September 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: dapm: Add missing return value check for snd_soc_dapm_new_dai
by Charles Keepax 10 Sep '18

10 Sep '18

snd_soc_dapm_new_dai may return an error pointer and currently this isn't checked for in dapm_connect_dai_link_widgets. Add code to check the return value and not add routes in that case. Fixes: 778ff5bb8689 ("ASoC: dapm: Move connection of CODEC to CODEC DAIs") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0dcdcc23dcfde..43983c69f6aad 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4087,6 +4087,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, playback = snd_soc_dapm_new_dai(card, rtd, playback_cpu, codec); + if (IS_ERR(playback)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(playback)); + continue; + } snd_soc_dapm_add_path(&card->dapm, playback_cpu, playback, NULL, NULL); @@ -4099,7 +4106,9 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, snd_soc_dapm_add_path(&card->dapm, playback, codec, NULL, NULL); } + } + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* connect BE DAI capture if widgets are valid */ codec = codec_dai->capture_widget; @@ -4108,6 +4117,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, capture = snd_soc_dapm_new_dai(card, rtd, codec, capture_cpu); + if (IS_ERR(capture)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(capture)); + continue; + } snd_soc_dapm_add_path(&card->dapm, capture, capture_cpu, NULL, NULL); -- 2.11.0

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[alsa-devel] Applied "ASoC: AMD: Fix simultaneous playback and capture on different channel" to the asoc tree
by Mark Brown 10 Sep '18

10 Sep '18

The patch ASoC: AMD: Fix simultaneous playback and capture on different channel 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 8dcb0c90c691de5b79608d04ec7941ef9b3fee9c Mon Sep 17 00:00:00 2001 From: Akshu Agrawal <akshu.agrawal(a)amd.com> Date: Mon, 10 Sep 2018 22:50:27 +0530 Subject: [PATCH] ASoC: AMD: Fix simultaneous playback and capture on different channel If capture and playback are started on different channel (I2S/BT) there is a possibilty that channel information passed from machine driver is overwritten before the configuration is done in dma driver. Example: 113.597588: cz_max_startup: ---playback sets BT channel 113.597694: cz_dmic1_startup: ---capture sets I2S channel 113.597979: acp_dma_hw_params: ---configures capture for I2S channel 113.598114: acp_dma_hw_params: ---configures playback for I2S channel This is fixed by having 2 separate instance for playback and capture. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp-da7219-max98357a.c | 40 +++++++++++++++++++++++----- sound/soc/amd/acp-pcm-dma.c | 8 ++++-- sound/soc/amd/acp.h | 3 ++- 3 files changed, 41 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3879cccbd2c0..717a017f0db6 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -133,7 +133,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; -static int cz_da7219_startup(struct snd_pcm_substream *substream) +static int cz_da7219_play_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->play_i2s_instance = I2S_SP_INSTANCE; + return da7219_clk_enable(substream); +} + +static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -177,7 +198,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->play_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -203,7 +224,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->cap_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -224,7 +245,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); } @@ -234,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +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_startup, + .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, }; @@ -266,7 +292,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - .ops = &cz_da7219_cap_ops, + .ops = &cz_da7219_play_ops, }, { .name = "amd-da7219-cap", diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index e359938e3d7e..8f3bc6e37f26 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -846,8 +846,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (pinfo) { - rtd->i2s_instance = pinfo->i2s_instance; - rtd->capture_channel = pinfo->capture_channel; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rtd->i2s_instance = pinfo->play_i2s_instance; + } else { + rtd->i2s_instance = pinfo->cap_i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index be3963e8f4fa..dbbb1a85638d 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -158,7 +158,8 @@ struct audio_drv_data { * and dma driver */ struct acp_platform_info { - u16 i2s_instance; + u16 play_i2s_instance; + u16 cap_i2s_instance; u16 capture_channel; }; -- 2.19.0.rc1

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[alsa-devel] [PATCH v9] ASoC: pxa: switch to new ac97 bus support
by Robert Jarzmik 10 Sep '18

10 Sep '18

Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v8: fixed the trivial compilation error v7 was in https://patchwork.kernel.org/patch/9951919/ --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++--- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++++++---------------------- 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 5fbd47a9177e..28867732a318 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -31,7 +31,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 776e148b0aa2..29f577e6dfc0 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -19,14 +19,13 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S select SND_PXA2XX_LIB diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9f779657bc86..f8a3aa6c6d4e 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -233,6 +226,9 @@ MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -245,10 +241,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -259,8 +259,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- 2.11.0

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[alsa-devel] [PATCH 1/2] ASoC: AMD: Fix simultaneous playback and capture on different channel
by Akshu Agrawal 10 Sep '18

10 Sep '18

If capture and playback are started on different channel (I2S/BT) there is a possibilty that channel information passed from machine driver is overwritten before the configuration is done in dma driver. Example: 113.597588: cz_max_startup: ---playback sets BT channel 113.597694: cz_dmic1_startup: ---capture sets I2S channel 113.597979: acp_dma_hw_params: ---configures capture for I2S channel 113.598114: acp_dma_hw_params: ---configures playback for I2S channel This is fixed by having lock between startup and prepare. This ensures no other codec startup gets called between a codec's startup(where channel info is set) and hw_params(where channel info is read). Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3879ccc..b98ffbc 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -47,6 +47,7 @@ static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_clk; +static struct mutex instance_lock; extern int bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) @@ -150,6 +151,7 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); @@ -160,6 +162,12 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_da7219_prepare(struct snd_pcm_substream *substream) +{ + mutex_unlock(&instance_lock); + return 0; +} + static int cz_max_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -177,6 +185,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -186,6 +195,12 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_max_prepare(struct snd_pcm_substream *substream) +{ + mutex_unlock(&instance_lock); + return 0; +} + static int cz_dmic0_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -203,6 +218,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -224,6 +240,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); @@ -234,24 +251,34 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_dmic_prepare(struct snd_pcm_substream *substream) +{ + mutex_unlock(&instance_lock); + return 0; +} + static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_startup, .shutdown = cz_da7219_shutdown, + .prepare = cz_da7219_prepare, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_max_shutdown, + .prepare = cz_max_prepare, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_dmic_shutdown, + .prepare = cz_dmic_prepare, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, + .prepare = cz_dmic_prepare, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { @@ -409,6 +436,7 @@ static int cz_probe(struct platform_device *pdev) card = &cz_card; cz_card.dev = &pdev->dev; platform_set_drvdata(pdev, card); + mutex_init(&instance_lock); snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); if (ret) { -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: AMD: Fix simultaneous playback and capture on different channel
by Akshu Agrawal 10 Sep '18

10 Sep '18

If capture and playback are started on different channel (I2S/BT) there is a possibilty that channel information passed from machine driver is overwritten before the configuration is done in dma driver. Example: 113.597588: cz_max_startup: ---playback sets BT channel 113.597694: cz_dmic1_startup: ---capture sets I2S channel 113.597979: acp_dma_hw_params: ---configures capture for I2S channel 113.598114: acp_dma_hw_params: ---configures playback for I2S channel This is fixed by having 2 separate instance for playback and capture. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 40 +++++++++++++++++++++++++++++------- sound/soc/amd/acp-pcm-dma.c | 8 ++++++-- sound/soc/amd/acp.h | 3 ++- 3 files changed, 41 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 55d7f61..0d97d00 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -133,7 +133,7 @@ static void da7219_clk_disable(void) .mask = 0, }; -static int cz_da7219_startup(struct snd_pcm_substream *substream) +static int cz_da7219_play_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->play_i2s_instance = I2S_SP_INSTANCE; + return da7219_clk_enable(substream); +} + +static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -177,7 +198,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->play_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -203,7 +224,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->cap_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -224,7 +245,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); } @@ -234,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +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_startup, + .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, }; @@ -266,7 +292,7 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - .ops = &cz_da7219_cap_ops, + .ops = &cz_da7219_play_ops, }, { .name = "amd-da7219-cap", diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index b43f0a1..b0e245c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -862,8 +862,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (pinfo) { - rtd->i2s_instance = pinfo->i2s_instance; - rtd->capture_channel = pinfo->capture_channel; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rtd->i2s_instance = pinfo->play_i2s_instance; + } else { + rtd->i2s_instance = pinfo->cap_i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index be3963e..dbbb1a8 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -158,7 +158,8 @@ struct audio_drv_data { * and dma driver */ struct acp_platform_info { - u16 i2s_instance; + u16 play_i2s_instance; + u16 cap_i2s_instance; u16 capture_channel; }; -- 1.9.1

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[alsa-devel] [PATCH] ALSA: emu10k1: fix possible info leak to userspace on SNDRV_EMU10K1_IOCTL_INFO
by Takashi Iwai 10 Sep '18

10 Sep '18

From: Willy Tarreau <w(a)1wt.eu> snd_emu10k1_fx8010_ioctl(SNDRV_EMU10K1_IOCTL_INFO) allocates memory using kmalloc() and partially fills it by calling snd_emu10k1_fx8010_info() before returning the resulting structure to userspace, leaving uninitialized holes. Let's just use kzalloc() here. BugLink: http://blog.infosectcbr.com.au/2018/09/linux-kernel-infoleaks.html Signed-off-by: Willy Tarreau <w(a)1wt.eu> Cc: Jann Horn <jannh(a)google.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/emu10k1/emufx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 90713741c2dc..6ebe817801ea 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -2540,7 +2540,7 @@ static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, un emu->support_tlv = 1; return put_user(SNDRV_EMU10K1_VERSION, (int __user *)argp); case SNDRV_EMU10K1_IOCTL_INFO: - info = kmalloc(sizeof(*info), GFP_KERNEL); + info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; snd_emu10k1_fx8010_info(emu, info); -- 2.18.0

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[alsa-devel] RaspberryPi vc4-hdmi softvol for alsamixer
by Andreas Müller 10 Sep '18

10 Sep '18

Hi, This is my first post here and guess what I am looking for support :) Up to now I used audioinjector and hifiberry on RaspberryPi and that worked without issues (from alsa point of view). Now I would like to select vc4hdmi on alsamixer but that complains for missing controls. I think vc4hdmi does not support hard volume setting - like some other devices. So I read here and there and found softvol. To test I modified vc4-hdmi.conf shipped with alsal-iib from: # default with dmix vc4-hdmi.pcm.default { @args [ CARD ] @args.CARD { type string } type asym playback.pcm { type plug slave.pcm { @func concat strings [ "dmix:" $CARD ] } } } to (stolen from PC-Speaker.conf) # default with dmix vc4-hdmi.pcm.default { @args [ CARD ] @args.CARD { type string } type asym playback.pcm { type plug slave.pcm { type softvol control { name "Master Playback Volume" card $CARD } slave.pcm { @func concat strings [ "dmix:" $CARD ] } } } } after reboot nothing changes! I build my images with Openembedded/Yocto and from my experience it is possible that there is some misconfiguration / path issue causes problems. I know the card works with the help of pulseaudio. So now my questions: * Would you expect that my modification of vc4-hdmi.conf adds softvol control to vc4hdmi? * If yes: The only information I have currently: It does not work as expected. So where can I ask alsa for some feedback what went wrong (log-files/dmesg/journal...)? Thanks in advance Andreas

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[alsa-devel] [PATCH] ASoC: dapm: Fix a couple uninitialized ret variables
by Dan Carpenter 10 Sep '18

10 Sep '18

Smatch complains that these variables could be uninitialized. The first one in snd_soc_dai_link_event() is probably a false positive, because probably we know the lists are not empty. I would normally ignore the warning, but GCC complains here as well so I just silenced the warning. The "ret" in snd_soc_dapm_new_dai() does need to be initialized or it leads to a bogus dereference in the caller. Fixes: 3bbf5d34fd4a ("ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9f4edcd19e02..0dcdcc23dcfd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3623,7 +3623,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; unsigned int fmt; - int ret; + int ret = 0; config = rtd->dai_link->params + rtd->params_select; @@ -3938,8 +3938,10 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); - if (IS_ERR(w)) + if (IS_ERR(w)) { + ret = PTR_ERR(w); goto outfree_kcontrol_news; + } w->priv = rtd;

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[alsa-devel] Applied "ASoC: q6core: Use kmemdup to replace kzalloc + memcpy" to the asoc tree
by Mark Brown 10 Sep '18

10 Sep '18

The patch ASoC: q6core: Use kmemdup to replace kzalloc + memcpy 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 18fbe800e6066050ab6ae7751708da04975cdc22 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Sat, 8 Sep 2018 16:36:19 +0800 Subject: [PATCH] ASoC: q6core: Use kmemdup to replace kzalloc + memcpy kmemdup has implemented the function that kzalloc() + memcpy() will do. and we prefer to use the kmemdup rather than the open coded implementation. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6core.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index 06f03a5fe9bd..ca1be7305524 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -105,12 +105,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) bytes = sizeof(*fwk) + fwk->num_services * sizeof(fwk->svc_api_info[0]); - core->fwk_version = kzalloc(bytes, GFP_ATOMIC); + core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC); if (!core->fwk_version) return -ENOMEM; - memcpy(core->fwk_version, data->payload, bytes); - core->fwk_version_supported = true; core->resp_received = true; @@ -124,12 +122,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]); - core->svc_version = kzalloc(len, GFP_ATOMIC); + core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC); if (!core->svc_version) return -ENOMEM; - memcpy(core->svc_version, data->payload, len); - core->get_version_supported = true; core->resp_received = true; -- 2.19.0.rc1

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[alsa-devel] Applied "ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy" to the asoc tree
by Mark Brown 10 Sep '18

10 Sep '18

The patch ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy 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 ca92cc4636fdedf0d7ee88a5e50cd2b85c246a3b Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Sat, 8 Sep 2018 16:36:20 +0800 Subject: [PATCH] ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy kmemdup has implemented the function that kzalloc() + memcpy() will do. and we prefer to kmemdup rather than the open coded implementation. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2620d77729c5..52a9915da0f5 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -898,11 +898,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, bc = (struct skl_algo_data *)sb->dobj.private; if (bc->set_params == SKL_PARAM_BIND) { - params = kzalloc(bc->max, GFP_KERNEL); + params = kmemdup(bc->params, bc->max, GFP_KERNEL); if (!params) return -ENOMEM; - memcpy(params, bc->params, bc->max); skl_fill_sink_instance_id(ctx, params, bc->max, mconfig); -- 2.19.0.rc1

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