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[PATCH 0/2] ASoC: remove obsolete drivers
by Arnd Bergmann 20 Jan '21

20 Jan '21

From: Arnd Bergmann <arnd(a)arndb.de> A few Arm platforms are getting removed in v5.12, this removes the corresponding sound drivers. Link: https://lore.kernel.org/linux-arm-kernel/20210120124812.2800027-1-arnd@kern… Arnd Bergmann (2): ASoC: remove sirf prima/atlas drivers ASoC: remove zte zx drivers .../bindings/sound/sirf-audio-codec.txt | 17 - .../devicetree/bindings/sound/sirf-usp.txt | 27 - .../devicetree/bindings/sound/zte,tdm.txt | 30 - .../bindings/sound/zte,zx-aud96p22.txt | 24 - .../devicetree/bindings/sound/zte,zx-i2s.txt | 45 -- .../bindings/sound/zte,zx-spdif.txt | 27 - sound/soc/Kconfig | 2 - sound/soc/Makefile | 2 - sound/soc/codecs/Makefile | 4 - sound/soc/codecs/sirf-audio-codec.c | 575 ------------------ sound/soc/codecs/zx_aud96p22.c | 401 ------------ sound/soc/sirf/Kconfig | 21 - sound/soc/sirf/Makefile | 8 - sound/soc/sirf/sirf-audio-port.c | 86 --- sound/soc/sirf/sirf-audio.c | 160 ----- sound/soc/sirf/sirf-usp.c | 435 ------------- sound/soc/sirf/sirf-usp.h | 292 --------- sound/soc/zte/Kconfig | 26 - sound/soc/zte/Makefile | 4 - sound/soc/zte/zx-i2s.c | 452 -------------- sound/soc/zte/zx-spdif.c | 363 ----------- sound/soc/zte/zx-tdm.c | 458 -------------- 22 files changed, 3459 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt delete mode 100644 Documentation/devicetree/bindings/sound/sirf-usp.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,tdm.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-aud96p22.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/zte,zx-spdif.txt delete mode 100644 sound/soc/codecs/sirf-audio-codec.c delete mode 100644 sound/soc/codecs/zx_aud96p22.c delete mode 100644 sound/soc/sirf/Kconfig delete mode 100644 sound/soc/sirf/Makefile delete mode 100644 sound/soc/sirf/sirf-audio-port.c delete mode 100644 sound/soc/sirf/sirf-audio.c delete mode 100644 sound/soc/sirf/sirf-usp.c delete mode 100644 sound/soc/sirf/sirf-usp.h delete mode 100644 sound/soc/zte/Kconfig delete mode 100644 sound/soc/zte/Makefile delete mode 100644 sound/soc/zte/zx-i2s.c delete mode 100644 sound/soc/zte/zx-spdif.c delete mode 100644 sound/soc/zte/zx-tdm.c -- 2.29.2

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[PATCH] ALSA: usb-audio: Fix hw constraints dependencies
by Takashi Iwai 20 Jan '21

20 Jan '21

Since the recent refactoring, it's been reported that some USB-audio devices (typically webcams) are no longer detected properly by PulseAudio. The debug session revealed that it's failing at probing by PA to try the sample rate 44.1kHz while the device has discrete sample rates other than 44.1kHz. But the puzzle was that arecord works as is, and some other devices with the discrete rates work, either. After all, this turned out to be the lack of the dependencies in a few hw constraint rules: snd_pcm_hw_rule_add() has the (variable) arguments specifying the dependent parameters, and some functions didn't set the target parameter itself as the dependencies. This resulted in an invalid parameter that could be generated only in a certain call pattern. This bug itself has been present in the code, but it didn't trigger errors just because the rules were casually avoiding such a corner case. After the recent refactoring and cleanup, however, the hw constraints work "as expected", and the problem surfaced now. For fixing the problem above, this patch adds the missing dependent parameters to each snd_pcm_hw_rule() call. Fixes: bc4e94aa8e72 ("ALSA: usb-audio: Handle discrete rates properly in hw constraints") BugLink: http://bugzilla.opensuse.org/show_bug.cgi?id=1181014 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f71965bf815f..078bb4c94033 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -981,6 +981,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, subs, + SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, param_period_time_if_needed, @@ -990,6 +991,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, + SNDRV_PCM_HW_PARAM_CHANNELS, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, param_period_time_if_needed, @@ -998,6 +1000,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_format, subs, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_CHANNELS, param_period_time_if_needed, -- 2.26.2

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[PATCH 2/2] ASoC: rt5682: add device property to control LDO2 power
by shumingf＠realtek.com 20 Jan '21

20 Jan '21

From: Shuming Fan <shumingf(a)realtek.com> The application circuit shall provide MICVDD power. In default, the codec driver doesn't need to enable LDO2 power. In case, the customers use VBAT for micbias, it should enable the device property "realtek,ldo2-pow-en". Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 18 +++++++++++++++++- 2 files changed, 18 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index 3900a07e3935..4b4c61b48380 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -41,6 +41,7 @@ struct rt5682_platform_data { unsigned int dmic_clk_rate; unsigned int dmic_delay; bool dmic_clk_driving_high; + bool ldo2_pow_en; const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 42ad5bea86b5..68d29d71d6e4 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1343,6 +1343,19 @@ static int is_using_asrc(struct snd_soc_dapm_widget *w, } } +static int is_using_ldo2(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + if (rt5682->pdata.ldo2_pow_en) + return 1; + + return 0; +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5682_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5682_STO1_ADC_MIXER, @@ -1854,7 +1867,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"CLKDET SYS", NULL, "CLKDET"}, - {"IN1P", NULL, "LDO2"}, + {"IN1P", NULL, "LDO2", is_using_ldo2}, {"BST1 CBJ", NULL, "IN1P"}, @@ -2999,6 +3012,9 @@ int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) rt5682->pdata.dmic_clk_driving_high = device_property_read_bool(dev, "realtek,dmic-clk-driving-high"); + rt5682->pdata.ldo2_pow_en = device_property_read_bool(dev, + "realtek,ldo2-pow-en"); + return 0; } EXPORT_SYMBOL_GPL(rt5682_parse_dt); -- 2.29.0

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[PATCH] ASoC: Intel: Skylake: Check the kcontrol against NULL
by Lukasz Majczak 20 Jan '21

20 Jan '21

There is no check for the kcontrol against NULL and in some cases it causes kernel to crash. Fixes: 2d744ecf2b984 ("ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT") Cc: <stable(a)vger.kernel.org> # 5.4+ Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- sound/soc/intel/skylake/skl-topology.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ae466cd592922..c9abbe4ff0ba3 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3618,12 +3618,18 @@ static void skl_tplg_complete(struct snd_soc_component *component) int i; list_for_each_entry(dobj, &component->dobj_list, list) { - struct snd_kcontrol *kcontrol = dobj->control.kcontrol; - struct soc_enum *se = - (struct soc_enum *)kcontrol->private_value; - char **texts = dobj->control.dtexts; + struct snd_kcontrol *kcontrol; + struct soc_enum *se; + char **texts; char chan_text[4]; + kcontrol = dobj->control.kcontrol; + if(!kcontrol) + continue; + + se = (struct soc_enum *)kcontrol->private_value; + texts = dobj->control.dtexts; + if (dobj->type != SND_SOC_DOBJ_ENUM || dobj->control.kcontrol->put != skl_tplg_multi_config_set_dmic) base-commit: 69fe63aa100220c8fd1f451dd54dd0895df1441d -- 2.25.1

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handling different frame sizes in snd_pcm_extplug_callback_t transfer callback
by Arsen Arsenović 20 Jan '21

20 Jan '21

Hello! I am writing an external noise cancellation plugin for ALSA, based on rnnoise, but I've hit a bit of a roadblock: rnnoise processes chunks of 480 floats at a time and does no buffering, but alsa expects the transfer callback to take all the data from the input and put it onto the output, with no way to signal that any buffering is or should be happening. I have tried to return a multiple of 480 from the transfer callback and keep a buffer with the leftover data, to be prepended to the next payload, but it looks like ALSA will, when a plugin returns too little data, just send back the same non-processed data in a new call, which breaks this idea. Any way I can work around these limitations? Another solution could be for external plugins to report the amount they consumed and produced, rather than a single number for both, but I am not aware of how to do that, if it's even possible with the filter plugin SDK. Thanks in advance! EDIT: I had previously posted this question before I confirmed my mailing list membership, but seeing as it wasn't approved for a few days I cancelled the request and reposted it. Sorry about that. -- Arsen Arsenović

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sof topology compilation fails
by GitHub issues - opened 20 Jan '21

20 Jan '21

alsa-project/alsa-lib issue #114 was opened from juimonen: sof topologies fail to compile because of following commit: commit 51e1d486ce4535923692bd1d7e59d5aa3a29cd50 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Thu Jan 7 17:40:42 2021 +0100 topology: parse_tuple_set() - remove dead condition code Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> I don't completely understand the change...? we are in "case SND_SOC_TPLG_TUPLE_TYPE_WORD" and then we check for SND_SOC_TPLG_TUPLE_TYPE_WORD (and not the value anymore) and error out? Issue URL : https://github.com/alsa-project/alsa-lib/issues/114 Repository URL: https://github.com/alsa-project/alsa-lib

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Add jack2 as test
by GitHub pull_request - opened 20 Jan '21

20 Jan '21

alsa-project/alsa-utils pull request #74 was opened from bengan: Jack2 been around for a long time. It should be included in the test. Request URL : https://github.com/alsa-project/alsa-utils/pull/74 Patch URL : https://github.com/alsa-project/alsa-utils/pull/74.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH 1/2] ASoC: dt-bindings: rt5682: add LDO2 power control device property
by shumingf＠realtek.com 20 Jan '21

20 Jan '21

From: Shuming Fan <shumingf(a)realtek.com> The "realtek,ldo2-pow-en" device property uses to enable LDO2 power while the headset recording. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5682.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index 9c5fadb6ac82..eba2ea588564 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -46,6 +46,8 @@ Optional properties: - realtek,dmic-clk-driving-high : Set the high drving of the DMIC clock out. +- realtek,ldo2-pow-en : Enable LDO2 power control if the application circuit uses VBAT, not MICVDD. + Pins on the device (for linking into audio routes) for RT5682: * DMIC L1 -- 2.29.0

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[PATCH] Add basic capture support for Pioneer DJ DJM-250MK2
by František Kučera 20 Jan '21

20 Jan '21

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows recording. However, DVS is not possible yet (see the comment in code). Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> --- sound/usb/pcm.c | 1 + sound/usb/quirks-table.h | 62 ++++++++++++++++++++++++++++++++++------ sound/usb/quirks.c | 1 + 3 files changed, 56 insertions(+), 8 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5600751803cf..f5fa11deea4d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -374,6 +374,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ ep = 0x82; ifnum = 0; goto add_sync_ep_from_ifnum; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f4fb002e3ef4..c6b2f0b8d3e2 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3549,14 +3549,40 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), { /* * Pioneer DJ DJM-250MK2 - * PCM is 8 channels out @ 48 fixed (endpoints 0x01). - * The output from computer to the mixer is usable. - * - * The input (phono or line to computer) is not working. - * It should be at endpoint 0x82 and probably also 8 channels, - * but it seems that it works only with Pioneer proprietary software. - * Even on officially supported OS, the Audacity was unable to record - * and Mixxx to recognize the control vinyls. + * PCM is 8 channels out @ 48 fixed (endpoint 0x01) + * and 8 channels in @ 48 fixed (endpoint 0x82). + * + * Both playback and recording is working, even simultaneously. + * + * Playback channels could be mapped to: + * - CH1 + * - CH2 + * - AUX + * + * Recording channels could be mapped to: + * - Post CH1 Fader + * - Post CH2 Fader + * - Cross Fader A + * - Cross Fader B + * - MIC + * - AUX + * - REC OUT + * + * There is remaining problem with recording directly from PHONO/LINE. + * If we map a channel to: + * - CH1 Control Tone PHONO + * - CH1 Control Tone LINE + * - CH2 Control Tone PHONO + * - CH2 Control Tone LINE + * it is silent. + * There is no signal even on other operating systems with official drivers. + * The signal appears only when a supported application is started. + * This needs to be investigated yet... + * (there is quite a lot communication on the USB in both directions) + * + * In current version this mixer could be used for playback + * and for recording from vinyls (through Post CH* Fader) + * but not for DVS (Digital Vinyl Systems) like in Mixxx. */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { @@ -3580,6 +3606,26 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .rate_max = 48000, .nr_rates = 1, .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // inputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index abf99b814a0f..6a0be7e7f4fe 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1493,6 +1493,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, set_format_emu_quirk(subs, fmt); break; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ pioneer_djm_set_format_quirk(subs); break; case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ -- 2.20.1

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Dual ALC1220 Bug MSI Godlike X570
by Aeonik 20 Jan '21

20 Jan '21

Hello, I believe I have run into an Alsa bug due to new/unsupported hardware. I am running Arch Linux Kernel version 5.10.8 on my new MSI Godlike X570 motherboard, and I believe Alsa is having issues with the Dual ALC1220 Codecs. I saw that this has been an issue before <https://bugzilla.kernel.org/show_bug.cgi?id=195305> in the and looks solved, but for some reason, my Alsa utilities are crashing when trying to access my sound card (Alsamixer and amixer). Here is an example with the error message. amixer -c 0 ALSA lib simple_none.c:1544:(simple_add1) helem (MIXER,'Master Playback Switch',0,1,0) appears twice or more cannot load mixer controls: Invalid argument I have tried solving the issue in a variety of ways, including explicitly setting the kernel module snd-hda_intel model name. I also experience the same bug in a fresh Arch Linux ISO, which rules out most configuration issues, I think. I reviewed your documentation, and it looks like my next step is to file a bug report. I already reached out to the Arch Linux community, and you can review my post at the following link. I have also attached the output of alsa-info.sh to this email. https://bbs.archlinux.org/viewtopic.php?pid=1949255#p1949255 Let me know if you need anything else from me, I appreciate all your work, and look forward to your response! Aeonik

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