February 2019 - Alsa-devel - mailman.alsa-project.org

[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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