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

[alsa-devel] [PATCH] ASoC: topology: Use the correct dobj to free enum control values and texts
by Ranjani Sridharan 08 Apr '19

08 Apr '19

The control values and texts of the enum kcontrol associated with a widget need to be freed when the widget is removed. However, both struct snd_soc_dapm_widget and struct soc_enum contain a dobj member, which resulted in a confusion. The existing code generates a null pointer dereference by attempting to free the values and texts from the dobj which belongs to the widget instead of the dobj belonging to the enum kcontrol. The suggested fix is to use the correct dobj member (se->dobj) of the enum kcontrol. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-topology.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 472f7705da93..0f6a6ad01570 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -482,10 +482,11 @@ static void remove_widget(struct snd_soc_component *comp, snd_ctl_remove(card, kcontrol); - kfree(dobj->control.dvalues); + /* free enum kcontrol's dvalues and dtexts */ + kfree(se->dobj.control.dvalues); for (j = 0; j < se->items; j++) - kfree(dobj->control.dtexts[j]); - kfree(dobj->control.dtexts); + kfree(se->dobj.control.dtexts[j]); + kfree(se->dobj.control.dtexts); kfree(se); kfree(w->kcontrol_news[i].name); -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: stm32: fix sai driver name initialisation
by Arnaud Pouliquen 08 Apr '19

08 Apr '19

This patch fixes the sai driver structure overwriting which results in a cpu dai name equal NULL. Fixes: 3e086ed ("ASoC: stm32: add SAI driver") Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 2a90641..3dd54bc 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1420,7 +1420,6 @@ static int stm32_sai_sub_dais_init(struct platform_device *pdev, if (!sai->cpu_dai_drv) return -ENOMEM; - sai->cpu_dai_drv->name = dev_name(&pdev->dev); if (STM_SAI_IS_PLAYBACK(sai)) { memcpy(sai->cpu_dai_drv, &stm32_sai_playback_dai, sizeof(stm32_sai_playback_dai)); @@ -1430,6 +1429,7 @@ static int stm32_sai_sub_dais_init(struct platform_device *pdev, sizeof(stm32_sai_capture_dai)); sai->cpu_dai_drv->capture.stream_name = sai->cpu_dai_drv->name; } + sai->cpu_dai_drv->name = dev_name(&pdev->dev); return 0; } -- 2.7.4

2 1

[alsa-devel] [PATCH 00/20] Switch to SPDX identifier
by Andra Danciu 08 Apr '19

08 Apr '19

This patch set switches to SPDX identifier inside the fsl directory. Andra Danciu (20): ASoC: imx-pcm: Switch to SPDX identifier ASoC: efika-audio-fabric.c: Switch to SPDX identifier ASoC: eukrea-tlv320: Switch to SPDX identifier ASoC: fsl-dma: Switch to SPDX identifier ASoC: imx-audmux: Switch to SPDX identifier ASoC: imx-es8328: Switch to SPDX identifier ASoC: imx-mc13783: Switch to SPDX identifier ASoC: imx-pcm-fiq: Switch to SPDX identifier ASoC: imx-spdif: Switch to SPDX identifier ASoC: imx-ssi: Switch to SPDX identifier ASoC: mpc5200_dma: Switch to SPDX identifier ASoC: mpc5200_psc_ac97: Switch to SPDX identifier ASoC: mpc5200_psc_i2s: Switch to SPDX identifier ASoC: mpc8610_hpcd: Switch to SPDX identifier ASoC: mx27vis-aic32x4: Switch to SPDX identifier ASoC: p1022_ds: Switch to SPDX identifier ASoC: p1022_rdk: Switch to SPDX identifier ASoC: pcm030-audio-fabric: Switch to SPDX identifier ASoC: phycore-ac97: Switch to SPDX identifier ASoC: wm1133-ev1: Switch to SPDX identifier sound/soc/fsl/efika-audio-fabric.c | 17 ++++------- sound/soc/fsl/eukrea-tlv320.c | 26 +++++++---------- sound/soc/fsl/fsl_dma.c | 26 +++++++---------- sound/soc/fsl/fsl_dma.h | 5 +--- sound/soc/fsl/imx-audmux.c | 26 ++++++----------- sound/soc/fsl/imx-es8328.c | 15 +++------- sound/soc/fsl/imx-mc13783.c | 22 ++++++-------- sound/soc/fsl/imx-pcm-fiq.c | 21 ++++++-------- sound/soc/fsl/imx-pcm.h | 6 +--- sound/soc/fsl/imx-spdif.c | 13 ++------- sound/soc/fsl/imx-ssi.c | 57 ++++++++++++++++--------------------- sound/soc/fsl/imx-ssi.h | 6 +--- sound/soc/fsl/mpc5200_dma.c | 14 ++++----- sound/soc/fsl/mpc5200_psc_ac97.c | 16 ++++------- sound/soc/fsl/mpc5200_psc_i2s.c | 14 ++++----- sound/soc/fsl/mpc8610_hpcd.c | 18 +++++------- sound/soc/fsl/mx27vis-aic32x4.c | 29 +++++-------------- sound/soc/fsl/p1022_ds.c | 18 +++++------- sound/soc/fsl/p1022_rdk.c | 32 +++++++++------------ sound/soc/fsl/pcm030-audio-fabric.c | 18 +++++------- sound/soc/fsl/phycore-ac97.c | 16 ++++------- sound/soc/fsl/wm1133-ev1.c | 21 ++++++-------- 22 files changed, 161 insertions(+), 275 deletions(-) -- 2.11.0

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[alsa-devel] Applied "ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect" to the asoc tree
by Mark Brown 08 Apr '19

08 Apr '19

The patch ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect 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 ba49cf6f8e4ad44ee752eb13413a26a7191893ae Mon Sep 17 00:00:00 2001 From: Paul Cercueil <paul(a)crapouillou.net> Date: Fri, 29 Mar 2019 22:50:44 +0100 Subject: [PATCH] ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect Add a quirk to support boards whose jack detection mechanism is inverted. It will set the 'everest,jack-detect-inverted' boolean device property for the es8316 codec driver. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bytcht_es8316.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 6937c00cf63d..ccf552e92d53 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -40,6 +40,9 @@ #include "../atom/sst-atom-controls.h" #include "../common/sst-dsp.h" +/* jd-inv + terminating entry */ +#define MAX_NO_PROPS 2 + struct byt_cht_es8316_private { struct clk *mclk; struct snd_soc_jack jack; @@ -55,6 +58,7 @@ enum { #define BYT_CHT_ES8316_MAP(quirk) ((quirk) & GENMASK(3, 0)) #define BYT_CHT_ES8316_SSP0 BIT(16) #define BYT_CHT_ES8316_MONO_SPEAKER BIT(17) +#define BYT_CHT_ES8316_JD_INVERTED BIT(18) static int quirk; @@ -72,6 +76,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0 enabled"); if (quirk & BYT_CHT_ES8316_MONO_SPEAKER) dev_info(dev, "quirk MONO_SPEAKER enabled\n"); + if (quirk & BYT_CHT_ES8316_JD_INVERTED) + dev_info(dev, "quirk JD_INVERTED enabled\n"); } static int byt_cht_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, @@ -438,12 +444,14 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { static const char * const mic_name[] = { "in1", "in2" }; + struct property_entry props[MAX_NO_PROPS] = {}; struct byt_cht_es8316_private *priv; struct device *dev = &pdev->dev; struct snd_soc_acpi_mach *mach; const char *platform_name; struct acpi_device *adev; struct device *codec_dev; + unsigned int cnt = 0; int dai_index = 0; int i; int ret = 0; @@ -513,6 +521,15 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) if (!codec_dev) return -EPROBE_DEFER; + if (quirk & BYT_CHT_ES8316_JD_INVERTED) + props[cnt++] = PROPERTY_ENTRY_BOOL("everest,jack-detect-inverted"); + + if (cnt) { + ret = device_add_properties(codec_dev, props); + if (ret) + return ret; + } + devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios); priv->speaker_en_gpio = gpiod_get_index(codec_dev, "speaker-enable", 0, -- 2.20.1

1 0

[alsa-devel] [PATCH 1/3] ASoC: es8316: Add support for inverted jack detect
by Paul Cercueil 08 Apr '19

08 Apr '19

On some devices (Teclast X98+ II tablet, maybe others), the jack detection has been wired backwards, so when the ES8316 reports headphones being present it means they are actually not plugged. Use a quirk around this incorrect behaviour, which can be enabled through the 'everest,jack-detect-inverted' boolean device property. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- sound/soc/codecs/es8316.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 6d4a323f786b..ec2770b3f77d 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -43,6 +43,7 @@ struct es8316_priv { unsigned int sysclk; unsigned int allowed_rates[NR_SUPPORTED_MCLK_LRCK_RATIOS]; struct snd_pcm_hw_constraint_list sysclk_constraints; + bool jd_inverted; }; /* @@ -577,6 +578,9 @@ static irqreturn_t es8316_irq(int irq, void *data) if (!es8316->jack) goto out; + if (es8316->jd_inverted) + flags ^= ES8316_GPIO_FLAG_HP_NOT_INSERTED; + dev_dbg(comp->dev, "gpio flags %#04x\n", flags); if (flags & ES8316_GPIO_FLAG_HP_NOT_INSERTED) { /* Jack removed, or spurious IRQ? */ @@ -592,6 +596,8 @@ static irqreturn_t es8316_irq(int irq, void *data) /* Jack inserted, determine type */ es8316_enable_micbias_for_mic_gnd_short_detect(comp); regmap_read(es8316->regmap, ES8316_GPIO_FLAG, &flags); + if (es8316->jd_inverted) + flags ^= ES8316_GPIO_FLAG_HP_NOT_INSERTED; dev_dbg(comp->dev, "gpio flags %#04x\n", flags); if (flags & ES8316_GPIO_FLAG_HP_NOT_INSERTED) { /* Jack unplugged underneath us */ @@ -633,6 +639,14 @@ static void es8316_enable_jack_detect(struct snd_soc_component *component, { struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + /* + * Init es8316->jd_inverted here and not in the probe, as we cannot + * guarantee that the bytchr-es8316 driver, which might set this + * property, will probe before us. + */ + es8316->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + mutex_lock(&es8316->lock); es8316->jack = jack; -- 2.11.0

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[alsa-devel] Applied "ASoC: core: conditionally increase module refcount on component open" to the asoc tree
by Mark Brown 08 Apr '19

08 Apr '19

The patch ASoC: core: conditionally increase module refcount on component open 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 b4ed6b51f356224c6c71540ed94087f7f09b84af Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Date: Fri, 5 Apr 2019 09:57:08 -0700 Subject: [PATCH] ASoC: core: conditionally increase module refcount on component open Recently, for Intel platforms the "ignore_module_refcount" field was introduced for the component driver. In order to avoid a deadlock preventing the PCI modules from being removed even when the card was idle, the refcounts were not incremented for the device driver module during component probe. However, this change introduced a nasty side effect: the device driver module can be unloaded while a pcm stream is open. This patch proposes to change the field to be renamed as "module_get_upon_open". When this field is set, the module refcount should be incremented on pcm open amd decremented upon pcm close. This will enable modules to be removed when no PCM playback/capture happens and prevent removal when the component is actually in use. Also, align with the skylake component driver with the new name. Fixes: b450b878('ASoC: core: don't increase component module refcount unconditionally' Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 9 +++++++-- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/soc-core.c | 4 ++-- 3 files changed, 10 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1e2be35ed36f..482b4ea87c3c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -802,8 +802,13 @@ struct snd_soc_component_driver { int probe_order; int remove_order; - /* signal if the module handling the component cannot be removed */ - unsigned int ignore_module_refcount:1; + /* + * signal if the module handling the component should not be removed + * if a pcm is open. Setting this would prevent the module + * refcount being incremented in probe() but allow it be incremented + * when a pcm is opened and decremented when it is closed. + */ + unsigned int module_get_upon_open:1; /* bits */ unsigned int idle_bias_on:1; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 57031b6d4d45..9735e2412251 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1475,7 +1475,7 @@ static const struct snd_soc_component_driver skl_component = { .ops = &skl_platform_ops, .pcm_new = skl_pcm_new, .pcm_free = skl_pcm_free, - .ignore_module_refcount = 1, /* do not increase the refcount in core */ + .module_get_upon_open = 1, /* increment refcount when a pcm is opened */ }; int skl_platform_register(struct device *dev) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d88757659729..46e3ab0fced4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -947,7 +947,7 @@ static void soc_cleanup_component(struct snd_soc_component *component) snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; - if (!component->driver->ignore_module_refcount) + if (!component->driver->module_get_upon_open) module_put(component->dev->driver->owner); } @@ -1381,7 +1381,7 @@ static int soc_probe_component(struct snd_soc_card *card, return 0; } - if (!component->driver->ignore_module_refcount && + if (!component->driver->module_get_upon_open && !try_module_get(component->dev->driver->owner)) return -ENODEV; -- 2.20.1

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: core: conditionally increase module refcount on component open
by Ranjani Sridharan 08 Apr '19

08 Apr '19

Recently, for Intel platforms the "ignore_module_refcount" field was introduced for the component driver. In order to avoid a deadlock preventing the PCI modules from being removed even when the card was idle, the refcounts were not incremented for the device driver module during component probe. However, this change introduced a nasty side effect: the device driver module can be unloaded while a pcm stream is open. This patch proposes to change the field to be renamed as "module_get_upon_open". When this field is set, the module refcount should be incremented on pcm open amd decremented upon pcm close. This will enable modules to be removed when no PCM playback/capture happens and prevent removal when the component is actually in use. Also, align with the skylake component driver with the new name. Fixes: b450b878('ASoC: core: don't increase component module refcount unconditionally' Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- include/sound/soc.h | 9 +++++++-- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/soc-core.c | 4 ++-- 3 files changed, 10 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1e2be35ed36f..482b4ea87c3c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -802,8 +802,13 @@ struct snd_soc_component_driver { int probe_order; int remove_order; - /* signal if the module handling the component cannot be removed */ - unsigned int ignore_module_refcount:1; + /* + * signal if the module handling the component should not be removed + * if a pcm is open. Setting this would prevent the module + * refcount being incremented in probe() but allow it be incremented + * when a pcm is opened and decremented when it is closed. + */ + unsigned int module_get_upon_open:1; /* bits */ unsigned int idle_bias_on:1; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 56099db8f86d..ee53d047c3bd 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1468,7 +1468,7 @@ static const struct snd_soc_component_driver skl_component = { .ops = &skl_platform_ops, .pcm_new = skl_pcm_new, .pcm_free = skl_pcm_free, - .ignore_module_refcount = 1, /* do not increase the refcount in core */ + .module_get_upon_open = 1, /* increment refcount when a pcm is opened */ }; int skl_platform_register(struct device *dev) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6f4842977b8d..e78534bbd092 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -947,7 +947,7 @@ static void soc_cleanup_component(struct snd_soc_component *component) snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; - if (!component->driver->ignore_module_refcount) + if (!component->driver->module_get_upon_open) module_put(component->dev->driver->owner); } @@ -1381,7 +1381,7 @@ static int soc_probe_component(struct snd_soc_card *card, return 0; } - if (!component->driver->ignore_module_refcount && + if (!component->driver->module_get_upon_open && !try_module_get(component->dev->driver->owner)) return -ENODEV; -- 2.17.1

3 3

[alsa-devel] [PATCH] ASoC: Intel:kbl_da7219_max98927: remove headset
by Jenny TC 08 Apr '19

08 Apr '19

kbl_da7219_max98927 supports two configurations - One with da7219 & max98927/373 and other one with max98927/373 alone. This patch removes the headset dais from max98927/373 configuration. Since the snd_soc_dapm_ignore_suspend for DMIC is called from da7219_codec_init, the code is re-arranged to invoke snd_soc_dapm_ignore_suspend from kabylake_dmic_init. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 52 ++++++++++++---------------- 1 file changed, 23 insertions(+), 29 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 2768a57..f72a7bf 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -52,7 +52,6 @@ struct kbl_codec_private { enum { KBL_DPCM_AUDIO_PB = 0, - KBL_DPCM_AUDIO_CP, KBL_DPCM_AUDIO_ECHO_REF_CP, KBL_DPCM_AUDIO_REF_CP, KBL_DPCM_AUDIO_DMIC_CP, @@ -60,6 +59,7 @@ enum { KBL_DPCM_AUDIO_HDMI2_PB, KBL_DPCM_AUDIO_HDMI3_PB, KBL_DPCM_AUDIO_HS_PB, + KBL_DPCM_AUDIO_CP, }; static int platform_clock_control(struct snd_soc_dapm_widget *w, @@ -311,6 +311,12 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) da7219_aad_jack_det(component, &ctx->kabylake_headset); + return 0; +} + +static int kabylake_dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); if (ret) dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret); @@ -581,20 +587,6 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) .dpcm_playback = 1, .ops = &kabylake_da7219_fe_ops, }, - [KBL_DPCM_AUDIO_CP] = { - .name = "Kbl Audio Capture Port", - .stream_name = "Audio Record", - .cpu_dai_name = "System Pin", - .platform_name = "0000:00:1f.3", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &kabylake_da7219_fe_ops, - }, [KBL_DPCM_AUDIO_ECHO_REF_CP] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", @@ -690,6 +682,20 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) .ops = &kabylake_da7219_fe_ops, }, + [KBL_DPCM_AUDIO_CP] = { + .name = "Kbl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + .ops = &kabylake_da7219_fe_ops, + }, /* Back End DAI links */ { @@ -733,6 +739,7 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", + .init = kabylake_dmic_init, .platform_name = "0000:00:1f.3", .be_hw_params_fixup = kabylake_dmic_fixup, .ignore_suspend = 1, @@ -792,20 +799,6 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) .dpcm_playback = 1, .ops = &kabylake_da7219_fe_ops, }, - [KBL_DPCM_AUDIO_CP] = { - .name = "Kbl Audio Capture Port", - .stream_name = "Audio Record", - .cpu_dai_name = "System Pin", - .platform_name = "0000:00:1f.3", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &kabylake_da7219_fe_ops, - }, [KBL_DPCM_AUDIO_ECHO_REF_CP] = { .name = "Kbl Audio Echo Reference cap", .stream_name = "Echoreference Capture", @@ -911,6 +904,7 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", + .init = kabylake_dmic_init, .platform_name = "0000:00:1f.3", .be_hw_params_fixup = kabylake_dmic_fixup, .ignore_suspend = 1, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Fix spacing
by Annaliese McDermond 08 Apr '19

08 Apr '19

Fix some cosmetic spacing issues reported by Julia Lawall <julia.lawall(a)lip6.fr>. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4-clk.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 667ec2c03508..9e4899eb1d8e 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -165,8 +165,8 @@ static int clk_aic32x4_pll_calc_muldiv(struct clk_aic32x4_pll_muldiv *settings, * of the multiplier. This is because we can't do floating point * math in the kernel. */ - multiplier = (u64) rate * settings->p * 10000; - do_div(multiplier, parent_rate); + multiplier = (u64) rate * settings->p * 10000; + do_div(multiplier, parent_rate); /* * J can't be over 64, so R can scale this. -- 2.19.1

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Re: [alsa-devel] 'modern dailink' transition
by Mark Brown 08 Apr '19

08 Apr '19

On Mon, Mar 25, 2019 at 10:54:52AM +0900, Kuninori Morimoto wrote: > This is my understanding, please correct me if I was wrong. > I think current many sound card which doesn't need "platfrom" are 2 patterns. > 1) select snd-soc-dummy as platfrom > 2) select cpu component as platfrom > Current ASoC selects 1) automatically if .platfrom_name was NULL. > And driver needs to have below if it want to be 2) > dai_link->platform_of_node = dai_link->cpu_of_node > But, I think one of them is enough ? > I mean select 2) automatically can be OK? > In other words, current some sound card which doesn't need > platfrom is calling snd-soc-dummy platfrom method in 1) case. > But, is it needed ? I'm not sure... > It seems snd-soc-dummy platfrom is caring about DPCM-BE case, > but, I think CPU is snd-soc-dummy in such case. > Maybe we need same cade to dummy CPU (?), but *my* DPCM system > is working correctly without it. It *should* work without it. If it actually does work is a separate question - that code is a bit fragile so there may be some problems. Since the most complicated user of DPCM is x86 I think if a change to this stuff tests out well there it should be OK.

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