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

[alsa-devel] [PATCH v15 0/8] ASoC: sun4i-codec: Add Line-In, FM-In, Mic 2
by Danny Milosavljevic 07 Feb '19

07 Feb '19

This patchset adds some mixer controls to sun4i-codec for the Allwinner A10 and the Allwinner A20. It also adds the PGA for the MIC2 preamp. Where possible, it uses SOC_DAPM_DOUBLE in order to cut down on the number of distinct controls in alsamixer. v15 changes compared to v14 are: - Instead of adding controls dynamically, add an extra sun7i_codec_controls structure and duplicate the sun4i_codec_controls there, with small changes. - Split Mic Playback Volume into extra patch. - LEFT OFF adding Capture Selects since it's unclear how we want to structure it. - LEFT OFF adding Differential Line Source since it might also change if the Capture Select implementation is different. v14 changes compared to v13 are: - Merged some of the patches together if it made sense to test them together. - Use snd_soc_component_driver. - Moved SUN4I_CODEC_DAC_ACTL_LFMS, SUN4I_CODEC_DAC_ACTL_RFMS to the correct patch in the series. - Kept "Left Mixer Left DAC Playback Switch", "Right Mixer Left DAC Playback Switch" and "Right Mixer Right DAC Playback Switch" unchanged compared to the released version - for backward compatibility. v13 changes compared to v12 are: - Added my "Signed-off-by". - Clarified some commit message text. v12 changes compared to v11 are: - Split up patchset in another way. - Renamed "Mic1 Capture Volume" to "Mic1 Boost Volume". - Renamed "Mic2 Capture Volume" to "Mic2 Boost Volume". - Renamed "Line Capture Volume" to "Line Boost Volume". - Renamed "Differential Line Capture Switch" to "Differential Line Source". v11 changes compared to v10 are: - Split up patchset. - Fixed typo in Differential Line Capture Switch. - Renamed "Non-Differential" value to "Stereo". - Removed duplicate PA Volume mixer control. v10 changes compared to v9 are: - Use SOC_DAPM_DOUBLE where possible and it makes sense in order to cut down on the number of controls. v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (8): ASoC: sun4i-codec: Add MIC2 Pre-Amplifier, Mic2 ASoC: sun4i-codec: Add Mic Playback Volume ASoC: sun4i-codec: Add sun7i_codec_controls, sun7i_codec_codec. ASoC: sun4i-codec: Add Mic1 Boost Volume, Mic2 Boost Volume ASoC: sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls ASoC: sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch ASoC: sun4i-codec: Add FM Playback Volume, FM Left, FM Right, FM Playback Switch ASoC: sun4i-codec: Add Line Playback Volume, Line Boost Volume, Line Right, Line Left, Line Playback Switch sound/soc/sunxi/sun4i-codec.c | 149 ++++++++++++++++++++++++++++++---- 1 file changed, 134 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH v10 0/4] Media Device Allocator API
by Shuah Khan 06 Feb '19

06 Feb '19

Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - This patch series is tested on 5.0-rc3 and addresses comments on v9 series from Hans Verkuil. - v9 was tested on 4.20-rc6. - Tested sharing resources with kaffeine, vlc, xawtv, tvtime, and arecord. When analog is streaming, digital and audio user-space applications detect that the tuner is busy and exit. When digital is streaming, analog and audio applications detect that the tuner is busy and exit. When arecord is owns the tuner, digital and analog detect that the tuner is busy and exit. - Tested media device allocator API with bind/unbind testing on snd-usb-audio and au0828 drivers to make sure /dev/mediaX is released only when the last driver is unbound. - Addressed review comments from Hans on the RFC v8 (rebased on 4.19) - Updated change log to describe the use-case more clearly. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. Changes since v9: - Patch 1: Fix mutex assert warning from find_module() calls. This code was written before the change to find_module() that requires callers to hold module_mutex. I missed this during my testing on 4.20-rc6. Hans Verkuil reported the problem. - Patch 4: sound/usb: Initializes all the entities it can before registering the device based on comments from Hans Verkuil - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - No changes to Patches 2 and 3. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (4): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 41 ++++ drivers/media/Makefile | 4 + drivers/media/media-dev-allocator.c | 144 +++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 327 +++++++++++++++++++++++++ sound/usb/media.h | 74 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 723 insertions(+), 22 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.1

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[alsa-devel] [PATCH v2] ALSA: hda/tegra: enable clock during probe
by Sameer Pujar 05 Feb '19

05 Feb '19

If CONFIG_PM is disabled or runtime PM calls are forbidden, the clocks will not be ON. This could cause issue during probe, where hda init setup is done. This patch enables clocks unconditionally during probe. Along with above, follwoing changes are done. * enable runtime PM before exiting from probe work. This helps to avoid usage of pm_runtime_get_sync/pm_runtime_put() in probe work. * hda_tegra_disable_clocks() is moved out of CONFIG_PM_SLEEP check. * runtime PM callbacks moved out of CONFIG_PM check Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Reviewed-by: Ravindra Lokhande <rlokhande(a)nvidia.com> Reviewed-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 26 +++++++++++++++++--------- 1 file changed, 17 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index c8d18dc..ba6175f 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -219,7 +219,6 @@ static int hda_tegra_enable_clocks(struct hda_tegra *data) return rc; } -#ifdef CONFIG_PM_SLEEP static void hda_tegra_disable_clocks(struct hda_tegra *data) { clk_disable_unprepare(data->hda2hdmi_clk); @@ -227,6 +226,7 @@ static void hda_tegra_disable_clocks(struct hda_tegra *data) clk_disable_unprepare(data->hda_clk); } +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -257,7 +257,6 @@ static int hda_tegra_resume(struct device *dev) } #endif /* CONFIG_PM_SLEEP */ -#ifdef CONFIG_PM static int hda_tegra_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -283,7 +282,7 @@ static int hda_tegra_runtime_resume(struct device *dev) int rc; rc = hda_tegra_enable_clocks(hda); - if (rc != 0) + if (rc) return rc; if (chip && chip->running) { hda_tegra_init(hda); @@ -292,7 +291,6 @@ static int hda_tegra_runtime_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops hda_tegra_pm = { SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) @@ -551,9 +549,9 @@ static int hda_tegra_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, card); - pm_runtime_enable(hda->dev); - if (!azx_has_pm_runtime(chip)) - pm_runtime_forbid(hda->dev); + err = hda_tegra_enable_clocks(hda); + if (err) + goto out_free; schedule_work(&hda->probe_work); @@ -571,7 +569,6 @@ static void hda_tegra_probe_work(struct work_struct *work) struct platform_device *pdev = to_platform_device(hda->dev); int err; - pm_runtime_get_sync(hda->dev); err = hda_tegra_first_init(chip, pdev); if (err < 0) goto out_free; @@ -592,8 +589,15 @@ static void hda_tegra_probe_work(struct work_struct *work) chip->running = 1; snd_hda_set_power_save(&chip->bus, power_save * 1000); + /* set device state as active */ + if (pm_runtime_set_active(hda->dev) < 0) + goto out_free; + /* enable runtime PM */ + pm_runtime_enable(hda->dev); + if (!azx_has_pm_runtime(chip)) + pm_runtime_forbid(hda->dev); + out_free: - pm_runtime_put(hda->dev); return; /* no error return from async probe */ } @@ -603,6 +607,10 @@ static int hda_tegra_remove(struct platform_device *pdev) ret = snd_card_free(dev_get_drvdata(&pdev->dev)); pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) { + hda_tegra_runtime_suspend(&pdev->dev); + pm_runtime_set_suspended(&pdev->dev); + } return ret; } -- 2.7.4

7 29

[alsa-devel] [PATCH] ASoC: soc-core: clear platform pointers on error
by Curtis Malainey 04 Feb '19

04 Feb '19

Originally snd_soc_init_platform was not cleaning up its pointers, this was fixed to always reallocate dynamic memory but created a memory leak when snd_soc_init_platform was called multiple times during the same probe attempt and also threw away any changes made to the struct between calls. In order to avoid reallocating memory that is still valid, the behaviour will be changed to clear the dynamically set pointers on a probe error and a unregister event and snd_soc_init_platform will go back to its original behaviour of only allocating null pointers so it will stop throwing away valid changes. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/soc-core.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d6b5edba8d2d..9431dd605103 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1053,7 +1053,7 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * soc.h :: struct snd_soc_dai_link */ /* convert Legacy platform link */ - if (!platform || dai_link->legacy_platform) { + if (!platform) { platform = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); @@ -1076,6 +1076,24 @@ static int snd_soc_init_platform(struct snd_soc_card *card, return 0; } +static void soc_cleanup_platform(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link; + int i; + /* + * FIXME + * + * this function should be removed with snd_soc_init_platform + */ + + for_each_card_prelinks(card, i, link) { + if (link->legacy_platform) { + link->legacy_platform = 0; + link->platforms = NULL; + } + } +} + static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -2020,6 +2038,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove and free each DAI */ soc_remove_dai_links(card); soc_remove_pcm_runtimes(card); + soc_cleanup_platform(card); /* remove auxiliary devices */ soc_remove_aux_devices(card); -- 2.20.1.495.gaa96b0ce6b-goog

2 1

[alsa-devel] [PATCH 1/2] ASoC: dapm: Only power up active channels from a DAI
by Charles Keepax 02 Feb '19

02 Feb '19

Currently all widgets attached to a DAI link will be powered up when the DAI is active, however this may include routes that are not actually in use if there are unused channels available on the DAI. The macros for creating AIF widgets already include an entry for slot, it is proposed to change that to channel. The effective difference here being respresenting the logical channel index rather than the physical slot index. The CODECs currently using the slot entry on the DAPM_AIF macros are using it in a manner consistent with this, the CODECs not using it just have the field set to zero. A variable is added to snd_soc_dapm_widget to represent this channel index and then for each AIF widget attached to a DAI this is compared against the number of channels on the stream. Enabling the links for those which will be in use. This has the nice property that the CODECs which haven't used the slot/channel entry in the macro will function exactly as before due to all the AIF widgets having a channel of zero and a stream by definition having at least one channel. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc-dapm.h | 22 +++++++++----- sound/soc/soc-dapm.c | 76 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-pcm.c | 4 +++ 3 files changed, 94 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 46f2ba3ffcb7c..79b4ddfb8e9e2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -214,21 +214,21 @@ struct device; .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} /* stream domain */ -#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \ +#define SND_SOC_DAPM_AIF_IN(wname, stname, wchan, wreg, wshift, winvert) \ { .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } -#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wslot, wreg, wshift, winvert, \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } +#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wchan, wreg, wshift, winvert, \ wevent, wflags) \ { .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .event = wevent, .event_flags = wflags } -#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \ +#define SND_SOC_DAPM_AIF_OUT(wname, stname, wchan, wreg, wshift, winvert) \ { .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } -#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wslot, wreg, wshift, winvert, \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), } +#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wchan, wreg, wshift, winvert, \ wevent, wflags) \ { .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ - SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .event = wevent, .event_flags = wflags } #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, \ @@ -407,6 +407,10 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); +int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); + /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_card *card); void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm); @@ -627,6 +631,8 @@ struct snd_soc_dapm_widget { int endpoints[2]; struct clk *clk; + + int channel; }; struct snd_soc_dapm_update { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c4c134195392..1714f612451e1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2525,6 +2525,78 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); +static int dapm_update_dai_chan(struct snd_soc_dapm_path *p, + struct snd_soc_dapm_widget *w, + int channels) +{ + switch (w->id) { + case snd_soc_dapm_aif_out: + case snd_soc_dapm_aif_in: + break; + default: + return 0; + } + + dev_dbg(w->dapm->dev, "%s DAI route %s -> %s\n", + w->channel < channels ? "Connecting" : "Disconnecting", + p->source->name, p->sink->name); + + if (w->channel < channels) + soc_dapm_connect_path(p, true, "dai update"); + else + soc_dapm_connect_path(p, false, "dai update"); + + return 0; +} + +static int dapm_update_dai_unlocked(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int dir = substream->stream; + int channels = params_channels(params); + struct snd_soc_dapm_path *p; + struct snd_soc_dapm_widget *w; + int ret; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) + w = dai->playback_widget; + else + w = dai->capture_widget; + + dev_dbg(dai->dev, "Update DAI routes for %s %s\n", dai->name, + dir == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); + + snd_soc_dapm_widget_for_each_sink_path(w, p) { + ret = dapm_update_dai_chan(p, p->sink, channels); + if (ret < 0) + return ret; + } + + snd_soc_dapm_widget_for_each_source_path(w, p) { + ret = dapm_update_dai_chan(p, p->source, channels); + if (ret < 0) + return ret; + } + + return 0; +} + +int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret; + + mutex_lock_nested(&rtd->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = dapm_update_dai_unlocked(substream, params, dai); + mutex_unlock(&rtd->card->dapm_mutex); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_update_dai); + /* * dapm_update_widget_flags() - Re-compute widget sink and source flags * @w: The widget for which to update the flags @@ -3689,6 +3761,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = soc_dai_hw_params(&substream, params, source); if (ret < 0) goto out; + + dapm_update_dai_unlocked(&substream, params, source); } substream.stream = SNDRV_PCM_STREAM_PLAYBACK; @@ -3709,6 +3783,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = soc_dai_hw_params(&substream, params, sink); if (ret < 0) goto out; + + dapm_update_dai_unlocked(&substream, params, sink); } break; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 03f36e534050f..a5b40e82dea4a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -969,6 +969,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_dai->channels = params_channels(&codec_params); codec_dai->sample_bits = snd_pcm_format_physical_width( params_format(&codec_params)); + + snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); } ret = soc_dai_hw_params(substream, params, cpu_dai); @@ -998,6 +1000,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); + snd_soc_dapm_update_dai(substream, params, cpu_dai); + ret = soc_pcm_params_symmetry(substream, params); if (ret) goto component_err; -- 2.11.0

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[alsa-devel] [PATCH] ASoC: dapm: Add warnings for widget overwrite when adding route
by Zhiwei Jiang 02 Feb '19

02 Feb '19

Currently, in some complex cases, more than one widgets have same name and registed from differnt dapm context, and route add from another context too. When snd_soc_dapm_add_route, the previous registered widget will overwritten by the latest same name widget, will cause unexpect error. For Asoc framework we cant avoid this situation and we cant decide which widget that wanted with route. At least we can give users a notice. Signed-off-by: Zhiwei Jiang <qq282012236(a)gmail.com> --- sound/soc/soc-dapm.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..af45ffcf712d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2741,6 +2741,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, char prefixed_sink[80]; char prefixed_source[80]; const char *prefix; + unsigned int sink_ref = 0; + unsigned int source_ref = 0; int ret; prefix = soc_dapm_prefix(dapm); @@ -2774,6 +2776,11 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, if (wsource) break; } + sink_ref++; + if (sink_ref > 1) + dev_warn(dapm->dev, + "ASoC: sink widget %s overwritten\n", + w->name); continue; } if (!wsource && !(strcmp(w->name, source))) { @@ -2783,6 +2790,11 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, if (wsink) break; } + source_ref++; + if (source_ref > 1) + dev_warn(dapm->dev, + "ASoC: source widget %s overwritten\n", + w->name); } } /* use widget from another DAPM context if not found from this */ -- 2.19.0

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[alsa-devel] [PATCH 0/2] ALSA: pcm: implement the anonymous dup v3
by Jaroslav Kysela 01 Feb '19

01 Feb '19

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup The filtering might be refined. It depends on the real requirements. Perhaps, we may create more ioctl groups. Any comments are more than welcome. v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) v3 of the patches: - group integer declarations in snd_pcm_anonymous_dup() - replaced substream->pcm with pcm in snd_pcm_anonymous_dup() - added SNDRV_PCM_PERM_RW check for read/write/readv/writev syscalls Cc: Phil Burk <philburk(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (2): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: implement the ioctl/mmap filter for the anonymous dup include/sound/pcm.h | 9 +-- include/uapi/sound/asound.h | 12 +++- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 ++-- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 158 ++++++++++++++++++++++++++++++++++++++++++-- 6 files changed, 176 insertions(+), 19 deletions(-) -- 2.13.6

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[alsa-devel] [PATCH 0/7] ASoC: topology: fixes and improvements
by Pierre-Louis Bossart 01 Feb '19

01 Feb '19

Module load/unload tests exposed a set of weaknesses in both the SOF and Skylake drivers, some of which are related to the topology framework. This patchset provides a combined set of fixes identified by the two teams. DAPM routes were also not properly handled, leading to confusing issues when the topology resources were released. All these patches were tested on top of the SOF tree, which tracks Mark's branch on a weekly basis. Amadeusz Sławiński (3): ASoC: topology: Reduce number of dereferences when accessing dobj ASoC: topology: Remove widgets from dobj list ASoC: topology: Fix memory leak from soc_tplg_denum_create_texts Bard liao (2): ASoC: topology: fix memory leak in soc_tplg_dapm_widget_create ASoC: topology: unload physical dai link in remove Ranjani Sridharan (2): ASoC: topology: add SND_SOC_DOBJ_GRAPH type for dapm routes ASoC: topology: modify dapm route loading routine and add dapm route unloading include/sound/soc-dapm.h | 2 + include/sound/soc-topology.h | 8 +- sound/soc/soc-topology.c | 165 +++++++++++++++++++++++++++++------ 3 files changed, 144 insertions(+), 31 deletions(-) -- 2.17.1

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Re: [alsa-devel] auto_mute for early patch issue
by Kailang 01 Feb '19

01 Feb '19

Hi Takashi, I move check line to below. It will show value for hp_pins[0]. As I know, search from verb table in function snd_hda_parse_pin_defcfg(). It will fill pins at auto_pin_cfg. Why it need to run below function to get cfg->hp_pins? static int check_auto_mute_availability(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; int present = 0; int i, err; - if (spec->suppress_auto_mute) - return 0; if (cfg->hp_pins[0]) present++; if (cfg->line_out_pins[0]) present++; if (cfg->speaker_pins[0]) present++; if (present < 2) /* need two different output types */ return 0; if (!cfg->speaker_pins[0] && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { memcpy(cfg->speaker_pins, cfg->line_out_pins, sizeof(cfg->speaker_pins)); cfg->speaker_outs = cfg->line_outs; } if (!cfg->hp_pins[0] && cfg->line_out_type == AUTO_PIN_HP_OUT) { memcpy(cfg->hp_pins, cfg->line_out_pins, sizeof(cfg->hp_pins)); cfg->hp_outs = cfg->line_outs; } + if (spec->suppress_auto_mute) + return 0; ...... ..... ...... } BR, Kailang -----Original Message----- From: Kailang Sent: Thursday, January 31, 2019 12:08 PM To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> Subject: auto_mute for early patch issue Hi Takashi, Chrome OS use early patch firmware file to load model and use auto_mute =no hint options. I find an issue for early patch issue about hint option auto_mute = no. This will cause all spec->init_hook can't get hp_pin values. For example: alc225_init() alc225_shutup() alc256_init() alc256_shutup() Early patch file hint auto_mute = no equal spec->suppress_auto_mute = 1 It will return function check_auto_mute_availability() in had_generic.c. hp_pin = spec->gen.autocfg.hp_pins[0]; ==> It will get value 0. I think this issue was for a long time. Please see the attach dmesg. One is kill hint auto_mute option. BR, Kailang

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[alsa-devel] [PATCH] Headset microphone support for System76 darp5
by Jeremy Soller 31 Jan '19

31 Jan '19

Headset microphone support for System76 darp5 Signed-off-by: Jeremy Soller <jeremy(a)system76.com> --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b4f472157ebd..15fedb0eeae1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5591,6 +5591,7 @@ enum { ALC294_FIXUP_ASUS_HEADSET_MIC, ALC294_FIXUP_ASUS_SPK, ALC225_FIXUP_HEADSET_JACK, + ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6537,6 +6538,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, }, + [ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6606,6 +6616,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY), + SND_PCI_QUIRK(0x1558, 0x1325, "System76", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), /* ALC282 */ SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),

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