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

[alsa-devel] [PATCH 0/5] ALSA: hda/realtek - Comprehensive model list
by Takashi Iwai 26 Jun '18

26 Jun '18

Hi, this is a patch set to fill the missing entries for Realtek HD-audio codec quirk model strings. Takashi === Takashi Iwai (5): ALSA: hda/realtek - Comprehensive model list for ALC662 & co ALSA: hda/realtek - Comprehensive model list for ALC259 & co ALSA: hda/realtek - Comprehensive model list for ALC268 ALSA: hda/realtek - Comprehensive model list for ALC262 ALSA: hda/realtek - Comprehensive model list for ALC882 & co Documentation/sound/hd-audio/models.rst | 264 +++++++++++++++++++++++- sound/pci/hda/patch_realtek.c | 134 ++++++++++++ 2 files changed, 396 insertions(+), 2 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] ASoC: soc-utils: Fix unregistration order
by Fabio Estevam 26 Jun '18

26 Jun '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> The unregistration should happen in the opposite order of the registration, so change it accordingly. No real issue has been noticed, but it is good practice to keep the correct unregistration order. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/soc-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 2d9e98b..a863bb3 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -381,6 +381,6 @@ int __init snd_soc_util_init(void) void __exit snd_soc_util_exit(void) { - platform_device_unregister(soc_dummy_dev); platform_driver_unregister(&soc_dummy_driver); + platform_device_unregister(soc_dummy_dev); } -- 2.7.4

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[alsa-devel] [PATCH 00/11] ASoC: qdsp6: Fix dt based module loading
by Srinivas Kandagatla 26 Jun '18

26 Jun '18

Recent discussion on https://lkml.org/lkml/2018/6/18/1282 suggested that qdsp6 code is using less common of_platform apis which would not give dt based module loading support. Suggestion from Rob was to add compatible strings to the dais nodes This patchset adds this. Also this patchset has 2 trivial fixes which is removing redundant owner and fix a error check in q6routing. Thanks, srini Srinivas Kandagatla (11): ASoC: q6adm: dt-bindings: add compatible string to routing ASoC: q6asm: dt-bindings: add compatible string to dais ASoC: q6afe: dt-bindings: add compatible string to dais ASoC: qdsp6: q6adm: use of_platform_populate/depopulate() ASoC: qdsp6: q6asm: use of_platform_populate/depopulate() ASoC: qdsp6: q6afe: use of_platform_populate/depopulate() ASoC: qdsp6: q6afe-dai: support dt based module loading ASoC: qdsp6: q6asm-dai: support dt based module loading ASoC: qdsp6: q6routing: support dt based module loading ASoC: qcom: apq8096: remove redundant owner assignment ASoC: qdsp6: q6routing: add proper error check Documentation/devicetree/bindings/sound/qcom,q6adm.txt | 6 ++++++ Documentation/devicetree/bindings/sound/qcom,q6afe.txt | 6 ++++++ Documentation/devicetree/bindings/sound/qcom,q6asm.txt | 6 ++++++ sound/soc/qcom/apq8096.c | 1 - sound/soc/qcom/qdsp6/q6adm.c | 15 ++------------- sound/soc/qcom/qdsp6/q6afe-dai.c | 8 ++++++++ sound/soc/qcom/qdsp6/q6afe.c | 16 ++-------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 8 ++++++++ sound/soc/qcom/qdsp6/q6asm.c | 17 ++--------------- sound/soc/qcom/qdsp6/q6routing.c | 10 +++++++++- 10 files changed, 49 insertions(+), 44 deletions(-) -- 2.16.2

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[alsa-devel] [PATCH v2 00/12] ASoC: qdsp6: Fix dt based module loading and minor fixes
by Srinivas Kandagatla 26 Jun '18

26 Jun '18

Recent discussion on https://lkml.org/lkml/2018/6/18/1282 suggested that qdsp6 code is using less common of_platform apis which would not give dt based module loading support. Suggestion from Rob was to add compatible strings to the dais nodes This patchset adds this. Also this patchset has 3 trivial fixes which is removing redundant owner, remove unused struct member and fix a error check in q6routing. Changes since v1: - Addressed few trivial comments from Niklas Thanks, srini Srinivas Kandagatla (12): ASoC: q6adm: dt-bindings: add compatible string to routing ASoC: q6asm: dt-bindings: add compatible string to dais ASoC: q6afe: dt-bindings: add compatible string to dais ASoC: qdsp6: q6adm: use of_platform_populate/depopulate() ASoC: qdsp6: q6asm: use of_platform_populate/depopulate() ASoC: qdsp6: q6afe: use of_platform_populate/depopulate() ASoC: qdsp6: q6afe-dai: support dt based module loading ASoC: qdsp6: q6asm-dai: support dt based module loading ASoC: qdsp6: q6routing: support dt based module loading ASoC: qcom: apq8096: remove redundant owner assignment ASoC: qdsp6: q6routing: add proper error check ASoC: qdsp6: q6asm: remove unused struct q6asm member Documentation/devicetree/bindings/sound/qcom,q6adm.txt | 6 ++++++ Documentation/devicetree/bindings/sound/qcom,q6afe.txt | 6 ++++++ Documentation/devicetree/bindings/sound/qcom,q6asm.txt | 6 ++++++ sound/soc/qcom/apq8096.c | 1 - sound/soc/qcom/qdsp6/q6adm.c | 16 ++-------------- sound/soc/qcom/qdsp6/q6afe-dai.c | 7 +++++++ sound/soc/qcom/qdsp6/q6afe.c | 16 ++-------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 7 +++++++ sound/soc/qcom/qdsp6/q6asm.c | 17 ++--------------- sound/soc/qcom/qdsp6/q6routing.c | 9 ++++++++- 10 files changed, 46 insertions(+), 45 deletions(-) -- 2.16.2

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[alsa-devel] [PATCH 1/1] ASoC: soc-pcm: DPCM cares BE channel constraint
by jiada_wang＠mentor.com 26 Jun '18

26 Jun '18

From: Jiada Wang <jiada_wang(a)mentor.com> Current DPCM is caring only FE channel configuration. Sometimes it will be trouble if user selects channel which isn't supported by BE. This patch adds new .dpcm_merged_chan on struct snd_soc_dai_link. DPCM will use FE / BE merged channel if struct snd_soc_dai_link has it. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-pcm.c | 46 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 48 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1378dcd2128a..f7579f82cc00 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -957,6 +957,8 @@ struct snd_soc_dai_link { /* DPCM used FE & BE merged format */ unsigned int dpcm_merged_format:1; + /* DPCM used FE & BE merged channel */ + unsigned int dpcm_merged_chan:1; /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 45b52f7b9690..19ebfc958b9d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1715,6 +1715,46 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) return formats; } +static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, + unsigned int *channels_min, + unsigned int *channels_max) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + struct snd_soc_dpcm *dpcm; + int stream = substream->stream; + + if (!fe->dai_link->dpcm_merged_chan) + return; + + *channels_min = 0; + *channels_max = UINT_MAX; + + /* + * It returns merged BE codec channel; + * if FE want to use it (= dpcm_merged_chan) + */ + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + int i; + + for (i = 0; i < be->num_codecs; i++) { + codec_dai_drv = be->codec_dais[i]->driver; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + + *channels_min = max(*channels_min, + codec_stream->channels_min); + *channels_max = min(*channels_max, + codec_stream->channels_max); + } + } +} + static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1722,11 +1762,17 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; u64 format = dpcm_runtime_base_format(substream); + unsigned int channels_min = 0, channels_max = UINT_MAX; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format); else dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format); + + dpcm_runtime_base_chan(substream, &channels_min, &channels_max); + + runtime->hw.channels_min = max(channels_min, runtime->hw.channels_min); + runtime->hw.channels_max = min(channels_max, runtime->hw.channels_max); } static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); -- 2.17.0

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[alsa-devel] [PATCH 1/2] ASoC: add generic simple-amplifier support
by Nicolò Veronese 26 Jun '18

26 Jun '18

Many boards have speaker amplifier attached on an analog output. Most of them are only managed by a mute or shutdown GPIO. In order to reduce duplicate driver, this patch adds generic/simple-amplifier. Signed-off-by: Nicolò Veronese <nicveronese(a)gmail.com> --- sound/soc/generic/Kconfig | 5 ++ sound/soc/generic/Makefile | 2 + sound/soc/generic/simple-amplifier.c | 109 +++++++++++++++++++++++++++++++++++ 3 files changed, 116 insertions(+) create mode 100644 sound/soc/generic/simple-amplifier.c diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index c954be0..ace17c3a 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -7,6 +7,11 @@ config SND_SIMPLE_CARD help This option enables generic simple sound card support +config SND_SIMPLE_AMPLIFIER + tristate "ASoC Simple Amplifier support" + help + This option enables generic simple amplifier support + config SND_SIMPLE_SCU_CARD tristate "ASoC Simple SCU sound card support" depends on OF diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9dec293..805a746 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,12 +1,14 @@ # SPDX-License-Identifier: GPL-2.0 snd-soc-simple-card-utils-objs := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o +snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-simple-scu-card-objs := simple-scu-card.o snd-soc-audio-graph-card-objs := audio-graph-card.o snd-soc-audio-graph-scu-card-objs := audio-graph-scu-card.o obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o +obj-$(CONFIG_SND_SIMPLE_AMPLIFIER) += snd-soc-simple-amplifier.o obj-$(CONFIG_SND_SIMPLE_SCU_CARD) += snd-soc-simple-scu-card.o obj-$(CONFIG_SND_AUDIO_GRAPH_CARD) += snd-soc-audio-graph-card.o obj-$(CONFIG_SND_AUDIO_GRAPH_SCU_CARD) += snd-soc-audio-graph-scu-card.o diff --git a/sound/soc/generic/simple-amplifier.c b/sound/soc/generic/simple-amplifier.c new file mode 100644 index 0000000..e135a43 --- /dev/null +++ b/sound/soc/generic/simple-amplifier.c @@ -0,0 +1,109 @@ +// SPDX-License-Identifier: GPL-2.0 +// ALSA SoC simple amplifier driver +// +// Copyright 2018 +// Author: Nicolò Veronese <nicveronese(a)gmail.com> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio/consumer.h> +#include <linux/regulator/consumer.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +/* This struct is used to save the context */ +struct simple_amp_data { + struct device *dev; + struct regmap *regmap; + struct gpio_desc *power_gpio; +}; + +static int simple_amp_power_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kctrl, int event) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(widget->dapm); + struct simple_amp_data *data = snd_soc_component_get_drvdata(c); + + /* Shutdown GPIO */ + if (data->power_gpio) + gpiod_set_value_cansleep(data->power_gpio, SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LEFT AMP IN"), + SND_SOC_DAPM_INPUT("RIGHT AMP IN"), + + SND_SOC_DAPM_OUTPUT("SPK LEFT"), + SND_SOC_DAPM_OUTPUT("SPK RIGHT"), + + SND_SOC_DAPM_REGULATOR_SUPPLY("VCC", 20, 0), + SND_SOC_DAPM_SPK("MUTE", simple_amp_power_event), +}; + +static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = { + { "MUTE", NULL, "LEFT AMP IN" }, + { "MUTE", NULL, "RIGHT AMP IN" }, + + { "SPK LEFT", NULL, "VCC" }, + { "SPK RIGHT", NULL, "VCC" }, + + { "SPK LEFT", NULL, "MUTE" }, + { "SPK RIGHT", NULL, "MUTE" }, +}; + +static const struct snd_soc_component_driver simple_amp_component_driver = { + .name = "simple-amplifier", + .dapm_widgets = simple_amp_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(simple_amp_dapm_widgets), + .dapm_routes = simple_amp_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(simple_amp_dapm_routes), +}; + +static int asoc_simple_amp_probe(struct platform_device *pdev) +{ + struct simple_amp_data *data; + struct device *dev = &pdev->dev; + int ret; + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + data->dev = &pdev->dev; + platform_set_drvdata(pdev, data); + + if(of_property_read_bool(pdev->dev.of_node,"shutdown-gpios")) { + data->power_gpio = devm_gpiod_get_optional(&pdev->dev, "shutdown", GPIOD_OUT_LOW); + + if (IS_ERR(data->power_gpio)) { + ret = PTR_ERR(data->power_gpio); + dev_err(&pdev->dev, "Failed to get shutdown gpio: %d\n", ret); + return ret; + } + } + + return devm_snd_soc_register_component(dev, + &simple_amp_component_driver, NULL, 0); +} + +static const struct of_device_id asoc_simple_of_match[] = { + { .compatible = "simple-audio-amplifier", }, + {}, +}; +MODULE_DEVICE_TABLE(of, asoc_simple_of_match); + +static struct platform_driver asoc_simple_amp = { + .driver = { + .name = "asoc-simple-amplifier", + .of_match_table = of_match_ptr(asoc_simple_of_match), + }, + .probe = asoc_simple_amp_probe, +}; + +module_platform_driver(asoc_simple_amp); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("ASoC Simple Amplifier"); +MODULE_AUTHOR("Nicolò Veronese <nicveronese(a)gmail.com>"); -- 2.7.4

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[alsa-devel] [PATCH v6 0/3] ASoC: Add Multi CPU DAI support
by Shreyas NC 26 Jun '18

26 Jun '18

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v6: - Addressed review comments from Pierre in calculating delay for CPU DAI, add check for compress DAIs and other nits - add proper exit check in dpcm_prune_path() for CPU DAIs - rebased on latest for-next branch Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 6 + sound/soc/soc-core.c | 290 +++++++++++++++++++++--------- sound/soc/soc-dapm.c | 71 +++++--- sound/soc/soc-pcm.c | 491 ++++++++++++++++++++++++++++++++++----------------- 4 files changed, 589 insertions(+), 269 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 0/2] slimbus: Add Stream Support
by Srinivas Kandagatla 25 Jun '18

25 Jun '18

This patchset adds basic stream support for SLIMbus devices and controllers. Mostly inspired by soundwire stream patches. But slimbus stream is much simpler compared to soundwire >From slim_device side, we have below 6 new apis. slim_stream_allocate() - allocating runtime slim stream slim_stream_prepare() - to configure runtime stream with config slim_stream_enable() - enable channels/ports for data slim_stream_disable() - disable channels/ports. slim_stream_unprepare() - un configure runtime stream slim_stream_free() - free all associated memory. >From Controller side: Data channel Management and reconfiguration messages are applicable for all the controllers which are inline with SLIMbus specs. However light weight controller like NGD which have user specific implementation of some messages need to be invoked instead of standard message commands. For usecases like this simple enable/disable stream apis are provided. Assumptions: 1> Current design assumes that the channel and ports are statically allocated to the device during SoC integration, which is the case with all the Qualcomm SoCs. 2> One-to-One mapping between Port and Channel, SLIMBus versions earlier than 2.0 has only one endpoint per port. Current patchset can be extended to support more than one endpoints per port. 3> Only audio usecase, This patchset only supports Isochronous and Push/Pull transport protocols, which are sufficient for audio use cases. 4> DSP does all the data handling for the allocated channels. Which is true for Qcom SoCs. TODO: Bandwidth management. Dependency: This patchset has dependency on the NGD driver https://patchwork.kernel.org/patch/10474959/ Tested this patchset with WCD9335 codec playback on DB820c on top of 4.18-rc1 with qdsp6. I have pushed my working branch to [1] incase someone want to try. [1]:https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=slimb… Thanks, srini Srinivas Kandagatla (2): slimbus: stream: add stream support slimbus: ngd: add stream support Documentation/driver-api/slimbus.rst | 5 + drivers/slimbus/Makefile | 2 +- drivers/slimbus/core.c | 2 + drivers/slimbus/qcom-ngd-ctrl.c | 144 +++++++++- drivers/slimbus/slimbus.h | 206 +++++++++++++++ drivers/slimbus/stream.c | 493 +++++++++++++++++++++++++++++++++++ include/linux/slimbus.h | 56 ++++ 7 files changed, 905 insertions(+), 3 deletions(-) create mode 100644 drivers/slimbus/stream.c -- 2.16.2

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[alsa-devel] first and last audio sample timestamp
by Subhashini Rao Beerisetty 25 Jun '18

25 Jun '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, Hello all, I’m using aplay and arecord utilities for playback and capture. I need to capture the timestamps for the first and last audio samples that arrive at the driver level. For this I’m capturing the CLOCK_MONOTONIC_RAW timestamp at the .trigger(for playback & capture) callback in SNDRV_PCM_TRIGGER_START case – is it the correct timestamp for the first audio sample arrived at the driver level? Similarly does .trigger callbacks SNDRV_PCM_TRIGGER_STOP gives the last audio sample timestamp?

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[alsa-devel] [PATCH] ASoC: dpcm: improve runtime update predictability
by Jerome Brunet 25 Jun '18

25 Jun '18

As it is, dpcm_runtime_update() performs the old path and new path update of a frontend before going on to the next frontend DAI. Depending the order of the FEs within the rtd list, the result of the update might be different. For example: * Frontend A connected to backend C, with a 48kHz playback * Frontend B connected to backend D, with a 44.1kHz playback * FE A appears before FE B in the rtd list of the card. If we reparent BE C to FE B (disconnecting BE D): * old path update of FE A will run first, and BE C will get hw_free() and shutdown() * new path update of FE B will run after and BE C, which is stopped, so it will be configured at 44.1kHz, as expected If we reparent BE D to FE A (disconnecting BE C): * new path update of FE A will run first but since BE D is still running at 44.1kHz, it won't be reconfigured (no call to startup() or hw_params()) * old path update of FE B runs after, nothing happens * In this case, we end up with a BE playing at 44.1kHz a stream which is supposed to be played at 48Khz (too slow) To improve this situation, this patch performs all the FE old paths update before going on to update the new paths. With this, the result should no longer depend on the order of the FE within the card rtd list. Please note that there might be a small performance penalty since dpcm_process_paths() is called twice per stream direction. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-pcm.c | 165 +++++++++++++++++++++++++++------------------------- 1 file changed, 86 insertions(+), 79 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5e7ae47a9658..5c2534414478 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2543,106 +2543,113 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) return ret; } -/* Called by DAPM mixer/mux changes to update audio routing between PCMs and - * any DAI links. - */ -int soc_dpcm_runtime_update(struct snd_soc_card *card) +int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { - struct snd_soc_pcm_runtime *fe; - int old, new, paths; + struct snd_soc_dapm_widget_list *list; + int count, paths; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - list_for_each_entry(fe, &card->rtd_list, list) { - struct snd_soc_dapm_widget_list *list; + if (!fe->dai_link->dynamic) + return 0; - /* make sure link is FE */ - if (!fe->dai_link->dynamic) - continue; + /* only check active links */ + if (!fe->cpu_dai->active) + return 0; - /* only check active links */ - if (!fe->cpu_dai->active) - continue; + /* DAPM sync will call this to update DSP paths */ + dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", + new ? "new" : "old", fe->dai_link->name); - /* DAPM sync will call this to update DSP paths */ - dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n", - fe->dai_link->name); + /* skip if FE doesn't have playback capability */ + if (!fe->cpu_dai->driver->playback.channels_min || + !fe->codec_dai->driver->playback.channels_min) + goto capture; - /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min - || !fe->codec_dai->driver->playback.channels_min) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active - || !fe->codec_dai->playback_active) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - mutex_unlock(&card->mutex); - return paths; - } + /* skip if FE isn't currently playing */ + if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) + goto capture; - /* update any new playback paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1); - if (new) { - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "playback"); + return paths; + } - /* update any old playback paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0); - if (old) { + /* update any playback paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + + dpcm_path_put(&list); + capture: - /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min - || !fe->codec_dai->driver->capture.channels_min) - continue; + /* skip if FE doesn't have capture capability */ + if (!fe->cpu_dai->driver->capture.channels_min || + !fe->codec_dai->driver->capture.channels_min) + return 0; - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active - || !fe->codec_dai->capture_active) - continue; + /* skip if FE isn't currently capturing */ + if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) + return 0; - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - mutex_unlock(&card->mutex); - return paths; - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "capture"); + return paths; + } - /* update any new capture paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1); - if (new) { + /* update any old capture paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); + if (count) { + if (new) dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - - /* update any old capture paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0); - if (old) { + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); } - mutex_unlock(&card->mutex); + dpcm_path_put(&list); + return 0; } + +/* Called by DAPM mixer/mux changes to update audio routing between PCMs and + * any DAI links. + */ +int soc_dpcm_runtime_update(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *fe; + int ret = 0; + + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* shutdown all old paths first */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 0); + if (ret) + goto out; + } + + /* bring new paths up */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 1); + if (ret) + goto out; + } + +out: + mutex_unlock(&card->mutex); + return ret; +} int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) { struct snd_soc_dpcm *dpcm; -- 2.14.4

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