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

[alsa-devel] [PATCH] ASoC: meson: aiu: simplify component addition
by Jerome Brunet 18 Feb '20

18 Feb '20

Now that the component name is unique within ASoC, there is no need to hack the debugfs prefix to add more than one ASoC component to a linux device. Remove the unnecessary function and use snd_soc_register_component() directly. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/aiu-acodec-ctrl.c | 7 +++---- sound/soc/meson/aiu-codec-ctrl.c | 7 +++---- sound/soc/meson/aiu.c | 20 -------------------- sound/soc/meson/aiu.h | 8 -------- 4 files changed, 6 insertions(+), 36 deletions(-) diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c index b8e88b1a4fc8..7078197e0cc5 100644 --- a/sound/soc/meson/aiu-acodec-ctrl.c +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -197,8 +197,7 @@ static const struct snd_soc_component_driver aiu_acodec_ctrl_component = { int aiu_acodec_ctrl_register_component(struct device *dev) { - return aiu_add_component(dev, &aiu_acodec_ctrl_component, - aiu_acodec_ctrl_dai_drv, - ARRAY_SIZE(aiu_acodec_ctrl_dai_drv), - "acodec"); + return snd_soc_register_component(dev, &aiu_acodec_ctrl_component, + aiu_acodec_ctrl_dai_drv, + ARRAY_SIZE(aiu_acodec_ctrl_dai_drv)); } diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c index 8646a953e3b3..4b773d3e8b07 100644 --- a/sound/soc/meson/aiu-codec-ctrl.c +++ b/sound/soc/meson/aiu-codec-ctrl.c @@ -144,9 +144,8 @@ static const struct snd_soc_component_driver aiu_hdmi_ctrl_component = { int aiu_hdmi_ctrl_register_component(struct device *dev) { - return aiu_add_component(dev, &aiu_hdmi_ctrl_component, - aiu_hdmi_ctrl_dai_drv, - ARRAY_SIZE(aiu_hdmi_ctrl_dai_drv), - "hdmi"); + return snd_soc_register_component(dev, &aiu_hdmi_ctrl_component, + aiu_hdmi_ctrl_dai_drv, + ARRAY_SIZE(aiu_hdmi_ctrl_dai_drv)); } diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 34b40b8b8299..d3e2d40e9562 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -71,26 +71,6 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, return 0; } -int aiu_add_component(struct device *dev, - const struct snd_soc_component_driver *component_driver, - struct snd_soc_dai_driver *dai_drv, - int num_dai, - const char *debugfs_prefix) -{ - struct snd_soc_component *component; - - component = devm_kzalloc(dev, sizeof(*component), GFP_KERNEL); - if (!component) - return -ENOMEM; - -#ifdef CONFIG_DEBUG_FS - component->debugfs_prefix = debugfs_prefix; -#endif - - return snd_soc_add_component(dev, component, component_driver, - dai_drv, num_dai); -} - static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component, struct of_phandle_args *args, const char **dai_name) diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index 097c26de7b7c..06a968c55728 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -11,9 +11,7 @@ struct clk; struct clk_bulk_data; struct device; struct of_phandle_args; -struct snd_soc_component_driver; struct snd_soc_dai; -struct snd_soc_dai_driver; struct snd_soc_dai_ops; enum aiu_clk_ids { @@ -45,12 +43,6 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, const char **dai_name, unsigned int component_id); -int aiu_add_component(struct device *dev, - const struct snd_soc_component_driver *component_driver, - struct snd_soc_dai_driver *dai_drv, - int num_dai, - const char *debugfs_prefix); - int aiu_hdmi_ctrl_register_component(struct device *dev); int aiu_acodec_ctrl_register_component(struct device *dev); -- 2.24.1

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[alsa-devel] [RFC] ASoC: soc-pcm: crash in snd_soc_dapm_new_dai
by Sameer Pujar 17 Feb '20

17 Feb '20

Crash happens in snd_soc_dapm_new_dai() when substream->private_data access is made and substream is NULL here. This is seen for DAIs where only playback or capture stream is defined. This seems to be happening for codec2codec DAI link. Both playback and capture are 0 during soc_new_pcm(). This is probably happening because cpu_dai and codec_dai are both validated either for SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE. Shouldn't be playback = 1 when, - playback stream is available for codec_dai AND - capture stream is available for cpu_dai and vice-versa for capture = 1? Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 74d340d..5aa9c0b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2855,10 +2855,10 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) for_each_rtd_codec_dai(rtd, i, codec_dai) { if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) playback = 1; if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) capture = 1; } -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda/realtek - Apply quirk for MSI GP63, too
by Takashi Iwai 17 Feb '20

17 Feb '20

The same quirk that was applied to MSI GL73 is needed for MSI GP63, too. Adding the entry with the SSID 1462:1228. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206503 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 6c8cb4ce517e..82485e06dde1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2447,6 +2447,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), + SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), -- 2.16.4

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[alsa-devel] [PATCH 0/2] ALSA: usb-audio: UAC2 effect unit parsing
by Takashi Iwai 17 Feb '20

17 Feb '20

Hi, UAC2 Effect Unit needs different parsing from Processing Unit and co, and this patch set implements the partial support for that. As it's the fix on top of the latest USB-audio fix, I'd like to apply through sound git tree. So please review and ack if USB header change is OK. thanks, Takashi === Takashi Iwai (2): usb: audio-v2: Add uac2_effect_unit_descriptor definition ALSA: usb-audio: Parse source ID of UAC2 effect unit include/linux/usb/audio-v2.h | 12 ++++++++++++ sound/usb/mixer.c | 6 ++++++ 2 files changed, 18 insertions(+) -- 2.16.4

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[alsa-devel] DMARC & gmail & cirrus.com
by Jaroslav Kysela 17 Feb '20

17 Feb '20

Hi all, today, all gmail users were unsubscribed from the mailing list, because we have users who sends e-mails from domains with the DMARC policy reject (like cirrus.com). The mailman counts those bounces and unsubscribe users who do not receive those e-mails. The nice explanation is here: https://www.linuxchix.org/content/mailing-list-changes I can enable the "Replace the From: header address" in the mailman config to "Munge From" like in the above example, but it will mean that From: will be altered (see the explanation). Another option is to disable the bounce check in mailman, but it will cause that the "dead" subscribers are not detected anymore. Any opinions on this? Jaroslav SMTP error: (host alt1.gmail-smtp-in.l.google.com[173.194.73.26] said: 550-5.7.26 Unauthenticated email from cirrus.com is not accepted due to domain's 550-5.7.26 DMARC policy. Please contact the administrator of cirrus.com domain 550-5.7.26 if this was a legitimate mail. Please visit 550-5.7.26 https://support.google.com/mail/answer/2451690 to learn about the 450 4.7.26 DMARC initiative. m14si10079981ljg.84 - gsmtp (in reply to end of DATA command)) DMARC settings for cirrus.com: v=DMARC1; p=reject; fo=1;rua=mailto:dmarc_rua@emaildefense.proofpoint.com;ruf=mailto:dmarc_ruf@emaildefense.proofpoint.com -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH v3 00/18] ASoC: soc-pcm cleanup step3
by Kuninori Morimoto 17 Feb '20

17 Feb '20

Hi Mark These are v3 of soc-pcm cleanup step3. Main change from v2 is that last patch was splited into each files (= [10/18] to [18/18]). And all patches have Reviewed-by tag from Ranjani Kuninori Morimoto (18): 1) ASoC: soc-pcm: move dai_get_widget() 2) ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() 3) ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() 4) ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() 5) ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() 6) ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() 7) ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code 8) ASoC: soc-pcm: use goto and remove multi return 9) ASoC: soc-pcm: merge playback/cature_active into stream_active 10) ASoC: soc.h: add for_each_pcm_streams() 11) ASoC: soc-core: use for_each_pcm_streams() macro 12) ASoC: soc-pcm: use for_each_pcm_streams() macro 13) ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro 14) ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro 15) ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro 16) ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro 17) ASoC: sof: sof-audio: use for_each_pcm_streams() macro 18) ALSA: usx2y: use for_each_pcm_streams() macro include/sound/pcm.h | 5 + include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 - sound/soc/codecs/cs4271.c | 4 +- sound/soc/dwc/dwc-i2s.c | 8 +- sound/soc/fsl/fsl_asrc_dma.c | 4 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/soc-core.c | 48 ++-- sound/soc/soc-generic-dmaengine-pcm.c | 8 +- sound/soc/soc-pcm.c | 327 +++++++++----------------- sound/soc/sof/sof-audio.c | 19 +- sound/usb/usx2y/usbusx2yaudio.c | 9 +- 12 files changed, 162 insertions(+), 284 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] drm/mediatek: fix race condition for HDMI jack status reporting
by Tzung-Bi Shih 17 Feb '20

17 Feb '20

hdmi_conn_detect and mtk_hdmi_audio_hook_plugged_cb would be called by different threads. Imaging the following calling sequence: Thread A Thread B -------------------------------------------------------------------- mtk_hdmi_audio_hook_plugged_cb() mtk_cec_hpd_high() -> disconnected hdmi_conn_detect() mtk_cec_hpd_high() -> connected plugged_cb(connected) plugged_cb(disconnected) The latest disconnected is false reported. Makes mtk_cec_hpd_high and plugged_cb atomic to fix. plugged_cb and codec_dev are also in danger of race condition. Instead of using mutex to protect them: - Checks NULLs first. - Uses WRITE_ONCE() to prevent store tearing (i.e. write to plugged_cb after codec_dev). - Uses codec_dev as a signal to report HDMI jack status. Fixes: 5d3c64477392 ("drm/mediatek: support HDMI jack status reporting") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Previous discussion: https://patchwork.kernel.org/patch/11367625/ Previous attempt: https://patchwork.kernel.org/patch/11378413/ drivers/gpu/drm/mediatek/mtk_hdmi.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c index 03aeb73005ef..b1e5d0c538fa 100644 --- a/drivers/gpu/drm/mediatek/mtk_hdmi.c +++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c @@ -12,6 +12,7 @@ #include <linux/io.h> #include <linux/kernel.h> #include <linux/mfd/syscon.h> +#include <linux/mutex.h> #include <linux/of_platform.h> #include <linux/of.h> #include <linux/of_gpio.h> @@ -171,6 +172,7 @@ struct mtk_hdmi { bool enabled; hdmi_codec_plugged_cb plugged_cb; struct device *codec_dev; + struct mutex update_plugged_status_lock; }; static inline struct mtk_hdmi *hdmi_ctx_from_bridge(struct drm_bridge *b) @@ -1199,10 +1201,13 @@ static void mtk_hdmi_clk_disable_audio(struct mtk_hdmi *hdmi) static enum drm_connector_status mtk_hdmi_update_plugged_status(struct mtk_hdmi *hdmi) { - bool connected = mtk_cec_hpd_high(hdmi->cec_dev); + bool connected; - if (hdmi->plugged_cb && hdmi->codec_dev) + mutex_lock(&hdmi->update_plugged_status_lock); + connected = mtk_cec_hpd_high(hdmi->cec_dev); + if (hdmi->codec_dev) hdmi->plugged_cb(hdmi->codec_dev, connected); + mutex_unlock(&hdmi->update_plugged_status_lock); return connected ? connector_status_connected : connector_status_disconnected; @@ -1669,8 +1674,12 @@ static int mtk_hdmi_audio_hook_plugged_cb(struct device *dev, void *data, { struct mtk_hdmi *hdmi = data; - hdmi->plugged_cb = fn; - hdmi->codec_dev = codec_dev; + if (!fn || !codec_dev) + return -EINVAL; + + /* Use WRITE_ONCE() to prevent store tearing. */ + WRITE_ONCE(hdmi->plugged_cb, fn); + WRITE_ONCE(hdmi->codec_dev, codec_dev); mtk_hdmi_update_plugged_status(hdmi); return 0; @@ -1729,6 +1738,7 @@ static int mtk_drm_hdmi_probe(struct platform_device *pdev) return ret; } + mutex_init(&hdmi->update_plugged_status_lock); platform_set_drvdata(pdev, hdmi); ret = mtk_hdmi_output_init(hdmi); -- 2.25.0.225.g125e21ebc7-goog

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[alsa-devel] [PATCH v2 00/10] ASoC: soc-pcm cleanup step3
by Kuninori Morimoto 17 Feb '20

17 Feb '20

Hi Mark These are v2 of soc-pcm cleanup step3. I removed unneeded Multi-Codec patch. And re-ordered / merged my local tree and add [09/10] patch to make [10/10] more effective. Kuninori Morimoto (10): 1) ASoC: soc-pcm: move dai_get_widget() 2) ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() 3) ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() 4) ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() 5) ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() 6) ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() 7) ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code 8) ASoC: soc-pcm: use goto and remove multi return 9) ASoC: soc-pcm: merge playback/cature_active into stream_active 10) ASoC: soc.h: add for_each_pcm_streams() include/sound/pcm.h | 5 + include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 - sound/soc/codecs/cs4271.c | 4 +- sound/soc/dwc/dwc-i2s.c | 8 +- sound/soc/fsl/fsl_asrc_dma.c | 4 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/soc-core.c | 48 ++-- sound/soc/soc-generic-dmaengine-pcm.c | 8 +- sound/soc/soc-pcm.c | 327 +++++++++----------------- sound/soc/sof/sof-audio.c | 19 +- sound/usb/usx2y/usbusx2yaudio.c | 9 +- 12 files changed, 162 insertions(+), 284 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH v2 0/3] simple-audio-card codec2codec support
by Samuel Holland 17 Feb '20

17 Feb '20

We are currently using simple-audio-card on the Allwinner A64 SoC. The digital audio codec there (sun8i-codec) has 3 AIFs, one each for the CPU, the modem, and Bluetooth. Adding support for the secondary AIFs requires adding codec2codec DAI links. Since the modem and bt-sco codec DAI drivers only have one set of possible PCM parameters (namely, 8kHz mono S16LE), there's no real need for a machine driver to specify the DAI link configuration. The parameters for these "simple" DAI links can be chosen automatically. This series adds codec2codec DAI link support to simple-audio-card. Codec to codec links are automatically detected when DAIs in the link belong to codec components. I tried to reuse as much code as possible, so the first two patches refactor a couple of helper functions to be more generic. The last patch adds the new feature and its documentation. Samuel Holland (3): ALSA: pcm: Add a non-runtime version of snd_pcm_limit_hw_rates ASoC: pcm: Export parameter intersection logic ASoC: simple-card: Add support for codec to codec DAI links Documentation/sound/soc/codec-to-codec.rst | 9 +++- include/sound/pcm.h | 9 +++- include/sound/soc.h | 3 ++ sound/core/pcm_misc.c | 18 +++---- sound/soc/generic/simple-card-utils.c | 50 ++++++++++++++++++++ sound/soc/soc-pcm.c | 55 +++++++++++++++------- 6 files changed, 116 insertions(+), 28 deletions(-) -- 2.24.1

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Re: [alsa-devel] Duet Firewire
by Takashi Sakamoto 17 Feb '20

17 Feb '20

Hi, (C.C.ed to alsa-devel) On Sun, Feb 16, 2020 at 03:52:27AM +0000, roberto lopez torres wrote: > Hi, i just got a Duet Firewire, and i'm trying to make it work with 5.3.0-29-gen > eric #31-Ubuntu > > I have inputs and outputs from the interface but can't config anything > at the moment, and i got a noise every 16 seconds on headphones > monitoring. > > I'm pretty new using linux. The ALSA support for Apogee Duet FireWire starts at Linux kernel v4.21. As you can see, the support is just for packet streaming, thus it's just possible to transmit/receive PCM frames and MIDI messages. However, the device is designed to run just depending on its internal clock (e.g. 47994 frames/sec), thus at present it generates periodical noise since the ALSA driver is designed to transmit just the same number of PCM frames as sampling rate (e.g. 48000 frames/sec). The support of any control is expected to be implemented in userspace application. This is the reason that you can see no elements in ALSA control application (e.g. alsamixer). However, the protocol to control the device is not clear yet. As long as I know, it's complicated and requires the batch of codes for the application. This is the reason that ALSA in-kernel driver doesn't have the code. Regards Takashi Sakamoto

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