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

[alsa-devel] Applied "ASoC: tlv320aic32x4: Allow 192000 Sample Rate" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: tlv320aic32x4: Allow 192000 Sample Rate 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 6d56ee1550b8a81bc63c80051ff78d8d704b09ba Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:54 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Allow 192000 Sample Rate The clocking and processing blocks are now properly set up to support 192000 sample rates. Allow drivers to ask for that. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index a5457d3de1b1..75443efeda69 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -857,7 +857,7 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, return 0; } -#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_192000 #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -- 2.20.1

1 0

[alsa-devel] [PATCH] ALSA: aloop: Support S24 sample formats
by twischer＠de.adit-jv.com 25 Mar '19

25 Mar '19

From: Timo Wischer <twischer(a)de.adit-jv.com> Currently snd_aloop supports only S16 and S32 audio sample formats. With this patch the S24 formats are also supported. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/drivers/aloop.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 8c3fbe1..c14e57b 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -337,7 +337,7 @@ static int loopback_prepare(struct snd_pcm_substream *substream) loopback_timer_stop_sync(dpcm); - salign = (snd_pcm_format_width(runtime->format) * + salign = (snd_pcm_format_physical_width(runtime->format) * runtime->channels) / 8; bps = salign * runtime->rate; if (bps <= 0 || salign <= 0) @@ -562,6 +562,8 @@ static const struct snd_pcm_hardware loopback_pcm_hardware = SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE), .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000, -- 2.7.4

3 2

[alsa-devel] [PROBLEM] ALSA: hda: PCM streams are suspended while opening
by Jon Hunter 25 Mar '19

25 Mar '19

From: Jonathan Hunter <jonathanh(a)nvidia.com> This issue is not observed on the latest mainline kernel since the 'ALSA: PCM suspend cleanup' series has been applied and the snd_pcm_suspend_all() function call in the HDA codec driver has been removed. However, this issue impacts stable kernel branches and so I am trying to find a solution for these branches. When stress testing audio playback across multiple HDA streams simultaneously, the following failure is sometimes observed when starting playback ... Unable to set hw params for playback: File descriptor in bad state The problem is caused because ... 1. Playback is starting for one HDA codec and so the chip->open_mutex in azx_pcm_open() has been acquired. 2. Playback for another HDA codec is starting but is blocked waiting to acquire the chip->open_mutex in azx_pcm_open(). 3. For the HDA codec that is blocked, its runtime-pm status is still enabled from a previous playback session that has since completed and been closed, however its autosuspend timeout has not expired yet. 4. For the HDA codec that is blocked, the runtime-pm autosuspend timeout now occurs calling the runtime-pm suspend callback and this calls snd_pcm_suspend_all() placing all PCM streams in the suspended state. 5. The block HDA codec is now unblocked and fails to set the HW params because the PCM stream is in the suspended state. The above has been observed on Tegra platforms where the autosuspend delay is set to 1 second and there is a delay to an AZX command. This highlights that there is a window of time where autosuspend can place the HDA PCM stream in the suspended state while opening the stream and cause playback to fail. Looking at commit 3d21ef0b49f8 ('ALSA: pcm: Suspend streams globally via device type PM ops') it appears that the call to snd_pcm_suspend_all() has now been moved to the to the suspend handler for PCM streams and so I am wondering if it would be equivalent to make the following change to the HDA codec driver to achieve the same behaviour but only for the HDA driver. So far the issue has not been observed with this change. Please note that I am not sending this as a formal patch as I wanted to get some feedback/comments on the above. --- sound/pci/hda/hda_codec.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9f8d59e7e89f..26ecc6e1f524 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2923,12 +2923,9 @@ static void hda_call_codec_resume(struct hda_codec *codec) static int hda_codec_runtime_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - struct hda_pcm *pcm; unsigned int state; cancel_delayed_work_sync(&codec->jackpoll_work); - list_for_each_entry(pcm, &codec->pcm_list_head, list) - snd_pcm_suspend_all(pcm->pcm); state = hda_call_codec_suspend(codec); if (codec->link_down_at_suspend || (codec_has_clkstop(codec) && codec_has_epss(codec) && @@ -2948,11 +2945,22 @@ static int hda_codec_runtime_resume(struct device *dev) pm_runtime_mark_last_busy(dev); return 0; } + +static int hda_codec_suspend(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + struct hda_pcm *pcm; + + list_for_each_entry(pcm, &codec->pcm_list_head, list) + snd_pcm_suspend_all(pcm->pcm); + + return pm_runtime_force_suspend(dev); +} #endif /* CONFIG_PM */ /* referred in hda_bind.c */ const struct dev_pm_ops hda_codec_driver_pm = { - SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + SET_SYSTEM_SLEEP_PM_OPS(hda_codec_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume, NULL) -- 1.9.1

2 8

[alsa-devel] [PATCH] ASoC: nau8810: use 64-bit arithmetic instead of 32-bit
by John Hsu 25 Mar '19

25 Mar '19

Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index bfd74b86c9d2..e7fd0b2645ea 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -505,7 +505,7 @@ static int nau88l0_calc_pll(unsigned int pll_in, f2_max = 0; scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { - f2 = 256 * fs * 4 * nau8810_mclk_scaler[i] / 10; + f2 = 256ULL * fs * 4 * nau8810_mclk_scaler[i] / 10; if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && f2_max < f2) { f2_max = f2; -- 2.18.0

2 2

[alsa-devel] Applied "ASoC: AMD: Fix incorrect extern" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: AMD: Fix incorrect extern 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 eddb6084305e5418ce28b8379a6c3d0058a61463 Mon Sep 17 00:00:00 2001 From: Andi Kleen <ak(a)linux.intel.com> Date: Thu, 21 Mar 2019 15:00:05 -0700 Subject: [PATCH] ASoC: AMD: Fix incorrect extern When using bare externs outside include files that types should at least match. This fixes a type confusion between bool and int. Cc: broonie(a)kernel.org Signed-off-by: Andi Kleen <ak(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f37a588ba345..8a619a75b3a9 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -47,7 +47,7 @@ static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_clk; -extern int bt_uart_enable; +extern bool bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: stm32: sai: add power management
by Olivier Moysan 25 Mar '19

25 Mar '19

Add support of low power modes to STM32 SAI driver. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai.c | 80 ++++++++++++++++++++++++++++++++++++++----- sound/soc/stm/stm32_sai.h | 2 ++ sound/soc/stm/stm32_sai_sub.c | 45 ++++++++++++++++++------ 3 files changed, 108 insertions(+), 19 deletions(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index d68d62f12df5..7550d5f08be3 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -21,6 +21,7 @@ #include <linux/delay.h> #include <linux/module.h> #include <linux/of_platform.h> +#include <linux/pinctrl/consumer.h> #include <linux/reset.h> #include <sound/dmaengine_pcm.h> @@ -44,20 +45,41 @@ static const struct of_device_id stm32_sai_ids[] = { {} }; -static int stm32_sai_sync_conf_client(struct stm32_sai_data *sai, int synci) +static int stm32_sai_pclk_disable(struct device *dev) +{ + struct stm32_sai_data *sai = dev_get_drvdata(dev); + + clk_disable_unprepare(sai->pclk); + + return 0; +} + +static int stm32_sai_pclk_enable(struct device *dev) { + struct stm32_sai_data *sai = dev_get_drvdata(dev); int ret; - /* Enable peripheral clock to allow GCR register access */ ret = clk_prepare_enable(sai->pclk); if (ret) { dev_err(&sai->pdev->dev, "failed to enable clock: %d\n", ret); return ret; } + return 0; +} + +static int stm32_sai_sync_conf_client(struct stm32_sai_data *sai, int synci) +{ + int ret; + + /* Enable peripheral clock to allow GCR register access */ + ret = stm32_sai_pclk_enable(&sai->pdev->dev); + if (ret) + return ret; + writel_relaxed(FIELD_PREP(SAI_GCR_SYNCIN_MASK, (synci - 1)), sai->base); - clk_disable_unprepare(sai->pclk); + stm32_sai_pclk_disable(&sai->pdev->dev); return 0; } @@ -68,11 +90,9 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) int ret; /* Enable peripheral clock to allow GCR register access */ - ret = clk_prepare_enable(sai->pclk); - if (ret) { - dev_err(&sai->pdev->dev, "failed to enable clock: %d\n", ret); + ret = stm32_sai_pclk_enable(&sai->pdev->dev); + if (ret) return ret; - } dev_dbg(&sai->pdev->dev, "Set %pOFn%s as synchro provider\n", sai->pdev->dev.of_node, @@ -83,13 +103,13 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); - clk_disable_unprepare(sai->pclk); + stm32_sai_pclk_disable(&sai->pdev->dev); return -EINVAL; } writel_relaxed(FIELD_PREP(SAI_GCR_SYNCOUT_MASK, synco), sai->base); - clk_disable_unprepare(sai->pclk); + stm32_sai_pclk_disable(&sai->pdev->dev); return 0; } @@ -195,12 +215,54 @@ static int stm32_sai_probe(struct platform_device *pdev) return devm_of_platform_populate(&pdev->dev); } +#ifdef CONFIG_PM_SLEEP +/* + * When pins are shared by two sai sub instances, pins have to be defined + * in sai parent node. In this case, pins state is not managed by alsa fw. + * These pins are managed in suspend/resume callbacks. + */ +static int stm32_sai_suspend(struct device *dev) +{ + struct stm32_sai_data *sai = dev_get_drvdata(dev); + int ret; + + ret = stm32_sai_pclk_enable(dev); + if (ret) + return ret; + + sai->gcr = readl_relaxed(sai->base); + stm32_sai_pclk_disable(dev); + + return pinctrl_pm_select_sleep_state(dev); +} + +static int stm32_sai_resume(struct device *dev) +{ + struct stm32_sai_data *sai = dev_get_drvdata(dev); + int ret; + + ret = stm32_sai_pclk_enable(dev); + if (ret) + return ret; + + writel_relaxed(sai->gcr, sai->base); + stm32_sai_pclk_disable(dev); + + return pinctrl_pm_select_default_state(dev); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops stm32_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(stm32_sai_suspend, stm32_sai_resume) +}; + MODULE_DEVICE_TABLE(of, stm32_sai_ids); static struct platform_driver stm32_sai_driver = { .driver = { .name = "st,stm32-sai", .of_match_table = stm32_sai_ids, + .pm = &stm32_sai_pm_ops, }, .probe = stm32_sai_probe, }; diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h index 08de899c766b..9c36a393ab7b 100644 --- a/sound/soc/stm/stm32_sai.h +++ b/sound/soc/stm/stm32_sai.h @@ -268,6 +268,7 @@ struct stm32_sai_conf { * @version: SOC version * @irq: SAI interrupt line * @set_sync: pointer to synchro mode configuration callback + * @gcr: SAI Global Configuration Register */ struct stm32_sai_data { struct platform_device *pdev; @@ -279,4 +280,5 @@ struct stm32_sai_data { int irq; int (*set_sync)(struct stm32_sai_data *sai, struct device_node *np_provider, int synco, int synci); + u32 gcr; }; diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 55d802f51c15..2a90641cf164 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -168,6 +168,7 @@ static bool stm32_sai_sub_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case STM_SAI_DR_REGX: + case STM_SAI_SR_REGX: return true; default: return false; @@ -182,7 +183,6 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg) case STM_SAI_FRCR_REGX: case STM_SAI_SLOTR_REGX: case STM_SAI_IMR_REGX: - case STM_SAI_SR_REGX: case STM_SAI_CLRFR_REGX: case STM_SAI_DR_REGX: case STM_SAI_PDMCR_REGX: @@ -202,6 +202,7 @@ static const struct regmap_config stm32_sai_sub_regmap_config_f4 = { .volatile_reg = stm32_sai_sub_volatile_reg, .writeable_reg = stm32_sai_sub_writeable_reg, .fast_io = true, + .cache_type = REGCACHE_FLAT, }; static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { @@ -213,6 +214,7 @@ static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { .volatile_reg = stm32_sai_sub_volatile_reg, .writeable_reg = stm32_sai_sub_writeable_reg, .fast_io = true, + .cache_type = REGCACHE_FLAT, }; static int snd_pcm_iec958_info(struct snd_kcontrol *kcontrol, @@ -441,8 +443,8 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid) if (!flags) return IRQ_NONE; - regmap_update_bits(sai->regmap, STM_SAI_CLRFR_REGX, SAI_XCLRFR_MASK, - SAI_XCLRFR_MASK); + regmap_write_bits(sai->regmap, STM_SAI_CLRFR_REGX, SAI_XCLRFR_MASK, + SAI_XCLRFR_MASK); if (!sai->substream) { dev_err(&pdev->dev, "Device stopped. Spurious IRQ 0x%x\n", sr); @@ -704,9 +706,8 @@ static int stm32_sai_startup(struct snd_pcm_substream *substream, } /* Enable ITs */ - - regmap_update_bits(sai->regmap, STM_SAI_CLRFR_REGX, - SAI_XCLRFR_MASK, SAI_XCLRFR_MASK); + regmap_write_bits(sai->regmap, STM_SAI_CLRFR_REGX, + SAI_XCLRFR_MASK, SAI_XCLRFR_MASK); imr = SAI_XIMR_OVRUDRIE; if (STM_SAI_IS_CAPTURE(sai)) { @@ -738,10 +739,10 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, * SAI fifo threshold is set to half fifo, to keep enough space * for DMA incoming bursts. */ - regmap_update_bits(sai->regmap, STM_SAI_CR2_REGX, - SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, - SAI_XCR2_FFLUSH | - SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); + regmap_write_bits(sai->regmap, STM_SAI_CR2_REGX, + SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, + SAI_XCR2_FFLUSH | + SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); /* DS bits in CR1 not set for SPDIF (size forced to 24 bits).*/ if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { @@ -1492,10 +1493,34 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int stm32_sai_sub_suspend(struct device *dev) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + return 0; +} + +static int stm32_sai_sub_resume(struct device *dev) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops stm32_sai_sub_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(stm32_sai_sub_suspend, stm32_sai_sub_resume) +}; + static struct platform_driver stm32_sai_sub_driver = { .driver = { .name = "st,stm32-sai-sub", .of_match_table = stm32_sai_sub_ids, + .pm = &stm32_sai_sub_pm_ops, }, .probe = stm32_sai_sub_probe, }; -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: sirf-audio: Remove redundant of_match_node call" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: sirf-audio: Remove redundant of_match_node call 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 78540a259b050480d20354efaf1055c233728b3b Mon Sep 17 00:00:00 2001 From: Aditya Pakki <pakki001(a)umn.edu> Date: Fri, 22 Mar 2019 17:00:29 -0500 Subject: [PATCH] ASoC: sirf-audio: Remove redundant of_match_node call Unlike other drivers probe method, of_match_node return value is not used or checked. This patch removes the redundant code. Signed-off-by: Aditya Pakki <pakki001(a)umn.edu> Reviewed-by: Steven Price <steven.price(a)arm.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/sirf-audio-codec.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index e424499a8450..e0af21050078 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -461,9 +461,6 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) struct sirf_audio_codec *sirf_audio_codec; void __iomem *base; struct resource *mem_res; - const struct of_device_id *match; - - match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node); sirf_audio_codec = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_codec), GFP_KERNEL); -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: dapm: Fix NULL pointer dereference in snd_soc_dapm_free_kcontrol
by Pankaj Bharadiya 25 Mar '19

25 Mar '19

w_text_param can be NULL and it is being dereferenced without checking. Add the missing sanity check to prevent NULL pointer dereference. Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya(a)intel.com> --- sound/soc/soc-dapm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ec06ef..67b032c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3957,6 +3957,10 @@ snd_soc_dapm_free_kcontrol(struct snd_soc_card *card, int count; devm_kfree(card->dev, (void *)*private_value); + + if (!w_param_text) + return; + for (count = 0 ; count < num_params; count++) devm_kfree(card->dev, (void *)w_param_text[count]); devm_kfree(card->dev, w_param_text); -- 2.7.4

3 2

[alsa-devel] [v2] ASoC: AMD: Configure wclk and bclk of master codec
by Agrawal, Akshu 25 Mar '19

25 Mar '19

With CCF support in da7219, we can now set the correct rate of wclk and bclk. Setting bclk at lower rate at 1.53Mhz from its earlier default rate of 3Mhz, also fixes noise issue observed on some dmics. v2: Removed clk set rate and enable/disable from da7219 ops as da7219 codec takes care of them internally. Clock configuration kept for those codecs where da7219 acts as master of clock. As suggested by Adam Thomson. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 45 +++++++++++++++++----------- 1 file changed, 28 insertions(+), 17 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 8e2ce369a295..7737c6dc026a 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -46,7 +46,8 @@ #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -static struct clk *da7219_dai_clk; +static struct clk *da7219_dai_wclk; +static struct clk *da7219_dai_bclk; extern int bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) @@ -72,7 +73,8 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - da7219_dai_clk = clk_get(component->dev, "da7219-dai-bclk"); + da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); + da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | @@ -94,12 +96,15 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int da7219_clk_enable(struct snd_pcm_substream *substream) +static int da7219_clk_enable(struct snd_pcm_substream *substream, + int wclk_rate, int bclk_rate) { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - ret = clk_prepare_enable(da7219_dai_clk); + clk_set_rate(da7219_dai_wclk, wclk_rate); + clk_set_rate(da7219_dai_bclk, bclk_rate); + ret = clk_prepare_enable(da7219_dai_bclk); if (ret < 0) { dev_err(rtd->dev, "can't enable master clock %d\n", ret); return ret; @@ -110,7 +115,7 @@ static int da7219_clk_enable(struct snd_pcm_substream *substream) static void da7219_clk_disable(void) { - clk_disable_unprepare(da7219_dai_clk); + clk_disable_unprepare(da7219_dai_bclk); } static const unsigned int channels[] = { @@ -151,7 +156,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_SP_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) @@ -173,12 +178,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; - return da7219_clk_enable(substream); -} - -static void cz_da7219_shutdown(struct snd_pcm_substream *substream) -{ - da7219_clk_disable(); + return 0; } static int cz_max_startup(struct snd_pcm_substream *substream) @@ -199,7 +199,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static void cz_max_shutdown(struct snd_pcm_substream *substream) @@ -225,7 +225,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->cap_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_dmic1_startup(struct snd_pcm_substream *substream) @@ -247,7 +247,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; - return da7219_clk_enable(substream); + return 0; } static void cz_dmic_shutdown(struct snd_pcm_substream *substream) @@ -255,29 +255,40 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int wclk, bclk; + + wclk = params_rate(params); + bclk = wclk * params_channels(params) * + snd_pcm_format_width(params_format(params)); + return da7219_clk_enable(substream, wclk, bclk); +} static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, - .shutdown = cz_da7219_shutdown, }; static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_cap_startup, - .shutdown = cz_da7219_shutdown, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_max_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_dmic_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, + .hw_params = cz_da7219_hw_params, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { -- 2.17.1

2 2

[alsa-devel] [PATCH v3 00/11] ASoC: tlv320aic32x4: Rework Clock Setting
by Annaliese McDermond 25 Mar '19

25 Mar '19

The current tlv320aic32x4 code sets the various clock parameters by looking in a static table in the kernel. This works well enough, but it has a few disadvantages. 1) If the master clock doesn't match one of the precalculated values the driver cannot work. This could be because the designer decided to use a different crystal, or because the clock comes from a PLL that can't quite achieve the proper frequency exactly. 2) The driver only supports certain pre-calculated sample rates. The actual hardware can support many more. These changes enable those rates. Additionally, certain sample rates at certain clock rates were previously unavailable as combinations. This patch dynamically calculates the various PLL and divider values to find something that works. Additionally, to enable much of this, the clock tree is modeled as a part of the Common Clock Framework. This allows for easier debugigng as you can use the standard tools such as /sys/kernel/debug/clk/clk_summary. It also simplifies enabling the entirety of the clock tree needed, and makes sure unused clocks become disabled. These patches have been tested and are working on a Raspberry Pi with the simplecard driver. Changes in v2: - Update 0002 patch to change Kconfig to add dependency on CCF - Update 0002 patch to correct the SPDX identifier - Update 0002 patch to use EXPORT_SYMBOL_GPL() instead of EXPORT_SYMBOL() - Update patches 0002, 0004 and 0005 to use prepare/unprepare instead of enable/disable since the i2c operations are not atomic - Update 0004 patch to fix indentation in commit message - Drop patch 0009 because it is a duplicate of the already committed patch in 667e9334 - Update 0019 patch to keep aic32x4_set_dai_sysclk but to have it use clk_set_rate to set the master clock rate Changes in v3: - Rebase on current for-next - Move "Properly Set Processing Blocks" to the start of the patches so that it can potentially be applied as a bug fix Annaliese McDermond (11): ASoC: tlv320aic32x4: Properly Set Processing Blocks ASoC: tlv320aic32x4: Model PLL in CCF ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF ASoC: tlv320aic32x4: Model DAC/ADC dividers in CCF ASoC: tlv320aic32x4: Model BDIV divider in CCF ASoC: tlv320aic32x4: Control clock gating with CCF ASoC: tlv320aic32x4: Move aosr and dosr setting to separate functions ASoC: tlv320aic32x4: Dynamically Determine Clocking ASoC: tlv320aic32x4: Restructure set_dai_sysclk ASoC: tlv320aic32x4: Remove mclk references ASoC: tlv320aic32x4: Allow 192000 Sample Rate sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/tlv320aic32x4-clk.c | 483 +++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 357 ++++++++++---------- sound/soc/codecs/tlv320aic32x4.h | 11 + 5 files changed, 665 insertions(+), 189 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic32x4-clk.c -- 2.19.1

2 15