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

[alsa-devel] [PATCH] ALSA: hda/ca0132 - remove redundant assignment to variable 'changed'
by Colin King 04 Jul '19

04 Jul '19

From: Colin Ian King <colin.king(a)canonical.com> The variable 'changed' is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c0565437ffed..0d51823d7270 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -5980,7 +5980,7 @@ static int ca0132_alt_volume_put(struct snd_kcontrol *kcontrol, int ch = get_amp_channels(kcontrol); long *valp = ucontrol->value.integer.value; hda_nid_t vnid = 0; - int changed = 1; + int changed; switch (nid) { case 0x02: -- 2.20.1

2 1

[alsa-devel] [PATCH v2 33/35] sound/pci: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 04 Jul '19

04 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memcpy, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memcpy. Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- Changes in v2: - Fix a typo in commit message (memset -> memcpy) sound/pci/echoaudio/echoaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index b612a536a5a1..35bd3e7c8ce1 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2189,11 +2189,10 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - commpage_bak = kmalloc(sizeof(*commpage), GFP_KERNEL); + commpage_bak = kmemdup(commpage, sizeof(*commpage), GFP_KERNEL); if (commpage_bak == NULL) return -ENOMEM; commpage = chip->comm_page; - memcpy(commpage_bak, commpage, sizeof(*commpage)); err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); if (err < 0) { -- 2.11.0

2 1

[alsa-devel] Dell Headphone Mic can't record after S3
by Kailang 04 Jul '19

04 Jul '19

Hi Takashi, Headphone Mic can't record after S3 on Dell headset mode platform. The S3 mode was deep. S2idle didn't have this issue. BR, Kailang

2 1

[alsa-devel] [PATCH] ASoC: core: Return -ENOTSUPP from set_channel_map() if no operation provided
by Srinivas Kandagatla 04 Jul '19

04 Jul '19

It makes it easier for common code to work with snd_soc_dai_set_channel_map() by distinguishing between operation not being supported and an error. This is done inline with others snd_soc_dai.* apis. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f08bef7328d7..62aae5a5c14b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2688,7 +2688,7 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, return dai->driver->ops->set_channel_map(dai, tx_num, tx_slot, rx_num, rx_slot); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); -- 2.21.0

2 1

[alsa-devel] [PATCH 0/2] ASoC: meson: axg-tdm-formatter: add g12a reset
by Jerome Brunet 04 Jul '19

04 Jul '19

This patchset add the possibility to provide a reset to the tdm formatter. Such reset is available on the g12a SoC family and helps solve a random channel output shift when using more than one output lane. Jerome Brunet (2): ASoC: meson: axg-tdm-formatter: add reset to the bindings documentation ASoC: meson: axg-tdm-formatter: add reset .../sound/amlogic,axg-tdm-formatters.txt | 6 +++- sound/soc/meson/axg-tdm-formatter.c | 29 +++++++++++++++++++ 2 files changed, 34 insertions(+), 1 deletion(-) -- 2.21.0

2 4

[alsa-devel] [v3] ASoC: max98357a: avoid speaker pop when playback startup
by mac.chiang＠intel.com 04 Jul '19

04 Jul '19

From: Mac Chiang <mac.chiang(a)intel.com> Loud speaker pop happens during playback even when in slience playback. Specify Max98357a amp delay times to make sure clocks are always earlier than sdmode on. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v3: - rebase to branch for-next v2: - added device_property_read_u32() func --- --- .../devicetree/bindings/sound/max98357a.txt | 4 ++ sound/soc/codecs/max98357a.c | 55 ++++++++++++++++++---- 2 files changed, 50 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt index 28645a2..4bce14c 100644 --- a/Documentation/devicetree/bindings/sound/max98357a.txt +++ b/Documentation/devicetree/bindings/sound/max98357a.txt @@ -9,6 +9,10 @@ Optional properties: - sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. If this option is not specified then driver does not manage the pin state (e.g. chip is always on). +- sdmode-delay : specify delay time for SD_MODE pin. + If this option is specified, which means it's required i2s clocks + ready before SD_MODE is unmuted in order to avoid the speaker pop noise. + It's observed that 5ms is sufficient. Example: diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 6f724c9..6f0e28f 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -19,24 +19,42 @@ #include <sound/soc-dai.h> #include <sound/soc-dapm.h> +struct max98357a_priv { + struct delayed_work enable_sdmode_work; + struct gpio_desc *sdmode; + unsigned int sdmode_delay; +}; + +static void max98357a_enable_sdmode_work(struct work_struct *work) +{ + struct max98357a_priv *max98357a = + container_of(work, struct max98357a_priv, + enable_sdmode_work.work); + + gpiod_set_value(max98357a->sdmode, 1); +} + static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + struct max98357a_priv *max98357a = snd_soc_dai_get_drvdata(dai); - if (!sdmode) + if (!max98357a->sdmode) return 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(sdmode, 1); + queue_delayed_work(system_power_efficient_wq, + &max98357a->enable_sdmode_work, + msecs_to_jiffies(max98357a->sdmode_delay)); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(sdmode, 0); + cancel_delayed_work_sync(&max98357a->enable_sdmode_work); + gpiod_set_value(max98357a->sdmode, 0); break; } @@ -90,14 +108,33 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { static int max98357a_platform_probe(struct platform_device *pdev) { - struct gpio_desc *sdmode; + struct max98357a_priv *max98357a; + int ret; - sdmode = devm_gpiod_get_optional(&pdev->dev, + max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL); + + if (!max98357a) + return -ENOMEM; + + max98357a->sdmode = devm_gpiod_get_optional(&pdev->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) - return PTR_ERR(sdmode); - dev_set_drvdata(&pdev->dev, sdmode); + if (IS_ERR(max98357a->sdmode)) + return PTR_ERR(max98357a->sdmode); + + ret = device_property_read_u32(&pdev->dev, "sdmode-delay", + &max98357a->sdmode_delay); + + if (ret) { + max98357a->sdmode_delay = 0; + dev_dbg(&pdev->dev, + "no optional property 'sdmode-delay' found, default: no delay\n"); + } + + dev_set_drvdata(&pdev->dev, max98357a); + + INIT_DELAYED_WORK(&max98357a->enable_sdmode_work, + max98357a_enable_sdmode_work); return devm_snd_soc_register_component(&pdev->dev, &max98357a_component_driver, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: audio-graph-card: fix use-after-free in graph_for_each_link
by Wen Yang 04 Jul '19

04 Jul '19

After calling of_node_put() on the codec_ep and codec_port variables, they are still being used, which may result in use-after-free. We fix this issue by calling of_node_put() after the last usage. Fixes: fce9b90c1ab7 ("ASoC: audio-graph-card: cleanup DAI link loop method - step2") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> 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: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/generic/audio-graph-card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index e438011..30a4e83 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -421,9 +421,6 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = of_get_parent(codec_ep); - of_node_put(codec_ep); - of_node_put(codec_port); - /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); graph_parse_convert(dev, codec_ep, &adata); @@ -443,6 +440,9 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, else ret = func_noml(priv, cpu_ep, codec_ep, li); + of_node_put(codec_ep); + of_node_put(codec_port); + if (ret < 0) return ret; -- 2.9.5

2 1

[alsa-devel] [PATCH] ASoC: qdsp6: q6afe-dai: Add missing Slimbus0 audio route
by Srinivas Kandagatla 04 Jul '19

04 Jul '19

For some reason SLIMBus RX0 playback is not added to audio routes. This patch adds the missing route. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index dc645ba4d8d0..c1a7624eaf17 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -447,6 +447,7 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"HDMI Playback", NULL, "HDMI_RX"}, {"Display Port Playback", NULL, "DISPLAY_PORT_RX"}, + {"Slimbus Playback", NULL, "SLIMBUS_0_RX"}, {"Slimbus1 Playback", NULL, "SLIMBUS_1_RX"}, {"Slimbus2 Playback", NULL, "SLIMBUS_2_RX"}, {"Slimbus3 Playback", NULL, "SLIMBUS_3_RX"}, -- 2.21.0

2 1

[alsa-devel] [PATCH v2 34/35] sound/soc/codecs: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 04 Jul '19

04 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memcpy, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memcpy. Acked-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- Changes in v2: - Fix a typo in commit message (memset -> memcpy) - Split into two patches sound/soc/codecs/wm0010.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 727d6703c905..807826f30f58 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -515,7 +515,7 @@ static int wm0010_stage2_load(struct snd_soc_component *component) dev_dbg(component->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ - img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); + img = kmemdup(&fw->data[0], fw->size, GFP_KERNEL | GFP_DMA); if (!img) { ret = -ENOMEM; goto abort2; @@ -527,8 +527,6 @@ static int wm0010_stage2_load(struct snd_soc_component *component) goto abort1; } - memcpy(img, &fw->data[0], fw->size); - spi_message_init(&m); memset(&t, 0, sizeof(t)); t.rx_buf = out; -- 2.11.0

2 1

[alsa-devel] GIT: Regarding the issue we are facing in the commit 37728639ae05de702825d96bd1d42e24ae772248
by vanitha.channaiah＠in.bosch.com 04 Jul '19

04 Jul '19

This mail is regarding the fix at below: commit 37728639ae05de702825d96bd1d42e24ae772248 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Sat Feb 7 15:01:31 2004 +00 The commit has following changes : ******************************* code ***************************************** static int snd_pcm_rate_hw_refine_cchange(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_hw_params_t *sparams) { + buffer_size = snd_pcm_hw_param_get_interval(params, SND_PCM_HW_PARAM_BUFFER_SIZE); + /* + * this condition probably needs more work: + * in case when the buffer_size is known and we are looking + * for best period_size, we should prefer situation when + * (buffer_size / period_size) * period_size == buffer_size + */ + if (snd_interval_single(buffer_size) && buffer_size->integer) { + snd_interval_t *period_size; + period_size = (snd_interval_t *)snd_pcm_hw_param_get_interval(params, SND_PCM_HW_PARAM_PERIOD_SIZE); + if (!snd_interval_checkempty(period_size) && + period_size->openmin && period_size->openmax && + period_size->min + 1 == period_size->max) { + if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { + snd_interval_set_value(period_size, period_size->min); + } else if ((buffer_size->max / period_size->max) * period_size->max == buffer_size->max) { + snd_interval_set_value(period_size, period_size->max); + } + } + } } ******************************* code **************************************** Update rate depending on period_size and period_time. Issue found in Hardware : a. RCAR salvator-xs which supports 2 channel. b. IMX which supports 8 channel. 1. Usecase which is PASSED without the mentioned commit and FAILED with the commit. For the usecase with below command : rate = 11025 channel = 6 $aplay -Dentertainment_main -r11025 -c6 -fS16_LE /dev/urandom a. With the commit: In the calculation of RATE at Rule 7: RATE=11025 dependent parameters are: PERIOD_SIZE=88 PERIOD_TIME=8000 parameters min max open_min open_max interval PERIOD_SIZE 88 88 0 0 0 PERIO_TIME 8000 8000 0 0 0 RATE 11025 11000 0 0 1 RATE values are calculated in snd_interval_refine() Return value: INVALID as rate_min > rate_max This is because, dependent parameter “period_size” is rounded to 88 in “rate plugin” in snd_pcm_rate_hw_refine_cchange() Since, the below condition gets satisfied(buffer_size = 352 aligned to period_size = 88), period_size gets rounded to 88. if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { This commit changes causing the issue to get rate_min > rate_max. The flow of code execution is as follows: - snd_pcm_hw_refine_slave() - Enters snd_pcm_rate_hw_refine_cchange(), rounding of period_size to 88. - snd_pcm_rate_hw_refine_cchange() exit. - snd_pcm_hw_refine_soft() is called, here exists params->rmask to evaluate for SAMPLE_BITS, FRAME_BITS, PERIOD_BYTES, BUFFER_BYTES - execution of RULES calculation. At RULE 7 rate calculation INVALID error is observed. b. Without the commit: In the case if period_size doesn’t get rounded off to 88, the usecase would get PASSED. Rate calculation goes fine with period_size open interval (88, 89) parameters min max open_min open_max interval PERIOD_SIZE 88 89 0 0 0 PERIOD_TIME 8000 8000 0 0 0 RATE 11025 11025 0 0 1 Here, there is no issue of rate_min > rate_max. **************************************************************************************************************************************************************** Now, I just have one scenario where with/without commit, rate calculation goes fine. 2. Usecase which is PASSED with and without above commit. For the usecase with below command : rate =11025 channel=2 $aplay -Dentertainment_main -r11025 -c2 -fS16_LE /dev/urandom a. Including the above commit code: Below is the calculation that goes: The flow of code execution is as follows: - snd_pcm_hw_refine_slave() - Enters snd_pcm_rate_hw_refine_cchange() , rounding of period_size to 88. - snd_pcm_rate_hw_refine_cchange() exit. - snd_pcm_hw_refine_soft() is called, here params->rmask is 0. - RULES calculation doesnt occur. b. Without the commit: In the case if period_size doesn’t get rounded off to 88. Rate calculation goes fine with period_size open interval (88, 89) Is there any dependency for the commit ? If yes, can you please suggest the corner case which fails without the commit ? Can we revert the changes ?

3 3