June 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 5.4 019/266] ALSA: hda/realtek - Introduce polarity for micmute LED GPIO
by Sasha Levin 18 Jun '20

18 Jun '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit dbd13179780555ecd3c992dea1222ca31920e892 ] Currently mute LED and micmute LED share the same GPIO polarity. So split the polarity for mute and micmute, in case they have different polarities. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200430083255.5093-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index df5afac0b600..459a7d61326e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -81,6 +81,7 @@ struct alc_spec { /* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */ int mute_led_polarity; + int micmute_led_polarity; hda_nid_t mute_led_nid; hda_nid_t cap_mute_led_nid; @@ -4080,11 +4081,9 @@ static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec, /* update LED status via GPIO */ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, - bool enabled) + int polarity, bool enabled) { - struct alc_spec *spec = codec->spec; - - if (spec->mute_led_polarity) + if (polarity) enabled = !enabled; alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */ } @@ -4095,7 +4094,8 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled) struct hda_codec *codec = private_data; struct alc_spec *spec = codec->spec; - alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); + alc_update_gpio_led(codec, spec->gpio_mute_led_mask, + spec->mute_led_polarity, enabled); } /* turn on/off mic-mute LED via GPIO per capture hook */ @@ -4104,6 +4104,7 @@ static void alc_gpio_micmute_update(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_update_gpio_led(codec, spec->gpio_mic_led_mask, + spec->micmute_led_polarity, spec->gen.micmute_led.led_value); } @@ -5808,7 +5809,8 @@ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, snd_hda_gen_hp_automute(codec, jack); /* mute_led_polarity is set to 0, so we pass inverted value here */ - alc_update_gpio_led(codec, 0x10, !spec->gen.hp_jack_present); + alc_update_gpio_led(codec, 0x10, spec->mute_led_polarity, + !spec->gen.hp_jack_present); } /* Manage GPIOs for HP EliteBook Folio 9480m. -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 013/266] ASoC: davinci-mcasp: Fix dma_chan refcnt leak when getting dma type
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit a697ae6ea56e23397341b027098c1b11d9ab13da ] davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a reference of the specified dma_chan object to "chan" with increased refcnt. When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in one exception handling path of davinci_mcasp_get_dma_type(). When chan device is NULL, the function forgets to decrease the refcnt increased by dma_request_chan(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when chan device is NULL. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 7aa3c32e4a49..0541071f454b 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1875,8 +1875,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - if (WARN_ON(!chan->device || !chan->device->dev)) + if (WARN_ON(!chan->device || !chan->device->dev)) { + dma_release_channel(chan); return -EINVAL; + } if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 010/266] ASoC: fsl_esai: Disable exception interrupt before scheduling tasklet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 1fecbb71fe0e46b886f84e3b6decca6643c3af6d ] Disable exception interrupt before scheduling tasklet, otherwise if the tasklet isn't handled immediately, there will be endless xrun interrupt. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/a8f2ad955aac9e52587beedc1133b3efbe746895.15879688… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7a49d03463a..84290be778f0 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -87,6 +87,10 @@ static irqreturn_t esai_isr(int irq, void *devid) if ((saisr & (ESAI_SAISR_TUE | ESAI_SAISR_ROE)) && esai_priv->reset_at_xrun) { dev_dbg(&pdev->dev, "reset module for xrun\n"); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xEIE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xEIE_MASK, 0); tasklet_schedule(&esai_priv->task); } -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 005/266] ASoC: SOF: imx8: Fix randbuild error
by Sasha Levin 18 Jun '20

18 Jun '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit fe17e6cdc0fefca96ba9659be4b2b07487cbf0c5 ] when do randconfig like this: CONFIG_SND_SOC_SOF_IMX8_SUPPORT=y CONFIG_SND_SOC_SOF_IMX8=y CONFIG_SND_SOC_SOF_OF=y CONFIG_IMX_DSP=m CONFIG_IMX_SCU=y there is a link error: sound/soc/sof/imx/imx8.o: In function 'imx8_send_msg': imx8.c:(.text+0x380): undefined reference to 'imx_dsp_ring_doorbell' Select IMX_DSP in SND_SOC_SOF_IMX8_SUPPORT to fix this Fixes: f9ad75468453 ("ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Link: https://lore.kernel.org/r/20200409071832.2039-2-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 71f318bc2c74..b4f0426685c4 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -14,7 +14,7 @@ if SND_SOC_SOF_IMX_TOPLEVEL config SND_SOC_SOF_IMX8_SUPPORT bool "SOF support for i.MX8" depends on IMX_SCU - depends on IMX_DSP + select IMX_DSP help This adds support for Sound Open Firmware for NXP i.MX8 platforms Say Y if you have such a device. -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 003/266] ASoC: tegra: tegra_wm8903: Support nvidia, headset property
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dmitry Osipenko <digetx(a)gmail.com> [ Upstream commit 3ef9d5073b552d56bd6daf2af1e89b7e8d4df183 ] The microphone-jack state needs to be masked in a case of a 4-pin jack when microphone and ground pins are shorted. Presence of nvidia,headset tells that WM8903 CODEC driver should mask microphone's status if short circuit is detected, i.e headphones are inserted. Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> Link: https://lore.kernel.org/r/20200330204011.18465-3-digetx@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/tegra/tegra_wm8903.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 6211dfda2195..0fa01cacfec9 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -159,6 +159,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_component *component = codec_dai->component; struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + int shrt = 0; if (gpio_is_valid(machine->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det; @@ -171,12 +172,15 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) &tegra_wm8903_hp_jack_gpio); } + if (of_property_read_bool(card->dev->of_node, "nvidia,headset")) + shrt = SND_JACK_MICROPHONE; + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, &tegra_wm8903_mic_jack, tegra_wm8903_mic_jack_pins, ARRAY_SIZE(tegra_wm8903_mic_jack_pins)); wm8903_mic_detect(component, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, - 0); + shrt); snd_soc_dapm_force_enable_pin(&card->dapm, "MICBIAS"); -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 366/388] ASoC: rt5645: Add platform-data for Asus T101HA
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 79d4f823a06796656289f97b922493da5690e46c ] The Asus T101HA uses the default jack-detect mode 3, but instead of using an analog microphone it is using a DMIC on dmic-data-pin 1, like the Asus T100HA. Note unlike the T100HA its jack-detect is not inverted. Add a DMI quirk with the correct settings for this model. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608204634.93407-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 6ba1849a77b0..e2e1d5b03b38 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3625,6 +3625,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = { .inv_jd1_1 = true, }; +static const struct rt5645_platform_data asus_t101ha_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_IN2N, + .dmic2_data_pin = RT5645_DMIC2_DISABLE, + .jd_mode = 3, +}; + static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { .jd_mode = 3, .in2_diff = true, @@ -3708,6 +3714,14 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&asus_t100ha_platform_data, }, + { + .ident = "ASUS T101HA", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"), + }, + .driver_data = (void *)&asus_t101ha_platform_data, + }, { .ident = "MINIX Z83-4", .matches = { -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 365/388] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 199a5e8fda54ab3c8c6f6bf980c004e97ebf5ccb ] The Toshiba Encore WT10-A tablet almost fully works with the default settings for Bay Trail CR devices. The only issue is that it uses a digital mic. connected the the DMIC1 input instead of an analog mic. Add a quirk for this model using the default settings with the input-map replaced with BYT_RT5640_DMIC1_MAP. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608204634.93407-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index fbfd53874b47..5c1a5e2aff6f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -754,6 +754,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_JD_NOT_INV | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT10-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 364/388] ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit ba4e5abc6c4e173af7c941c03c067263b686665d ] With additional checks on dailinks, we see errors such as [ 3.000418] sof-nocodec sof-nocodec: CPU DAI DMIC01 Pin for rtd NoCodec-6 does not support playback It's not clear why we set the dpcm_playback and dpcm_capture flags unconditionally, add a check on number of channels for each direction to avoid invalid configurations. Fixes: 8017b8fd37bf5e ('ASoC: SOF: Add Nocodec machine driver support') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608194415.4663-5-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/nocodec.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c index 2233146386cc..71cf5f9db79d 100644 --- a/sound/soc/sof/nocodec.c +++ b/sound/soc/sof/nocodec.c @@ -52,8 +52,10 @@ static int sof_nocodec_bes_setup(struct device *dev, links[i].platforms->name = dev_name(dev); links[i].codecs->dai_name = "snd-soc-dummy-dai"; links[i].codecs->name = "snd-soc-dummy"; - links[i].dpcm_playback = 1; - links[i].dpcm_capture = 1; + if (ops->drv[i].playback.channels_min) + links[i].dpcm_playback = 1; + if (ops->drv[i].capture.channels_min) + links[i].dpcm_capture = 1; } card->dai_link = links; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 363/388] ASoC: core: only convert non DPCM link to DPCM link
by Sasha Levin 18 Jun '20

18 Jun '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 607fa205a7e4dfad28b8a67ab1c985756ddbccb0 ] Additional checks for valid DAIs expose a corner case, where existing BE dailinks get modified, e.g. HDMI links are tagged with dpcm_capture=1 even if the DAIs are for playback. This patch makes those changes conditional and flags configuration issues when a BE dailink is has no_pcm=0 but dpcm_playback or dpcm_capture=1 (which makes no sense). As discussed on the alsa-devel mailing list, there are redundant flags for dpcm_playback, dpcm_capture, playback_only, capture_only. This will have to be cleaned-up in a future update. For now only correct and flag problematic configurations. Fixes: 218fe9b7ec7f3 ("ASoC: soc-core: Set dpcm_playback / dpcm_capture") Suggested-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608194415.4663-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 843b8b1c89d4..e5433e8fcf19 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1720,9 +1720,25 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dai_link->platforms->name = component->name; /* convert non BE into BE */ - dai_link->no_pcm = 1; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + if (!dai_link->no_pcm) { + dai_link->no_pcm = 1; + + if (dai_link->dpcm_playback) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n", + dai_link->name); + if (dai_link->dpcm_capture) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n", + dai_link->name); + + /* convert normal link into DPCM one */ + if (!(dai_link->dpcm_playback || + dai_link->dpcm_capture)) { + dai_link->dpcm_playback = !dai_link->capture_only; + dai_link->dpcm_capture = !dai_link->playback_only; + } + } /* override any BE fixups */ dai_link->be_hw_params_fixup = -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 362/388] ASoC: soc-pcm: dpcm: fix playback/capture checks
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b73287f0b0745961b14e5ebcce92cc8ed24d4d52 ] Recent changes in the ASoC core prevent multi-cpu BE dailinks from being used. DPCM does support multi-cpu DAIs for BE Dailinks, but not for FE. Handle the FE checks first, and make sure all DAIs support the same capabilities within the same dailink. Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> BugLink: https://github.com/thesofproject/linux/issues/2031 Link: https://lore.kernel.org/r/20200608194415.4663-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 44 ++++++++++++++++++++++++++++++++++---------- 1 file changed, 34 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1f302de44052..39ce61c5b874 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2908,20 +2908,44 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + int stream; int i; + if (rtd->dai_link->dynamic && rtd->num_cpus > 1) { + dev_err(rtd->dev, + "DPCM doesn't support Multi CPU for Front-Ends yet\n"); + return -EINVAL; + } + if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - cpu_dai = asoc_rtd_to_cpu(rtd, 0); - if (rtd->num_cpus > 1) { - dev_err(rtd->dev, - "DPCM doesn't support Multi CPU yet\n"); - return -EINVAL; + if (rtd->dai_link->dpcm_playback) { + stream = SNDRV_PCM_STREAM_PLAYBACK; + + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + if (!snd_soc_dai_stream_valid(cpu_dai, + stream)) { + dev_err(rtd->card->dev, + "CPU DAI %s for rtd %s does not support playback\n", + cpu_dai->name, + rtd->dai_link->stream_name); + return -EINVAL; + } + playback = 1; + } + if (rtd->dai_link->dpcm_capture) { + stream = SNDRV_PCM_STREAM_CAPTURE; + + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + if (!snd_soc_dai_stream_valid(cpu_dai, + stream)) { + dev_err(rtd->card->dev, + "CPU DAI %s for rtd %s does not support capture\n", + cpu_dai->name, + rtd->dai_link->stream_name); + return -EINVAL; + } + capture = 1; } - - playback = rtd->dai_link->dpcm_playback && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); - capture = rtd->dai_link->dpcm_capture && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); } else { /* Adapt stream for codec2codec links */ int cpu_capture = rtd->dai_link->params ? -- 2.25.1

1 0