September 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/hdmi: run eld notify in delay work
by Brent Lu 05 Oct '22

05 Oct '22

During resolution change, display driver would disable HDMI audio then enable it in a short time. There is possibility that eld notify for HDMI audio enable is called when previous runtime suspend is still running. In this case, the elf nofity just returns and not updating the status of corresponding HDMI pin/port. Here we move the eld nofity to a delay work so we don't lose it. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 38 +++++++++++++++++++++++++++++++++----- 1 file changed, 33 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6c209cd26c0c..a4c305ee8ff9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2907,13 +2907,21 @@ static int intel_port2pin(struct hda_codec *codec, int port) return spec->port_map[port]; } -static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) +struct pin_eld_notify { + void *audio_ptr; + int port; + int pipe; + struct delayed_work notify_work; +}; + +static void pin_eld_notify_work(struct work_struct *work) { - struct hda_codec *codec = audio_ptr; + struct pin_eld_notify *notify = container_of(work, struct pin_eld_notify, notify_work.work); + struct hda_codec *codec = notify->audio_ptr; int pin_nid; - int dev_id = pipe; + int dev_id = notify->pipe; - pin_nid = intel_port2pin(codec, port); + pin_nid = intel_port2pin(codec, notify->port); if (!pin_nid) return; /* skip notification during system suspend (but not in runtime PM); @@ -2922,13 +2930,33 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ - if (snd_hdac_is_in_pm(&codec->core)) + if (snd_hdac_is_in_pm(&codec->core)) { + schedule_delayed_work(&notify->notify_work, msecs_to_jiffies(10)); return; + } snd_hdac_i915_set_bclk(&codec->bus->core); check_presence_and_report(codec, pin_nid, dev_id); } +static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) +{ + struct hda_codec *codec = audio_ptr; + struct device *dev = hda_codec_dev(codec); + struct pin_eld_notify *notify; + + notify = devm_kzalloc(dev, sizeof(struct pin_eld_notify), GFP_KERNEL); + if (!notify) + return; + + notify->audio_ptr = audio_ptr; + notify->port = port; + notify->pipe = pipe; + INIT_DELAYED_WORK(&notify->notify_work, pin_eld_notify_work); + + schedule_delayed_work(&notify->notify_work, 0); +} + static const struct drm_audio_component_audio_ops intel_audio_ops = { .pin2port = intel_pin2port, .pin_eld_notify = intel_pin_eld_notify, -- 2.25.1

3 7

[PATCH] ALSA: usb-audio: Split endpoint setups for hw_params and prepare (take#2)
by Takashi Iwai 05 Oct '22

05 Oct '22

This is a second attempt to fix the bug appearing on Android with the recent kernel; the first try was ff878b408a03 and reverted at commit 79764ec772bc. The details taken from the v1 patch: One of the former changes for the endpoint management was the more consistent setup of endpoints at hw_params. snd_usb_endpoint_configure() is a single function that does the full setup, and it's called from both PCM hw_params and prepare callbacks. Although the EP setup at the prepare phase is usually skipped (by checking need_setup flag), it may be still effective in some cases like suspend/resume that requires the interface setup again. As it's a full and single setup, the invocation of snd_usb_endpoint_configure() includes not only the USB interface setup but also the buffer release and allocation. OTOH, doing the buffer release and re-allocation at PCM prepare phase is rather superfluous, and better to be done only in the hw_params phase. For those optimizations, this patch splits the endpoint setup to two phases: snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(), to be called from hw_params and from prepare, respectively. Note that this patch changes the driver operation slightly, effectively moving the USB interface setup again to PCM prepare stage instead of hw_params stage, while the buffer allocation and such initializations are still done at hw_params stage. And, the change of the USB interface setup timing (moving to prepare) gave an interesting "fix", too: it was reported that the recent kernels caused silent output at the beginning on playbacks on some devices on Android, and this change casually fixed the regression. It seems that those devices are picky about the sample rate change (or the interface change?), and don't follow the too immediate rate changes. Meanwhile, Android operates the PCM in the following order: - open, then hw_params with the possibly highest sample rate - close without prepare - re-open, hw_params with the normal sample rate - prepare, and start streaming This procedure ended up the hw_params twice with different rates, and because the recent kernel did set up the sample rate twice one and after, it screwed up the device. OTOH, the earlier kernels didn't set up the USB interface at hw_params, hence this problem didn't appear. Now, with this patch, the USB interface setup is again back to the prepare phase, and it works around the problem automagically. Although we should address the sample rate problem in a more solid way in future, let's keep things working as before for now. *** What's new in the take#2 patch: - The regression caused by the v1 patch (bko#216500) was due to the missing check of need_setup flag at hw_params. Now the check is added, and the snd_usb_endpoint_set_params() call is skipped when the running EP is re-opened. - There was another bug in v1 where the clock reference rate wasn't updated at hw_params phase, which may lead to a lack of the proper hw constraints when an application doesn't issue the prepare but only the hw_params call. This patch fixes it as well by tracking the clock rate change in the prepare callback with a new flag "need_update" for the clock reference object, just like others. - The configure_endpoints() are simplified and folded back into snd_usb_pcm_prepare(). Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Fixes: ff878b408a03 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare") Reported-by: chihhao chen <chihhao.chen(a)mediatek.com> Link: https://lore.kernel.org/r/87e6d6ae69d68dc588ac9acc8c0f24d6188375c3.camel@me… Link: https://lore.kernel.org/r/20220901124136.4984-1-tiwai@suse.de Link: https://bugzilla.kernel.org/show_bug.cgi?id=216500 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 76 +++++++++++++++++++++++++++----------------- sound/usb/endpoint.h | 6 ++-- sound/usb/pcm.c | 51 ++++++++++++----------------- 3 files changed, 70 insertions(+), 63 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index eb71df9da831..0c94ebc98e90 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -40,6 +40,7 @@ struct snd_usb_clock_ref { unsigned char clock; atomic_t locked; int rate; + bool need_setup; struct list_head list; }; @@ -758,7 +759,8 @@ bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, * The endpoint needs to be closed via snd_usb_endpoint_close() later. * * Note that this function doesn't configure the endpoint. The substream - * needs to set it up later via snd_usb_endpoint_configure(). + * needs to set it up later via snd_usb_endpoint_set_params() and + * snd_usb_endpoint_prepare(). */ struct snd_usb_endpoint * snd_usb_endpoint_open(struct snd_usb_audio *chip, @@ -1289,15 +1291,39 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep) return -ENOMEM; } +/* update the rate of the referred clock; return the actual rate */ +static int update_clock_ref_rate(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) +{ + struct snd_usb_clock_ref *clock = ep->clock_ref; + int rate = ep->cur_rate; + + if (!clock || clock->rate == rate) + return rate; + if (clock->rate) { + if (atomic_read(&clock->locked)) + return clock->rate; + if (clock->rate != rate) { + usb_audio_err(chip, "Mismatched sample rate %d vs %d for EP 0x%x\n", + clock->rate, rate, ep->ep_num); + return clock->rate; + } + } + clock->rate = rate; + clock->need_setup = true; + return rate; +} + /* * snd_usb_endpoint_set_params: configure an snd_usb_endpoint * + * It's called either from hw_params callback. * Determine the number of URBs to be used on this endpoint. * An endpoint must be configured before it can be started. * An endpoint that is already running can not be reconfigured. */ -static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { const struct audioformat *fmt = ep->cur_audiofmt; int err; @@ -1349,49 +1375,46 @@ static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, ep->maxframesize = ep->maxpacksize / ep->cur_frame_bytes; ep->curframesize = ep->curpacksize / ep->cur_frame_bytes; - return 0; + return update_clock_ref_rate(chip, ep); } static int init_sample_rate(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep) { struct snd_usb_clock_ref *clock = ep->clock_ref; - int err; + int rate, err; - if (clock) { - if (atomic_read(&clock->locked)) - return 0; - if (clock->rate == ep->cur_rate) - return 0; - if (clock->rate && clock->rate != ep->cur_rate) { - usb_audio_dbg(chip, "Mismatched sample rate %d vs %d for EP 0x%x\n", - clock->rate, ep->cur_rate, ep->ep_num); - return -EINVAL; - } - } + rate = update_clock_ref_rate(chip, ep); + if (rate < 0) + return rate; + if (clock && !clock->need_setup) + return 0; - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); - if (err < 0) + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, rate); + if (err < 0) { + if (clock) + clock->rate = 0; /* reset rate */ return err; + } if (clock) - clock->rate = ep->cur_rate; + clock->need_setup = false; return 0; } /* - * snd_usb_endpoint_configure: Configure the endpoint + * snd_usb_endpoint_prepare: Prepare the endpoint * * This function sets up the EP to be fully usable state. - * It's called either from hw_params or prepare callback. + * It's called either from prepare callback. * The function checks need_setup flag, and performs nothing unless needed, * so it's safe to call this multiple times. * * This returns zero if unchanged, 1 if the configuration has changed, * or a negative error code. */ -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { bool iface_first; int err = 0; @@ -1412,9 +1435,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; } - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; goto done; } @@ -1442,10 +1462,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; - err = snd_usb_select_mode_quirk(chip, ep->cur_audiofmt); if (err < 0) goto unlock; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6a9af04cf175..e67ea28faa54 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -17,8 +17,10 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, bool is_sync_ep); void snd_usb_endpoint_close(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep); +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock); bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d45d1d7e6664..e721fc12acde 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -433,35 +433,6 @@ static void close_endpoints(struct snd_usb_audio *chip, } } -static int configure_endpoints(struct snd_usb_audio *chip, - struct snd_usb_substream *subs) -{ - int err; - - if (subs->data_endpoint->need_setup) { - /* stop any running stream beforehand */ - if (stop_endpoints(subs, false)) - sync_pending_stops(subs); - if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - err = snd_usb_endpoint_configure(chip, subs->data_endpoint); - if (err < 0) - return err; - snd_usb_set_format_quirk(subs, subs->cur_audiofmt); - } else { - if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - } - - return 0; -} - /* * hw_params callback * @@ -551,7 +522,16 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, subs->cur_audiofmt = fmt; mutex_unlock(&chip->mutex); - ret = configure_endpoints(chip, subs); + if (!subs->data_endpoint->need_setup) + goto unlock; + + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_set_params(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_set_params(chip, subs->data_endpoint); unlock: if (ret < 0) @@ -634,9 +614,18 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (ret < 0) goto unlock; + else if (ret > 0) + snd_usb_set_format_quirk(subs, subs->cur_audiofmt); + ret = 0; /* reset the pointer */ subs->buffer_bytes = frames_to_bytes(runtime, runtime->buffer_size); -- 2.35.3

2 3

[PATCH v2 -next 0/4] Fix PM disable depth imbalance in probe
by Zhang Qilong 03 Oct '22

03 Oct '22

The pm_runtime_enable will increase power disable depth. Thus a pairing decrement is needed on the error handling path to keep it balanced according to context. We fix it by calling pm_runtime_disable when error returns. --- v2: - call pm_runtime_disable when error returns. Zhang Qilong (4): ASoC: wm8997: Fix PM disable depth imbalance in wm8997_probe ASoC: wm5110: Fix PM disable depth imbalance in wm5110_probe ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe ASoC: mt6660: Fix PM disable depth imbalance in mt6660_i2c_probe sound/soc/codecs/mt6660.c | 3 +++ sound/soc/codecs/wm5102.c | 1 + sound/soc/codecs/wm5110.c | 1 + sound/soc/codecs/wm8997.c | 1 + 4 files changed, 6 insertions(+) -- 2.25.1

4 10

Sound broken (first couple of seconds play looping) on Sony Vaio VPX11S1E (HDA, ALC262)
by Hans de Goede 01 Oct '22

01 Oct '22

Hi All, To test some kernel work (backlight refactor) which I have been doing I have booted Debian Testing with a 6.0 kernel on a Sony Vaio VPX11S1E (which is ancient). I noticed that when tab-completing something in a terminal the terminal bell sound would keep repeating and playing another longer sound sample gets stuck with the first couple of seconds of that sample looping. This is under GNOME3 with pulseaudio as sound server. I accidentally found an interesting workaround if I run: aplay -Dplughw:CARD=MID,DEV=0 /usr/share/sounds/alsa/Front_Left.wav once, then that works properly and after that the problem is gone... Note that this laptop does have only 1 speaker AFAICT. alsa-info output below. I would be happy to test any patches / module-options which might fix this. Regards, Hans upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.5.1 !!################################ !!Script ran on: Sat Sep 10 19:35:19 UTC 2022 !!Linux Distribution !!------------------ Debian GNU/Linux bookworm/sid \n \l PRETTY_NAME="Debian GNU/Linux bookworm/sid" NAME="Debian GNU/Linux" ID=debian HOME_URL="https://www.debian.org/" SUPPORT_URL="https://www.debian.org/support" BUG_REPORT_URL="https://bugs.debian.org/" !!DMI Information !!--------------- Manufacturer: Sony Corporation Product Name: VPCX11S1E Product Version: J003ZJ9Y Firmware Version: R0084U4 System SKU: N/A Board Vendor: Sony Corporation Board Name: VAIO !!ACPI Device Status Information !!--------------- /sys/bus/acpi/devices/PNP0103:00/status 15 /sys/bus/acpi/devices/PNP0C0A:00/status 31 /sys/bus/acpi/devices/PNP0C0F:00/status 9 /sys/bus/acpi/devices/PNP0C0F:01/status 9 /sys/bus/acpi/devices/PNP0C0F:02/status 9 /sys/bus/acpi/devices/PNP0C0F:03/status 9 /sys/bus/acpi/devices/PNP0C0F:04/status 9 /sys/bus/acpi/devices/PNP0C0F:05/status 9 /sys/bus/acpi/devices/PNP0C0F:06/status 9 /sys/bus/acpi/devices/PNP0C0F:07/status 9 /sys/bus/acpi/devices/SNY6001:00/status 13 !!Kernel Information !!------------------ Kernel release: 6.0.0-rc4-32+ Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k6.0.0-rc4-32+ Library version: 1.2.7.2 Utilities version: 1.2.7 !!Loaded ALSA modules !!------------------- snd_hda_intel (card 0) !!Sound Servers on this system !!---------------------------- PipeWire: Installed - Yes (/usr/bin/pipewire) Running - No Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [MID ]: HDA-Intel - HDA Intel MID HDA Intel MID at 0x942c0000 irq 19 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device [0403]: Intel Corporation US15W/US15X/US15L/UL11L SCH [Poulsbo] HD Audio Controller [8086:811b] (rev 06) Subsystem: Sony Corporation US15W/US15X/US15L/UL11L SCH [Poulsbo] HD Audio Controller [104d:905f] !!Modprobe options (Sound related) !!-------------------------------- snd_pcsp: index=-2 snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y dmic_detect : Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 1 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Sysfs card info !!--------------- !!Card: /sys/class/sound/card0 Driver: /sys/bus/pci/drivers/snd_hda_intel Tree: !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC262 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0262 Subsystem Id: 0x104d3c00 Revision Id: 0x100302 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP Power: setting=D0, actual=D0 GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x411: Stereo Converter: stream=3, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x411: Stereo Device: name="ALC262 Analog", type="Audio", device=0 Converter: stream=3, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x07 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x88 0x88] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x88 0x88] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC262 Analog", type="Audio", device=0 Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x19 0x19] Converter: stream=1, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100791: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Headphone Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=7, ofs=0 Control: name="Headphone Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=7, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 8 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x1e, nsteps=0x1e, stepsize=0x05, mute=0 Amp-Out vals: [0x01 0x01] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x1e, nsteps=0x1e, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] [0x80] Amp-Out caps: ofs=0x1e, nsteps=0x1e, stepsize=0x05, mute=0 Amp-Out vals: [0x1a] Connection: 2 0x02 0x0b Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x11 [Pin Complex] wcaps 0x400780: Mono Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x10 Node 0x12 [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x90a60120: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x15 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Node 0x16 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00] Pincap 0x00000010: OUT Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x19 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1c [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1e [Pin Complex] wcaps 0x400780: Mono Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400680: Mono Digital Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=32, direct=0, val=0 Unsolicited: tag=00, enabled=0 Connection: 0 Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x0b 0x12* Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 9 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x0b Node 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 9 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x0b --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 5 Sep 10 21:29 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 Sep 10 21:29 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 3 Sep 10 21:29 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 2 Sep 10 21:29 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Sep 10 21:29 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Sep 10 21:29 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Sep 10 21:29 . drwxr-xr-x 3 root root 180 Sep 10 21:29 .. lrwxrwxrwx 1 root root 12 Sep 10 21:29 pci-0000:00:1b.0 -> ../controlC0 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: MID [HDA Intel MID], device 0: ALC262 Analog [ALC262 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: MID [HDA Intel MID], device 0: ALC262 Analog [ALC262 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card MID Card sysdefault:0 'MID'/'HDA Intel MID at 0x942c0000 irq 19' Mixer name : 'Realtek ALC262' Components : 'HDA:10ec0262,104d3c00,00100302' Controls : 22 Simple ctrls : 11 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 30 Mono: Playback 26 [87%] [-6.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 30 Mono: Front Left: Playback 0 [0%] [-45.00dB] [off] Front Right: Playback 0 [0%] [-45.00dB] [off] Simple mixer control 'Headphone Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [off] Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Headphone Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Speaker',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 30 Mono: Playback 30 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 25 [81%] [25.50dB] [on] Front Right: Capture 25 [81%] [25.50dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Internal Mic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] Simple mixer control 'Loopback Mixing',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' !!Alsactl output !!-------------- --startcollapse-- state.MID { control.1 { iface MIXER name 'Headphone Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 30' dbmin -4500 dbmax 0 dbvalue.0 -4500 dbvalue.1 -4500 } } control.2 { iface MIXER name 'Headphone Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Speaker Playback Volume' value 30 comment { access 'read write' type INTEGER count 1 range '0 - 30' dbmin -4500 dbmax 0 dbvalue.0 0 } } control.4 { iface MIXER name 'Speaker Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.5 { iface MIXER name 'Loopback Mixing' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.6 { iface MIXER name 'Headphone Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.7 { iface MIXER name 'Headphone Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.8 { iface MIXER name 'Auto-Mute Mode' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.9 { iface MIXER name 'Capture Source' value 'Internal Mic' comment { access 'read write' type ENUMERATED count 1 item.0 'Internal Mic' item.1 'Headphone Mic' } } control.10 { iface MIXER name 'Capture Volume' value.0 25 value.1 25 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1200 dbmax 3450 dbvalue.0 2550 dbvalue.1 2550 } } control.11 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.12 { iface MIXER name 'Headphone Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } control.13 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.14 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.15 { iface MIXER name 'Master Playback Volume' value 26 comment { access 'read write' type INTEGER count 1 range '0 - 30' dbmin -4500 dbmax 0 dbvalue.0 -600 } } control.16 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.17 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.18 { iface CARD name 'Headphone Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.19 { iface CARD name 'Speaker Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.20 { iface PCM name 'Playback Channel Map' value.0 3 value.1 4 comment { access read type INTEGER count 2 range '0 - 36' } } control.21 { iface PCM name 'Capture Channel Map' value.0 3 value.1 4 comment { access read type INTEGER count 2 range '0 - 36' } } control.22 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!All Loaded Modules !!------------------ ac acpi_cpufreq ansi_cprng async_memcpy async_pq async_raid6_recov async_tx async_xor ata_generic ath ath9k ath9k_common ath9k_hw autofs4 battery blake2b_generic bluetooth bnep btbcm btintel btmtk btrfs btrtl btusb button ccm cfg80211 configfs coretemp crc16 crc32c_generic crc64 crc64_rocksoft crc64_rocksoft_generic crc_t10dif crct10dif_common crct10dif_generic ctr drbg drm drm_kms_helper ecc ecdh_generic ehci_hcd ehci_pci evdev ext4 fb_sys_fops fuse gma500_gfx i2c_algo_bit i2c_isch ip_tables irqbypass jbd2 jitterentropy_rng joydev kvm kvm_intel ledtrig_audio libarc4 libata libcrc32c linear lp lpc_sch mac80211 mbcache mc md_mod msr multipath parport parport_pc pata_sch pcspkr ppdev psmouse qcserial qrtr raid0 raid1 raid10 raid456 raid6_pq rfcomm rfkill scsi_common scsi_mod sd_mod serio_raw sg sha512_generic sky2 snd snd_hda_codec snd_hda_codec_generic snd_hda_codec_realtek snd_hda_core snd_hda_intel snd_hwdep snd_intel_dspcfg snd_intel_sdw_acpi snd_pcm snd_timer sony_laptop soundcore syscopyarea sysfillrect sysimgblt t10_pi uhci_hcd usb_common usb_wwan usbcore usbserial uvcvideo video videobuf2_common videobuf2_memops videobuf2_v4l2 videobuf2_vmalloc videodev wmi x_tables xor zstd_compress !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x11 0x411111f0 0x12 0x90a60120 0x14 0x411111f0 0x15 0x0321101f 0x16 0x90170110 0x18 0x411111f0 0x19 0x411111f0 0x1a 0x411111f0 0x1b 0x411111f0 0x1c 0x411111f0 0x1d 0x411111f0 0x1e 0x411111f0 0x1f 0x411111f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: !!ALSA/HDA dmesg !!-------------- [ 0.519705] ACPI: Added _OSI(Linux-Dell-Video) [ 0.519705] ACPI: Added _OSI(Linux-Lenovo-NV-HDMI-Audio) [ 0.519705] ACPI: Added _OSI(Linux-HPI-Hybrid-Graphics) -- [ 25.318161] cfg80211: Loading compiled-in X.509 certificates for regulatory database [ 25.469763] snd_hda_codec_realtek hdaudioC0D0: ALC262: SKU not ready 0x411111f0 [ 25.470190] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC262: line_outs=1 (0x16/0x0/0x0/0x0/0x0) type:speaker [ 25.470275] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 25.470288] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 25.470300] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 25.470308] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 25.470318] snd_hda_codec_realtek hdaudioC0D0: Mic=0x12 [ 25.515219] cfg80211: Loaded X.509 cert 'sforshee: 00b28ddf47aef9cea7' [ 25.519738] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 25.520143] input: HDA Intel MID Headphone Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 25.542404] cfg80211: loaded regulatory.db is malformed or signature is missing/invalid -- [ 259.241575] rfkill: input handler enabled [ 263.284660] snd_hda_intel 0000:00:1b.0: Invalid position buffer, using LPIB read method instead. [ 263.284826] snd_hda_intel 0000:00:1b.0: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. [ 288.223033] rfkill: input handler disabled !!Packages installed !!-------------------- ii alsa-topology-conf 1.2.5.1-2 all ALSA topology configuration files ii alsa-ucm-conf 1.2.7.2-1 all ALSA Use Case Manager configuration files ii alsa-utils 1.2.7-1 i386 Utilities for configuring and using ALSA

2 8

issues with the Arturia minifuse 1 sound card. i would like help with converting the minifuse 2 UCM config to work with minifuse 1
by GitHub issues - opened 30 Sep '22

30 Sep '22

alsa-project/alsa-ucm-conf issue #220 was opened from Ali32bit: greetings. so i have a sound card with multiple outputs and some loop back channels. it is USB complaint but those extra bells and whistles cause ALSA to think its a surround sound device when its actually just stereo. and this actually impacts the sound when i play games on it cause sounds from behind are basically muted. and thats the case with some music too but less obvious here is what alsa_info generated for me too : http://alsa-project.org/db/?f=b6db7723ffa65f32b93297109bf1f8e4d853cf4e there should already be a minifuse 2 setup so if possible please adapt it to minifuse 1 or help me convert and install it myself. the hardware is mostly the same but it has 1 input instead of 2 and no extra midi ports in the back. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/220 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.0-rc7
by Mark Brown 30 Sep '22

30 Sep '22

The following changes since commit 0a0342ede303fc420f3a388e1ae82da3ae8ff6bd: ASoC: tas2770: Reinit regcache on reset (2022-09-19 18:48:43 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.0-rc7 for you to fetch changes up to e18f6bcf8e864ea0e9690691d0d749c662b6a2c7: ASoC: wcd-mbhc-v2: Revert "ASoC: wcd-mbhc-v2: use pm_runtime_resume_and_get()" (2022-09-29 15:20:12 +0100) ---------------------------------------------------------------- ASoC: Last fixes for v6.0 In the even that there's another pull request here's some more driver specific fixes. ---------------------------------------------------------------- Krzysztof Kozlowski (3): ASoC: wcd9335: fix order of Slimbus unprepare/disable ASoC: wcd934x: fix order of Slimbus unprepare/disable ASoC: wcd-mbhc-v2: Revert "ASoC: wcd-mbhc-v2: use pm_runtime_resume_and_get()" Pierre-Louis Bossart (1): ASoC: es8316: fix register sync error in suspend/resume tests Shengjiu Wang (1): ASoC: fsl_audmux: Fix amixer write errors sound/soc/codecs/es8316.c | 2 ++ sound/soc/codecs/wcd-mbhc-v2.c | 10 ++++++---- sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wcd934x.c | 2 +- sound/soc/fsl/fsl_audmix.c | 16 ++++------------ 5 files changed, 14 insertions(+), 18 deletions(-)

2 1

[PATCH] ASoC: qcom: fix unmet direct dependencies for SND_SOC_QDSP6
by Srinivas Kandagatla 30 Sep '22

30 Sep '22

SND_SOC_QDSP6 already has COMPILE_TEST so remove that from SND_SOC_SC8280XP and also add QCOM_APR dependencies to SND_SOC_SC8280XP like other Qualcomm machine drivers. This should also fix below warning: on x86_64, when QCOM_APR is not set and COMPILE_TEST=y: WARNING: unmet direct dependencies detected for SND_SOC_QDSP6 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && QCOM_APR [=n] && COMMON_CLK [=y] Selected by [y]: - SND_SOC_SC8280XP [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && (QCOM_APR [=n] || COMPILE_TEST [=y]) && SOUNDWIRE [=y] && COMMON_CLK [=y] Fixes: 295aeea6646a ("ASoC: qcom: add machine driver for sc8280xp") Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 1b0252ec4665..d0e59e07b1fc 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -175,8 +175,7 @@ config SND_SOC_SM8250 config SND_SOC_SC8280XP tristate "SoC Machine driver for SC8280XP boards" - depends on QCOM_APR || COMPILE_TEST - depends on SOUNDWIRE + depends on QCOM_APR && SOUNDWIRE depends on COMMON_CLK select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON -- 2.25.1

2 1

[PATCH] ASoC: mediatek: mt8192-mt6359: Set the driver name for the card
by Nícolas F. R. A. Prado 30 Sep '22

30 Sep '22

The ASoC core automatically populates the driver name field in the card from the card name if left unset. However, since the driver name can be at most 16 characters long, wrapping will happen if the card name is longer, which is the case for the mt8192-mt6359 driver. Explicitly set the driver name for the card in order to avoid said wrapping and have a readable driver name exposed to userspace. Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)collabora.com> --- Issue noticed while upstreaming the UCM file for this driver [1]. [1] https://github.com/alsa-project/alsa-ucm-conf/pull/217 sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index ff40ccd36f7e..b93c3237ef2d 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -24,6 +24,8 @@ #include "mt8192-afe-clk.h" #include "mt8192-afe-gpio.h" +#define DRIVER_NAME "mt8192_mt6359" + #define RT1015_CODEC_DAI "rt1015-aif" #define RT1015_DEV0_NAME "rt1015.1-0028" #define RT1015_DEV1_NAME "rt1015.1-0029" @@ -1070,6 +1072,7 @@ static struct snd_soc_codec_conf rt1015_amp_conf[] = { static struct snd_soc_card mt8192_mt6359_rt1015_rt5682_card = { .name = RT1015_RT5682_CARD_NAME, + .driver_name = DRIVER_NAME, .owner = THIS_MODULE, .dai_link = mt8192_mt6359_dai_links, .num_links = ARRAY_SIZE(mt8192_mt6359_dai_links), @@ -1105,6 +1108,7 @@ static const struct snd_kcontrol_new mt8192_mt6359_rt1015p_rt5682x_controls[] = }; static struct snd_soc_card mt8192_mt6359_rt1015p_rt5682x_card = { + .driver_name = DRIVER_NAME, .owner = THIS_MODULE, .dai_link = mt8192_mt6359_dai_links, .num_links = ARRAY_SIZE(mt8192_mt6359_dai_links), @@ -1266,7 +1270,7 @@ static const struct dev_pm_ops mt8192_mt6359_pm_ops = { static struct platform_driver mt8192_mt6359_driver = { .driver = { - .name = "mt8192_mt6359", + .name = DRIVER_NAME, #ifdef CONFIG_OF .of_match_table = mt8192_mt6359_dt_match, #endif -- 2.37.3

3 2

[PATCH] ALSA: usb-audio: Fix potential memory leaks
by Takashi Iwai 30 Sep '22

30 Sep '22

When the driver hits -ENOMEM at allocating a URB or a buffer, it aborts and goes to the error path that releases the all previously allocated resources. However, when -ENOMEM hits at the middle of the sync EP URB allocation loop, the partially allocated URBs might be left without released, because ep->nurbs is still zero at that point. Fix it by setting ep->nurbs at first, so that the error handler loops over the full URB list. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 36f753a28341..48a3843a08f1 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1268,6 +1268,7 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep) if (!ep->syncbuf) return -ENOMEM; + ep->nurbs = SYNC_URBS; for (i = 0; i < SYNC_URBS; i++) { struct snd_urb_ctx *u = &ep->urb[i]; u->index = i; @@ -1287,8 +1288,6 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep) u->urb->complete = snd_complete_urb; } - ep->nurbs = SYNC_URBS; - return 0; out_of_memory: -- 2.35.3

1 0

[PATCH] ALSA: usb-audio: Fix NULL dererence at error path
by Takashi Iwai 30 Sep '22

30 Sep '22

At an error path to release URB buffers and contexts, the driver might hit a NULL dererence for u->urb pointer, when u->buffer_size has been already set but the actual URB allocation failed. Fix it by adding the NULL check of urb. Also, make sure that buffer_size is cleared after the error path or the close. Cc: <stable(a)vger.kernel.org> Reported-by: Sabri N. Ferreiro <snferreiro1(a)gmail.com> Link: https://lore.kernel.org/r/CAKG+3NRjTey+fFfUEGwuxL-pi_=T4cUskYG9OzpzHytF+tzY… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index b2d0b42b581f..36f753a28341 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -95,12 +95,13 @@ static inline unsigned get_usb_high_speed_rate(unsigned int rate) */ static void release_urb_ctx(struct snd_urb_ctx *u) { - if (u->buffer_size) + if (u->urb && u->buffer_size) usb_free_coherent(u->ep->chip->dev, u->buffer_size, u->urb->transfer_buffer, u->urb->transfer_dma); usb_free_urb(u->urb); u->urb = NULL; + u->buffer_size = 0; } static const char *usb_error_string(int err) -- 2.35.3

1 0