 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 295 +++ sound/soc/sof/utils.c | 173 ++ sound/soc/sof/xtensa/Kconfig | 3 + sound/soc/sof/xtensa/Makefile | 5 + sound/soc/sof/xtensa/core.c | 152 ++ 37 files changed, 9951 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/sound/sof/control.h create mode 100644 include/sound/sof/dai-intel.h create mode 100644 include/sound/sof/dai.h create mode 100644 include/sound/sof/header.h create mode 100644 include/sound/sof/info.h create mode 100644 include/sound/sof/pm.h create mode 100644 include/sound/sof/stream.h create mode 100644 include/sound/sof/topology.h create mode 100644 include/sound/sof/trace.h create mode 100644 include/sound/sof/xtensa.h create mode 100644 include/uapi/sound/sof/abi.h create mode 100644 include/uapi/sound/sof/eq.h create mode 100644 include/uapi/sound/sof/fw.h create mode 100644 include/uapi/sound/sof/header.h create mode 100644 include/uapi/sound/sof/manifest.h create mode 100644 include/uapi/sound/sof/tokens.h create mode 100644 include/uapi/sound/sof/tone.h create mode 100644 include/uapi/sound/sof/trace.h create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/nocodec.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/pm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c create mode 100644 sound/soc/sof/utils.c create mode 100644 sound/soc/sof/xtensa/Kconfig create mode 100644 sound/soc/sof/xtensa/Makefile create mode 100644 sound/soc/sof/xtensa/core.c -- 2.17.1

7 91

[alsa-devel] [PATCH 1/2] ASoC: pcm512x: Implement the set_bclk_ratio interface
by Dimitris Papavasiliou 30 Jan '19

30 Jan '19

Some boards, such as the HiFiBerry DAC+ Pro, use a pair of external oscillators, to generate 44.1 or 48kHz multiples and are forced to resort to hacks [1] in order to support 24-bit data without ending up with fractional dividers. This patch allows the machine driver to use 32-bit frames for 24-bit data to avoid such issues. Although the datasheet (p. 15) seems to suggest that only a handful of ratios are supported, it's not very explicit about it, so we allow the full range of values supported by the underlying register in the callback, to avoid needlessly rejecting potentially usable configurations. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143442.h… Signed-off-by: Dimitris Papavasiliou <dpapavas(a)gmail.com> --- Notes: This is essentially the same patch submitted previously as an RFC[1]. It has been split in two, in order to separate the set_bclk_ratio implementation from the clocking calculation fix and some conservative checking has been added to the argument of the callback. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2019-January/144213.ht… sound/soc/codecs/pcm512x.c | 28 ++++++++++++++++++++++++---- 1 file changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 6cb1653be804..b63d9392bae3 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -55,6 +55,7 @@ struct pcm512x_priv { unsigned long overclock_dsp; int mute; struct mutex mutex; + unsigned int bclk_ratio; }; /* @@ -915,10 +916,15 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, int fssp; int gpio; - lrclk_div = snd_soc_params_to_frame_size(params); - if (lrclk_div == 0) { - dev_err(dev, "No LRCLK?\n"); - return -EINVAL; + if (pcm512x->bclk_ratio > 0) { + lrclk_div = pcm512x->bclk_ratio; + } else { + lrclk_div = snd_soc_params_to_frame_size(params); + + if (lrclk_div == 0) { + dev_err(dev, "No LRCLK?\n"); + return -EINVAL; + } } if (!pcm512x->pll_out) { @@ -1383,6 +1389,19 @@ static int pcm512x_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int pcm512x_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + + if (ratio > 256) + return -EINVAL; + + pcm512x->bclk_ratio = ratio; + + return 0; +} + static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_component *component = dai->component; @@ -1438,6 +1457,7 @@ static const struct snd_soc_dai_ops pcm512x_dai_ops = { .hw_params = pcm512x_hw_params, .set_fmt = pcm512x_set_fmt, .digital_mute = pcm512x_digital_mute, + .set_bclk_ratio = pcm512x_set_bclk_ratio, }; static struct snd_soc_dai_driver pcm512x_dai = { -- 2.11.0

3 5

[alsa-devel] [PATCH] ASoC: rockchip: fix small typo
by lsh 30 Jan '19

30 Jan '19

codec_dai -> cpu_dai Signed-off-by: lsh <lshua312(a)163.com> --- sound/soc/rockchip/rockchip_rt5645.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 881c324..f88adf0 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -95,14 +95,16 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, SND_SOC_CLOCK_OUT); if (ret < 0) { - dev_err(codec_dai->dev, "Can't set codec clock %d\n", ret); + dev_err(cpu_dai->dev, + "Can't set clock for cpu_dai %d\n", ret); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (ret < 0) { - dev_err(codec_dai->dev, "Can't set codec clock %d\n", ret); + dev_err(codec_dai->dev, + "Can't set clock for codec_dai %d\n", ret); return ret; } -- 1.9.1

1 0

[alsa-devel] [PATCH] ALSA: hda/tegra: enable clock during probe
by Sameer Pujar 30 Jan '19

30 Jan '19

If CONFIG_PM is disabled or runtime PM calls are forbidden, the clocks will not be ON. This could cause issue during probe, where hda init setup is done. This patch checks whether runtime PM is enabled or not. If disabled, clocks are enabled in probe() and disabled in remove() This patch does following minor changes as cleanup, * return code check for pm_runtime_get_sync() to take care of failure and exit gracefully. * In remove path runtime PM is disabled before calling snd_card_free(). * hda_tegra_disable_clocks() is moved out of CONFIG_PM_SLEEP check. * runtime PM callbacks moved out of CONFIG_PM check Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Reviewed-by: Ravindra Lokhande <rlokhande(a)nvidia.com> Reviewed-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 31 ++++++++++++++++++++++--------- 1 file changed, 22 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index c8d18dc..155c2f5 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -219,7 +219,6 @@ static int hda_tegra_enable_clocks(struct hda_tegra *data) return rc; } -#ifdef CONFIG_PM_SLEEP static void hda_tegra_disable_clocks(struct hda_tegra *data) { clk_disable_unprepare(data->hda2hdmi_clk); @@ -227,6 +226,7 @@ static void hda_tegra_disable_clocks(struct hda_tegra *data) clk_disable_unprepare(data->hda_clk); } +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -257,7 +257,6 @@ static int hda_tegra_resume(struct device *dev) } #endif /* CONFIG_PM_SLEEP */ -#ifdef CONFIG_PM static int hda_tegra_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -283,7 +282,7 @@ static int hda_tegra_runtime_resume(struct device *dev) int rc; rc = hda_tegra_enable_clocks(hda); - if (rc != 0) + if (rc) return rc; if (chip && chip->running) { hda_tegra_init(hda); @@ -292,7 +291,6 @@ static int hda_tegra_runtime_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops hda_tegra_pm = { SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) @@ -555,6 +553,13 @@ static int hda_tegra_probe(struct platform_device *pdev) if (!azx_has_pm_runtime(chip)) pm_runtime_forbid(hda->dev); + /* explicit resume if runtime PM is disabled */ + if (!pm_runtime_enabled(hda->dev)) { + err = hda_tegra_runtime_resume(hda->dev); + if (err) + goto out_free; + } + schedule_work(&hda->probe_work); return 0; @@ -571,7 +576,14 @@ static void hda_tegra_probe_work(struct work_struct *work) struct platform_device *pdev = to_platform_device(hda->dev); int err; - pm_runtime_get_sync(hda->dev); + err = pm_runtime_get_sync(hda->dev); + if (err < 0) { + dev_err(hda->dev, + "failed in pm_runtime_get_syc with err = %d\n", + err); + return; + } + err = hda_tegra_first_init(chip, pdev); if (err < 0) goto out_free; @@ -599,12 +611,13 @@ static void hda_tegra_probe_work(struct work_struct *work) static int hda_tegra_remove(struct platform_device *pdev) { - int ret; - - ret = snd_card_free(dev_get_drvdata(&pdev->dev)); pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) { + hda_tegra_runtime_suspend(&pdev->dev); + pm_runtime_set_suspended(&pdev->dev); + } - return ret; + return snd_card_free(dev_get_drvdata(&pdev->dev)); } static void hda_tegra_shutdown(struct platform_device *pdev) -- 2.7.4

3 12

[alsa-devel] [PATCH 0/2] ALSA: pcm: implement the anonymous dup v2
by Jaroslav Kysela 30 Jan '19

30 Jan '19

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup The filtering might be refined. It depends on the real requirements. Perhaps, we may create more ioctl groups. Any comments are more than welcome. v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) Cc: Phil Burk <philburk(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (2): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: implement the ioctl/mmap filter for the anonymous dup include/sound/pcm.h | 9 +-- include/uapi/sound/asound.h | 12 +++- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 ++-- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 153 ++++++++++++++++++++++++++++++++++++++++++-- 6 files changed, 171 insertions(+), 19 deletions(-) -- 2.13.6

2 6

[alsa-devel] [PATCH] ALSA: pcm: Drop unused snd_pcm_substream.file field
by Takashi Iwai 30 Jan '19

30 Jan '19

It's assigned but nowhere used. Let's remove it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 1 - sound/core/oss/pcm_oss.c | 1 - sound/core/pcm_native.c | 4 +--- 3 files changed, 1 insertion(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2c30c1ad1b0d..a20d3a48df00 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -470,7 +470,6 @@ struct snd_pcm_substream { struct snd_pcm_group self_group; /* fake group for non linked substream (with substream lock inside) */ struct snd_pcm_group *group; /* pointer to current group */ /* -- assigned files -- */ - void *file; int ref_count; atomic_t mmap_count; unsigned int f_flags; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 467039b342b5..d5b0d7ba83c4 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2427,7 +2427,6 @@ static int snd_pcm_oss_open_file(struct file *file, } pcm_oss_file->streams[idx] = substream; - substream->file = pcm_oss_file; snd_pcm_oss_init_substream(substream, &setup[idx], minor); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 26afb6b0889a..63640d3af9db 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2452,10 +2452,8 @@ static int snd_pcm_open_file(struct file *file, return -ENOMEM; } pcm_file->substream = substream; - if (substream->ref_count == 1) { - substream->file = pcm_file; + if (substream->ref_count == 1) substream->pcm_release = pcm_release_private; - } file->private_data = pcm_file; return 0; -- 2.16.4

2 1

[alsa-devel] [PATCH 0/3] ALSA: pcm: implement the anonymous dup
by Jaroslav Kysela 30 Jan '19

30 Jan '19

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup The filtering might be refined. It depends on the real requirements. Perhaps, we may create more ioctl groups. Any comments are more than welcome. Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (3): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: remove the file member from struct pcm ALSA: pcm: implement the ioctl/mmap filter for the anonymous dup include/sound/pcm.h | 10 +-- include/uapi/sound/asound.h | 12 +++- sound/core/oss/pcm_oss.c | 3 +- sound/core/pcm.c | 48 +++++++++----- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 154 +++++++++++++++++++++++++++++++++++++++++--- 6 files changed, 195 insertions(+), 33 deletions(-) -- 2.13.6

3 8

[alsa-devel] [RFC PATCH V2 0/2] ASoC: mediatek: add btcvsd driver
by KaiChieh Chuang 30 Jan '19

30 Jan '19

The driver function for transferring/receiving BT encoded data to/from BT hardware. V2: prevent noisy log, dev_info -> dev_dbg use simple number control for Switch control *** BLURB HERE *** KaiChieh Chuang (2): ASoC: mediatek: add btcvsd driver ASoC: mediatek: add documents for btcvsd driver .../bindings/sound/mtk-btcvsd-snd.txt | 24 + sound/soc/mediatek/Kconfig | 9 + sound/soc/mediatek/common/Makefile | 2 + sound/soc/mediatek/common/mtk-btcvsd.c | 1364 +++++++++++++++++ 4 files changed, 1399 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt create mode 100644 sound/soc/mediatek/common/mtk-btcvsd.c -- 2.18.0

1 2

Re: [alsa-devel] hp_pin was NULL value
by Takashi Iwai 30 Jan '19

30 Jan '19

On Wed, 09 Jan 2019 13:32:51 +0100, Kailang wrote: > > > Hi Takashi, > > > OK! I will test it. > So, if ALC225 need to do this. I will assigned it to alc225_init(). Right? > > >>But ALC294 (and ALC700) have no init_hook, so far? > > spec->init_hook = alc_default_init; > It has default init_hook. I use this to do standard headphone depop function. Ah, OK. Then we need a call chain from alc294_init(), static void alc294_init() { coef(); alc294_hp_init(); alc_default_init(); } thanks, Takashi > > BR, > Kailang > ________________________________________ > 從: Takashi Iwai [tiwai(a)suse.de] > 寄件日期: 2019年1月9日下午 07:29 > 至: Kailang > 副本: (alsa-devel(a)alsa-project.org) > 主旨: Re: hp_pin was NULL value > > On Wed, 09 Jan 2019 12:19:14 +0100, > Kailang wrote: > > > > Hi Takashi, > > > > But spec->init_hook will call at resume back. > > alc294_hp_init() function just need to call at one time. > > As already mentioned, even if it has to be called only once and does > something for the hardware, it'll be missing if the machine goes > hibernate (S4) and resume. In S4, the machine boots from scratch and > switches immediately to the saved image that calls resume callback. > What's worse for us is that this switch happens usually in initrd, > i.e. before the sound driver gets loaded. So, the machine goes to > resume without processing alc294_hp_init(). > > IOW, even if it's needed only once, you'll have to call it at resume > unless it has a significant drawback. > > (Strictly speaking, we may use thaw PM callback, but the HD-audio > codec driver doesn't use it, so there is only resume callback for > now.) > > > And it need to call it before call spec->init_hook. > > But ALC294 (and ALC700) have no init_hook, so far? > > > ALC225 also need to call similar procedure. I will post it later. > > OK, thanks. > > > Takashi > > > > > BR, > > Kailang > > ________________________________________ > > 從: Takashi Iwai [tiwai(a)suse.de] > > 寄件日期: 2019年1月9日下午 05:56 > > 至: Kailang > > 副本: (alsa-devel(a)alsa-project.org) > > 主旨: Re: hp_pin was NULL value > > > > On Wed, 09 Jan 2019 10:45:25 +0100, > > Kailang wrote: > > > > > > >>But this made me wonder whether we don't need to call this function at resume as well? Currently it's called only at probing. > > > Yes, it only call at probing. Just need to call it at boot time. > > > > > > -----Original Message----- > > > From: Takashi Iwai <tiwai(a)suse.de> > > > Sent: Wednesday, January 9, 2019 5:43 PM > > > To: Kailang <kailang(a)realtek.com> > > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > > Subject: Re: hp_pin was NULL value > > > > > > On Wed, 09 Jan 2019 10:31:33 +0100, > > > Kailang wrote: > > > > > > > > Hi Takashi, > > > > > > > > Could I move the alc294_hp_init(codec) to below line. > > > > Because hp_pin = spec->gen.autocfg.hp_pins[0] was null value when alc294_hp_init(codec) at original line. > > > > Or move alc269_parse_auto_config() upward. > > > > > > It looks OK to me. But this made me wonder whether we don't need to call this function at resume as well? Currently it's called only at probing. > > > > Hrm, but it modifies many COEFs, and are these all preserved with > > suspend? For example, there is also hibernation (S4), which is a > > switch from the boot without the audio driver initialization. Then > > we'll resume from the uninitialized state. I guess other COEF changes > > in that area should be moved as the additional init hook as well. > > > > Actually, if we do call this at resume, the change would be easier, a > > patch like below. > > > > > > thanks, > > > > Takashi > > > > --- a/sound/pci/hda/patch_realtek.c > > +++ b/sound/pci/hda/patch_realtek.c > > @@ -7404,6 +7404,20 @@ static void alc294_hp_init(struct hda_codec *codec) > > msleep(50); > > } > > > > +static void alc294_init(struct hda_codec *codec) > > +{ > > + /* UAJ MIC Vref control by verb */ > > + alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); > > + alc294_hp_init(codec); > > +} > > + > > +static void alc700_init(struct hda_codec *codec) > > +{ > > + /* Combo jack auto trigger control */ > > + alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); > > + alc294_hp_init(codec); > > +} > > + > > /* > > */ > > static int patch_alc269(struct hda_codec *codec) > > @@ -7528,8 +7542,7 @@ static int patch_alc269(struct hda_codec *codec) > > case 0x10ec0294: > > spec->codec_variant = ALC269_TYPE_ALC294; > > spec->gen.mixer_nid = 0; /* ALC2x4 does not have any loopback mixer path */ > > - alc_update_coef_idx(codec, 0x6b, 0x0018, (1<<4) | (1<<3)); /* UAJ MIC Vref control by verb */ > > - alc294_hp_init(codec); > > + spec->init_hook = alc294_init; > > break; > > case 0x10ec0300: > > spec->codec_variant = ALC269_TYPE_ALC300; > > @@ -7540,8 +7553,7 @@ static int patch_alc269(struct hda_codec *codec) > > case 0x10ec0703: > > spec->codec_variant = ALC269_TYPE_ALC700; > > spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ > > - alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ > > - alc294_hp_init(codec); > > + spec->init_hook = alc700_init; > > break; > > > > } > > > > > > ------Please consider the environment before printing this e-mail.

2 18

[alsa-devel] [PATCH] ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check
by Kuninori Morimoto 30 Jan '19

30 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") added new rsnd_ssi_prepare() and moved rsnd_ssi_master_clk_start() to .prepare. But, ssi user count (= ssi->usrcnt) is incremented at .init (= rsnd_ssi_init()). Because of these timing exchange, ssi->usrcnt check at rsnd_ssi_master_clk_start() should be adjusted. Otherwise, 2nd master clock setup will be no check. This patch fixup this issue. Fixes: commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Reported-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> Reported-by: Valentine Barshak <valentine.barshak(a)cogentembedded.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 45ef295..f5afab6 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -286,7 +286,7 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; - if (ssi->usrcnt > 1) { + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); return -EINVAL; -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda - Fix suspend/resume calls
by Rodrigo Vivi 29 Jan '19

29 Jan '19

Besides the compilation issue the resume function was calling the force_suspend. Fixes: d81f8c6b6ced ("ALSA: hda - Record the current power state before suspend/resume calls") Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Rodrigo Vivi <rodrigo.vivi(a)intel.com> --- sound/pci/hda/hda_codec.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 43bcc6f1ea8c..0b86aa2b649c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2894,6 +2894,8 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) */ static void hda_call_codec_resume(struct hda_codec *codec) { + struct device *dev = hda_codec_dev(codec); + snd_hdac_enter_pm(&codec->core); if (codec->core.regmap) regcache_mark_dirty(codec->core.regmap); @@ -2955,10 +2957,16 @@ static int hda_codec_pm_suspend(struct device *dev) return pm_runtime_force_suspend(dev); } +static int hda_codec_pm_freeze(struct device *dev) +{ + dev->power.power_state = PMSG_FREEZE; + return pm_runtime_force_suspend(dev); +} + static int hda_codec_pm_resume(struct device *dev) { dev->power.power_state = PMSG_RESUME; - return pm_runtime_force_suspend(dev); + return pm_runtime_force_resume(dev); } static int hda_codec_pm_thaw(struct device *dev) -- 2.20.1

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[alsa-devel] [PATCH] ALSA: hda - Use standard device registration for beep
by Takashi Iwai 29 Jan '19

29 Jan '19

Currently the registration and free of beep input device was done manually from the register and the disconnect callbacks of the assigned codec object. This seems working in most cases, but this may be a cause of some races at probe. Moreover, due to these manual calls, the total code became unnecessarily lengthy. This patch rewrites the beep registration code to follow the standard sound device object style. This allows us reducing the code, in addition to avoiding the nested device registration calls. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_beep.c | 151 ++++++++++++++++++++++------------------------ sound/pci/hda/hda_beep.h | 5 -- sound/pci/hda/hda_codec.c | 10 --- 3 files changed, 72 insertions(+), 94 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 066b5b59c4d7..b7d9160ed868 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -127,44 +127,6 @@ static void turn_off_beep(struct hda_beep *beep) } } -static void snd_hda_do_detach(struct hda_beep *beep) -{ - if (beep->registered) - input_unregister_device(beep->dev); - else - input_free_device(beep->dev); - beep->dev = NULL; - turn_off_beep(beep); -} - -static int snd_hda_do_attach(struct hda_beep *beep) -{ - struct input_dev *input_dev; - struct hda_codec *codec = beep->codec; - - input_dev = input_allocate_device(); - if (!input_dev) - return -ENOMEM; - - /* setup digital beep device */ - input_dev->name = "HDA Digital PCBeep"; - input_dev->phys = beep->phys; - input_dev->id.bustype = BUS_PCI; - input_dev->dev.parent = &codec->card->card_dev; - - input_dev->id.vendor = codec->core.vendor_id >> 16; - input_dev->id.product = codec->core.vendor_id & 0xffff; - input_dev->id.version = 0x01; - - input_dev->evbit[0] = BIT_MASK(EV_SND); - input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); - input_dev->event = snd_hda_beep_event; - input_set_drvdata(input_dev, beep); - - beep->dev = input_dev; - return 0; -} - /** * snd_hda_enable_beep_device - Turn on/off beep sound * @codec: the HDA codec @@ -186,6 +148,38 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) } EXPORT_SYMBOL_GPL(snd_hda_enable_beep_device); +static int beep_dev_register(struct snd_device *device) +{ + struct hda_beep *beep = device->device_data; + int err; + + err = input_register_device(beep->dev); + if (!err) + beep->registered = true; + return err; +} + +static int beep_dev_disconnect(struct snd_device *device) +{ + struct hda_beep *beep = device->device_data; + + if (beep->registered) + input_unregister_device(beep->dev); + else + input_free_device(beep->dev); + turn_off_beep(beep); + return 0; +} + +static int beep_dev_free(struct snd_device *device) +{ + struct hda_beep *beep = device->device_data; + + beep->codec->beep = NULL; + kfree(beep); + return 0; +} + /** * snd_hda_attach_beep_device - Attach a beep input device * @codec: the HDA codec @@ -194,14 +188,16 @@ EXPORT_SYMBOL_GPL(snd_hda_enable_beep_device); * Attach a beep object to the given widget. If beep hint is turned off * explicitly or beep_mode of the codec is turned off, this doesn't nothing. * - * The attached beep device has to be registered via - * snd_hda_register_beep_device() and released via snd_hda_detach_beep_device() - * appropriately. - * * Currently, only one beep device is allowed to each codec. */ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { + static struct snd_device_ops ops = { + .dev_register = beep_dev_register, + .dev_disconnect = beep_dev_disconnect, + .dev_free = beep_dev_free, + }; + struct input_dev *input_dev; struct hda_beep *beep; int err; @@ -226,14 +222,41 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); mutex_init(&beep->mutex); - err = snd_hda_do_attach(beep); - if (err < 0) { - kfree(beep); - codec->beep = NULL; - return err; + input_dev = input_allocate_device(); + if (!input_dev) { + err = -ENOMEM; + goto err_free; } + /* setup digital beep device */ + input_dev->name = "HDA Digital PCBeep"; + input_dev->phys = beep->phys; + input_dev->id.bustype = BUS_PCI; + input_dev->dev.parent = &codec->card->card_dev; + + input_dev->id.vendor = codec->core.vendor_id >> 16; + input_dev->id.product = codec->core.vendor_id & 0xffff; + input_dev->id.version = 0x01; + + input_dev->evbit[0] = BIT_MASK(EV_SND); + input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); + input_dev->event = snd_hda_beep_event; + input_set_drvdata(input_dev, beep); + + beep->dev = input_dev; + + err = snd_device_new(codec->card, SNDRV_DEV_JACK, beep, &ops); + if (err < 0) + goto err_input; + return 0; + + err_input: + input_free_device(beep->dev); + err_free: + kfree(beep); + codec->beep = NULL; + return err; } EXPORT_SYMBOL_GPL(snd_hda_attach_beep_device); @@ -243,41 +266,11 @@ EXPORT_SYMBOL_GPL(snd_hda_attach_beep_device); */ void snd_hda_detach_beep_device(struct hda_codec *codec) { - struct hda_beep *beep = codec->beep; - if (beep) { - if (beep->dev) - snd_hda_do_detach(beep); - codec->beep = NULL; - kfree(beep); - } + if (!codec->bus->shutdown && codec->beep) + snd_device_free(codec->card, codec->beep); } EXPORT_SYMBOL_GPL(snd_hda_detach_beep_device); -/** - * snd_hda_register_beep_device - Register the beep device - * @codec: the HDA codec - */ -int snd_hda_register_beep_device(struct hda_codec *codec) -{ - struct hda_beep *beep = codec->beep; - int err; - - if (!beep || !beep->dev) - return 0; - - err = input_register_device(beep->dev); - if (err < 0) { - codec_err(codec, "hda_beep: unable to register input device\n"); - input_free_device(beep->dev); - codec->beep = NULL; - kfree(beep); - return err; - } - beep->registered = true; - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_register_beep_device); - static bool ctl_has_mute(struct snd_kcontrol *kcontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index f1457c6b3969..a25358a4807a 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -34,7 +34,6 @@ struct hda_beep { int snd_hda_enable_beep_device(struct hda_codec *codec, int enable); int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); -int snd_hda_register_beep_device(struct hda_codec *codec); #else static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { @@ -43,9 +42,5 @@ static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) static inline void snd_hda_detach_beep_device(struct hda_codec *codec) { } -static inline int snd_hda_register_beep_device(struct hda_codec *codec) -{ - return 0; -} #endif #endif diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index dc7b342f00ef..5f2005098a60 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -813,7 +813,6 @@ void snd_hda_codec_register(struct hda_codec *codec) if (codec->registered) return; if (device_is_registered(hda_codec_dev(codec))) { - snd_hda_register_beep_device(codec); codec_display_power(codec, true); pm_runtime_enable(hda_codec_dev(codec)); /* it was powered up in snd_hda_codec_new(), now all done */ @@ -828,14 +827,6 @@ static int snd_hda_codec_dev_register(struct snd_device *device) return 0; } -static int snd_hda_codec_dev_disconnect(struct snd_device *device) -{ - struct hda_codec *codec = device->device_data; - - snd_hda_detach_beep_device(codec); - return 0; -} - static int snd_hda_codec_dev_free(struct snd_device *device) { struct hda_codec *codec = device->device_data; @@ -921,7 +912,6 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, int err; static struct snd_device_ops dev_ops = { .dev_register = snd_hda_codec_dev_register, - .dev_disconnect = snd_hda_codec_dev_disconnect, .dev_free = snd_hda_codec_dev_free, }; -- 2.16.4

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