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

[PATCH v2 1/1] ASoC: soc-component: add snd_soc_component_ack()
by Gyeongtaek Lee 03 Apr '20

03 Apr '20

Current ALSA SoC is directly using component->driver->ops->xxx, thus, it is deep nested, and makes code difficult to read, and is not good for encapsulation. This patch adds new snd_soc_component_ack() and use it. Ack callback is revived because some ASoC driver is using it. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- include/sound/soc-component.h | 1 + sound/soc/soc-component.c | 18 ++++++++++++++++++ sound/soc/soc-pcm.c | 2 ++ 3 files changed, 21 insertions(+) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2eef525..50b079272b47 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -381,6 +381,7 @@ struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream, unsigned long offset); int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream); int snd_soc_pcm_component_new(struct snd_pcm *pcm); void snd_soc_pcm_component_free(struct snd_pcm *pcm); diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 79ffc2820ba9..b4be91cb9fa0 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -526,6 +526,24 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, return -EINVAL; } +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + /* FIXME. it returns 1st ack now */ + if (component->driver->ops && + component->driver->ops->ack) + return component->driver->ops->ack(substream); + } + + return -EINVAL; +} + int snd_soc_pcm_component_new(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b600d3eaaf5c..c8a5a9a9d35f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2941,6 +2941,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (!ops) continue; + if (ops->ack) + rtd->ops.ack = snd_soc_pcm_component_ack; if (ops->copy_user) rtd->ops.copy_user = snd_soc_pcm_component_copy_user; if (ops->page) base-commit: ef8e14794308a428b194f8b06ad9ae06b43466e4 -- 2.21.0

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[PATCH 1/1] ASoC: soc-component: add snd_soc_component_ack()
by Gyeongtaek Lee 03 Apr '20

03 Apr '20

Current ALSA SoC is directly using component->driver->ops->xxx, thus, it is deep nested, and makes code difficult to read, and is not good for encapsulation. This patch adds new snd_soc_component_ack() and use it. Ack callback is revived because some ASoC driver is using it. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- include/sound/soc-component.h | 3 +++ sound/soc/soc-component.c | 14 ++++++++++++++ sound/soc/soc-pcm.c | 2 ++ 3 files changed, 19 insertions(+) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 154d02fbbfed..a8c60efac8f7 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -104,6 +104,8 @@ struct snd_soc_component_driver { struct page *(*page)(struct snd_soc_component *component, struct snd_pcm_substream *substream, unsigned long offset); + int (*ack) (struct snd_soc_component *component, + struct snd_pcm_substream *substream); int (*mmap)(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct vm_area_struct *vma); @@ -415,6 +417,7 @@ struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream, unsigned long offset); int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream); int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd); void snd_soc_pcm_component_free(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 785a0385cc7f..b4b3245ef016 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -516,6 +516,20 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, return -EINVAL; } +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + int i; + + /* FIXME. it returns 1st ack now */ + for_each_rtd_components(rtd, i, component) + if (component->driver->ack) + return component->driver->ack(component, substream); + + return -EINVAL; +} + int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 289aebc15529..c6fddd1cb5aa 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -3027,6 +3027,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->ops.ioctl = snd_soc_pcm_component_ioctl; if (drv->sync_stop) rtd->ops.sync_stop = snd_soc_pcm_component_sync_stop; + if (drv->ack) + rtd->ops.ack = snd_soc_pcm_component_ack; if (drv->copy_user) rtd->ops.copy_user = snd_soc_pcm_component_copy_user; if (drv->page) -- 2.21.0

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[PATCH 1/1] ASoC: soc-component: add snd_soc_component_ack()
by Gyeongtaek Lee 03 Apr '20

03 Apr '20

Current ALSA SoC is directly using component->driver->ops->xxx, thus, it is deep nested, and makes code difficult to read, and is not good for encapsulation. This patch adds new snd_soc_component_ack() and use it. Ack callback is revived because some ASoC driver is using it. Change-Id: I85573cb43f3ca050c47ca9e3af3cfa9ec1fae3d0 Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- include/sound/soc-component.h | 1 + sound/soc/soc-component.c | 18 ++++++++++++++++++ sound/soc/soc-pcm.c | 2 ++ 3 files changed, 21 insertions(+) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2eef525..50b079272b47 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -381,6 +381,7 @@ struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream, unsigned long offset); int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream); int snd_soc_pcm_component_new(struct snd_pcm *pcm); void snd_soc_pcm_component_free(struct snd_pcm *pcm); diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 79ffc2820ba9..b4be91cb9fa0 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -526,6 +526,24 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, return -EINVAL; } +int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + /* FIXME. it returns 1st ack now */ + if (component->driver->ops && + component->driver->ops->ack) + return component->driver->ops->ack(substream); + } + + return -EINVAL; +} + int snd_soc_pcm_component_new(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b600d3eaaf5c..c8a5a9a9d35f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2941,6 +2941,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (!ops) continue; + if (ops->ack) + rtd->ops.ack = snd_soc_pcm_component_ack; if (ops->copy_user) rtd->ops.copy_user = snd_soc_pcm_component_copy_user; if (ops->page) -- 2.21.0

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[PATCH 0/1] reviving ack callback for 5.4
by Gyeongtaek Lee 03 Apr '20

03 Apr '20

ACK callback is useful to notify application pointer in DMA buffer to DSP, because it is called after application pointer is updated in ALSA. DSP can use the pointer to control preprocessing, detect underrun and so on. I'd like to revive it from 5.4 if it is possible. Gyeongtaek Lee (1): ASoC: soc-component: add snd_soc_component_ack() include/sound/soc-component.h | 1 + sound/soc/soc-component.c | 18 ++++++++++++++++++ sound/soc/soc-pcm.c | 2 ++ 3 files changed, 21 insertions(+) base-commit: ef8e14794308a428b194f8b06ad9ae06b43466e4 -- 2.21.0

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[PATCH] ALSA: hda/realtek - Add quirk for Lenovo Carbon X1 8th gen
by Hans de Goede 02 Apr '20

02 Apr '20

The audio setup on the Lenovo Carbon X1 8th gen is the same as that on the Lenovo Carbon X1 7th gen, as such it needs the same ALC285_FIXUP_THINKPAD_HEADSET_JACK quirk. This fixes volume control of the speaker not working among other things. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1820196 Cc: stable(a)vger.kernel.org Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63e1a56f705b..9c3bbf1df93e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7299,6 +7299,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.26.0

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sof-hda-dsp: microphone device named "Headphones2"
by GitHub issues - opened 02 Apr '20

02 Apr '20

alsa-project/alsa-ucm-conf issue #19 was opened from tanuk: I was looking at the sof-hda-dsp UCM config, and I noticed that it defines a device named "Headphones2" which seems to actually represent a microphone. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/19 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: Intel: Multiple I/O PCM format support for pipe
by Mateusz Gorski 02 Apr '20

02 Apr '20

If a pipe supports multiple input/output formats, kcontrol is created and selection of pipe input and output configuration is done based on control set. Effectively allows user to select configuration using amixer settings. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> Signed-off-by: S, Pavan K <pavan.k.s(a)intel.com> --- include/uapi/sound/skl-tplg-interface.h | 1 + sound/soc/intel/skylake/skl-topology.c | 110 ++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 1 + 3 files changed, 112 insertions(+) diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index 9eee32f5e407..f2711186c81f 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -18,6 +18,7 @@ */ #define SKL_CONTROL_TYPE_BYTE_TLV 0x100 #define SKL_CONTROL_TYPE_MIC_SELECT 0x102 +#define SKL_CONTROL_TYPE_MULTI_IO_SELECT 0x103 #define HDA_SST_CFG_MAX 900 /* size of copier cfg*/ #define MAX_IN_QUEUE 8 diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 344b06df0e15..de60b6873666 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -578,6 +578,38 @@ static int skl_tplg_unload_pipe_modules(struct skl_dev *skl, return ret; } +static bool skl_tplg_is_multi_fmt(struct skl_dev *skl, struct skl_pipe *pipe) +{ + int i; + struct skl_pipe_fmt *cur_fmt; + struct skl_pipe_fmt *next_fmt; + + if (pipe->nr_cfgs <= 1) + return false; + + if (pipe->conn_type != SKL_PIPE_CONN_TYPE_FE) + return true; + + for (i = 0; i < pipe->nr_cfgs - 1; i++) { + if (pipe->direction == SNDRV_PCM_STREAM_PLAYBACK) { + cur_fmt = &pipe->configs[i].out_fmt; + next_fmt = &pipe->configs[i+1].out_fmt; + } else { + cur_fmt = &pipe->configs[i].in_fmt; + next_fmt = &pipe->configs[i+1].in_fmt; + } + + if (!CHECK_HW_PARAMS(cur_fmt->channels, cur_fmt->freq, + cur_fmt->bps, + next_fmt->channels, + next_fmt->freq, + next_fmt->bps)) + return true; + } + + return false; +} + /* * Here, we select pipe format based on the pipe type and pipe * direction to determine the current config index for the pipeline. @@ -600,6 +632,14 @@ skl_tplg_get_pipe_config(struct skl_dev *skl, struct skl_module_cfg *mconfig) return 0; } + if (skl_tplg_is_multi_fmt(skl, pipe)) { + pipe->cur_config_idx = pipe->pipe_config_idx; + pipe->memory_pages = pconfig->mem_pages; + dev_dbg(skl->dev, "found pipe config idx:%d\n", + pipe->cur_config_idx); + return 0; + } + if (pipe->conn_type == SKL_PIPE_CONN_TYPE_NONE) { dev_dbg(skl->dev, "No conn_type detected, take 0th config\n"); pipe->cur_config_idx = 0; @@ -1314,6 +1354,71 @@ static int skl_tplg_pga_event(struct snd_soc_dapm_widget *w, return 0; } +static int skl_tplg_multi_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct hdac_bus *bus = snd_soc_component_get_drvdata(component); + struct skl_dev *skl = bus_to_skl(bus); + struct skl_pipeline *ppl; + struct skl_pipe *pipe = NULL; + u32 *pipe_id; + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + + if (!ec) + return -EINVAL; + + pipe_id = ec->dobj.private; + + list_for_each_entry(ppl, &skl->ppl_list, node) { + if (ppl->pipe->ppl_id == *pipe_id) { + pipe = ppl->pipe; + break; + } + } + if (!pipe) + return -EIO; + + ucontrol->value.enumerated.item[0] = pipe->pipe_config_idx; + + return 0; +} + +static int skl_tplg_multi_config_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct hdac_bus *bus = snd_soc_component_get_drvdata(component); + struct skl_dev *skl = bus_to_skl(bus); + struct skl_pipeline *ppl; + struct skl_pipe *pipe = NULL; + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + u32 *pipe_id; + + if (!ec) + return -EINVAL; + + if (ucontrol->value.enumerated.item[0] > ec->items) + return -EINVAL; + + pipe_id = ec->dobj.private; + + list_for_each_entry(ppl, &skl->ppl_list, node) { + if (ppl->pipe->ppl_id == *pipe_id) { + pipe = ppl->pipe; + break; + } + } + if (!pipe) + return -EIO; + + pipe->pipe_config_idx = ucontrol->value.enumerated.item[0]; + + return 0; +} + static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol, unsigned int __user *data, unsigned int size) { @@ -1853,6 +1958,11 @@ static const struct snd_soc_tplg_kcontrol_ops skl_tplg_kcontrol_ops[] = { .get = skl_tplg_mic_control_get, .put = skl_tplg_mic_control_set, }, + { + .id = SKL_CONTROL_TYPE_MULTI_IO_SELECT, + .get = skl_tplg_multi_config_get, + .put = skl_tplg_multi_config_set, + }, }; static int skl_tplg_fill_pipe_cfg(struct device *dev, diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index e967800dbb62..06576147cc29 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -306,6 +306,7 @@ struct skl_pipe { struct skl_path_config configs[SKL_MAX_PATH_CONFIGS]; struct list_head w_list; bool passthru; + u32 pipe_config_idx; }; enum skl_module_state { -- 2.17.1

2 1

[PATCH] ASoC: Intel: Skylake: Add alternative topology binary name
by Mateusz Gorski 02 Apr '20

02 Apr '20

This commit adds alternative topology binary file name based on used machine driver and fallback to use this name after failed attempt to load topology file with name based on NHLT. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- sound/soc/intel/skylake/skl-topology.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 69cd7a81bf2a..344b06df0e15 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3565,8 +3565,20 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", - skl->tplg_name, ret); + char alt_tplg_name[64]; + + snprintf(alt_tplg_name, sizeof(alt_tplg_name), "%s-tplg.bin", + skl->i2s_dev->name); + dev_info(bus->dev, "tplg fw %s load failed with %d, trying alternative tplg name %s", + skl->tplg_name, ret, alt_tplg_name); + + ret = request_firmware(&fw, alt_tplg_name, bus->dev); + if (!ret) + goto component_load; + + dev_info(bus->dev, "tplg %s failed with %d, falling back to dfw_sst.bin", + alt_tplg_name, ret); + ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { dev_err(bus->dev, "Fallback tplg fw %s load failed with %d\n", @@ -3575,6 +3587,9 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) } } +component_load: + + /* * The complete tplg for SKL is loaded as index 0, we don't use * any other index -- 2.17.1

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[PATCH] ALSA:atmel:Use platform_get_irq_optional()
by Tang Bin 02 Apr '20

02 Apr '20

In order to simply code,because platform_get_irq() already has dev_err() message. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/atmel/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index a1dce9725..25cfd5710 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -715,7 +715,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - irq = platform_get_irq(pdev, 0); + irq = platform_get_irq_optional(pdev, 0); if (irq < 0) { dev_dbg(&pdev->dev, "could not get irq: %d\n", irq); return irq; -- 2.20.1.windows.1

2 1

[PATCH] ASoC: dpcm: allow start or stop during pause for backend
by 이경택 01 Apr '20

01 Apr '20

soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. So, start or stop after pause in compress offload is always returned as error if the compress offload is used with dpcm. To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca0..8f6f0ad50288 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2236,7 +2236,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2266,7 +2267,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.21.0

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