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

[PATCH] ASoC: SOF: topology: No need to assign core ID if token parsing failed
by Peter Ujfalusi 07 Nov '22

07 Nov '22

Move the return value check before attempting to assign the core ID to the swidget since we are going to fail the sof_widget_ready() and free up swidget anyways. Fixes: 909dadf21aae ("ASoC: SOF: topology: Make DAI widget parsing IPC agnostic") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/topology.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 38855dd60617..6a0e7f3b5023 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1344,16 +1344,6 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, break; } - if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { - swidget->core = SOF_DSP_PRIMARY_CORE; - } else { - int core = sof_get_token_value(SOF_TKN_COMP_CORE_ID, swidget->tuples, - swidget->num_tuples); - - if (core >= 0) - swidget->core = core; - } - /* check token parsing reply */ if (ret < 0) { dev_err(scomp->dev, @@ -1365,6 +1355,16 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, return ret; } + if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { + swidget->core = SOF_DSP_PRIMARY_CORE; + } else { + int core = sof_get_token_value(SOF_TKN_COMP_CORE_ID, swidget->tuples, + swidget->num_tuples); + + if (core >= 0) + swidget->core = core; + } + /* bind widget to external event */ if (tw->event_type) { if (widget_ops[w->id].bind_event) { -- 2.38.1

2 1

[PATCH v1] ASoC: sof_es8336: reduce pop noise on speaker
by Zhu Ning 07 Nov '22

07 Nov '22

The Speaker GPIO needs to be turned on slightly behind the codec turned on. It also need to be turned off slightly before the codec turned down. Current code uses delay in DAPM_EVENT to do it but the mdelay delays the DAPM itself and thus has no effect. A delayed_work is added to turn on the speaker. The Speaker is turned off in .trigger since trigger is called slightly before the DAPM events. Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ------------ v1: cancel delayed work while disabling speaker. --- sound/soc/intel/boards/sof_es8336.c | 60 ++++++++++++++++++++++------- 1 file changed, 46 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index fbb42e54947a..70713e4b07dc 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -63,6 +63,7 @@ struct sof_es8336_private { struct snd_soc_jack jack; struct list_head hdmi_pcm_list; bool speaker_en; + struct delayed_work pcm_pop_work; }; struct sof_hdmi_pcm { @@ -111,6 +112,46 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk headset at mic1 port enabled\n"); } +static void pcm_pop_work_events(struct work_struct *work) +{ + struct sof_es8336_private *priv = + container_of(work, struct sof_es8336_private, pcm_pop_work.work); + + gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); + + if (quirk & SOF_ES8336_HEADPHONE_GPIO) + gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); + +} + +static int sof_8336_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + if (priv->speaker_en == false) + if (substream->stream == 0) { + cancel_delayed_work(&priv->pcm_pop_work); + gpiod_set_value_cansleep(priv->gpio_speakers, true); + } + break; + default: + return -EINVAL; + } + + return 0; +} + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -122,19 +163,7 @@ static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, priv->speaker_en = !SND_SOC_DAPM_EVENT_ON(event); - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); - - if (!(quirk & SOF_ES8336_HEADPHONE_GPIO)) - return 0; - - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); - + queue_delayed_work(system_wq, &priv->pcm_pop_work, msecs_to_jiffies(70)); return 0; } @@ -344,6 +373,7 @@ static int sof_es8336_hw_params(struct snd_pcm_substream *substream, /* machine stream operations */ static struct snd_soc_ops sof_es8336_ops = { .hw_params = sof_es8336_hw_params, + .trigger = sof_8336_trigger, }; static struct snd_soc_dai_link_component platform_component[] = { @@ -723,7 +753,8 @@ static int sof_es8336_probe(struct platform_device *pdev) } INIT_LIST_HEAD(&priv->hdmi_pcm_list); - + INIT_DELAYED_WORK(&priv->pcm_pop_work, + pcm_pop_work_events); snd_soc_card_set_drvdata(card, priv); if (mach->mach_params.dmic_num > 0) { @@ -752,6 +783,7 @@ static int sof_es8336_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + cancel_delayed_work(&priv->pcm_pop_work); gpiod_put(priv->gpio_speakers); device_remove_software_node(priv->codec_dev); put_device(priv->codec_dev); -- 2.37.2

2 1

[PATCH v4 0/3] Convert soundwire bindings to DT schema
by Srinivasa Rao Mandadapu 07 Nov '22

07 Nov '22

Convert soundwire bindings text file to DT schema and update device tree entries to follow strict dt-bindings. Changes Since V3: -- Remove subnode description and add appropriate pattern properties. -- Add interrput names in example. -- update some properties description. -- Revert minIteams change in previous version. -- Rebase to latest code base. -- Update dtsi node names, -- Remove redundant property in soundwire node. Srinivasa Rao Mandadapu (3): arm64: dts: qcom: Update soundwire slave node names arm64: dts: qcom: Remove redundant soundwire property dt-bindings: soundwire: Convert text bindings to DT Schema .../devicetree/bindings/soundwire/qcom,sdw.txt | 215 ------------------ .../bindings/soundwire/qcom,soundwire.yaml | 241 +++++++++++++++++++++ arch/arm64/boot/dts/qcom/qrb5165-rb5.dts | 4 +- arch/arm64/boot/dts/qcom/sc7280.dtsi | 1 - arch/arm64/boot/dts/qcom/sdm845-db845c.dts | 4 +- .../boot/dts/qcom/sdm850-lenovo-yoga-c630.dts | 4 +- arch/arm64/boot/dts/qcom/sdm850-samsung-w737.dts | 4 +- arch/arm64/boot/dts/qcom/sm8250-mtp.dts | 4 +- arch/arm64/boot/dts/qcom/sm8250.dtsi | 1 - 9 files changed, 251 insertions(+), 227 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,soundwire.yaml -- 2.7.4

3 9

ALC1220 DualCodec no optical out
by GitHub issues - opened 07 Nov '22

07 Nov '22

alsa-project/alsa-ucm-conf issue #239 was opened from Almoped: With current pipewire and alsa-ucm 1.2.8 I am unable to select profiles which has iec958 output, actually I am unable to select any card profile at all in pavucontrol. This is possibly because all the electrical output jacks are disconnected and optical out does not have jackdetect. Or maybe DualCodecs/HiFi.conf does not configure optical out. If I remove alsa/ucm2/HDA/DualCodecs/HiFi.conf it uses alsa-card-profile/mixer/profile-sets/default.conf, goes on a probing mission I get a working optical out and the possibility to select an array of other card profiles in pavucontrol. Can anyone recommend a better way to solve this? I've attached pactl list cards output with working and nonworking optical out. [pactl_no_optical.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9… [pactl_working_optical.txt](https://github.com/alsa-project/alsa-ucm-conf/fi… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/239 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: dapm: Don't use prefix for regulator name
by Paul Cercueil 07 Nov '22

07 Nov '22

When a component has a prefix, and uses a SND_SOC_DAPM_REGULATOR_SUPPLY, the name of the regulator should not use the prefix, otherwise it won't be properly matched in the DT/ACPI. Fixes: 3caac759681e ("ASoC: soc-dapm.c: fixup snd_soc_dapm_new_control_unlocked() error handling") Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d515e7a78ea8..879cf1be67a9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3645,7 +3645,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, switch (w->id) { case snd_soc_dapm_regulator_supply: - w->regulator = devm_regulator_get(dapm->dev, w->name); + w->regulator = devm_regulator_get(dapm->dev, widget->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); goto request_failed; -- 2.35.1

2 4

[GIT PULL] treewide: timers: Use timer_shutdown*() before freeing timers
by Steven Rostedt 07 Nov '22

07 Nov '22

Linus, As discussed here: https://lore.kernel.org/all/20221106212427.739928660@goodmis.org/ Add a "shutdown" state for timers. This is performed by the new timer_shutdown_sync() and timer_shutdown() function calls. When this is called on a timer, it will no longer be able to be re-armed. This should be called before a timer is freed to prevent it from being re-armed after being removed from the timer queue and then causing a crash in the timer code when the timer triggers. This required renaming some functions that were using the name timer_shutdown() statically to something more appropriate. Then a coccinelle script was executed on the entire kernel tree to find the trivial locations that remove the timer and then frees the object that the timer exists on. These changes are not enough to solve all the locations where timers may be of an issue. But by adding the shutdown infrastructure and the obvious cases, the more complex cases can be added after they have been reviewed more closely. Please pull the following tree, which can be found at: git://git.kernel.org/pub/scm/linux/kernel/git/rostedt/linux-trace.git add-timer-shutdown Tag SHA1: 7685328352dfd2908e23048f563e328dbd3526e9 Head SHA1: 870556da63870e01ade9bb8418ac5a21862f2f10 Steven Rostedt (Google) (5): ARM: spear: Do not use timer namespace for timer_shutdown() function clocksource/drivers/arm_arch_timer: Do not use timer namespace for timer_shutdown() function clocksource/drivers/sp804: Do not use timer namespace for timer_shutdown() function timers: Add timer_shutdown_sync() and timer_shutdown() to be called before freeing timers treewide: Convert del_timer*() to timer_shutdown*() ---- .../RCU/Design/Requirements/Requirements.rst | 2 +- Documentation/core-api/local_ops.rst | 2 +- Documentation/kernel-hacking/locking.rst | 5 ++ arch/arm/mach-spear/time.c | 8 +-- arch/sh/drivers/push-switch.c | 2 +- block/blk-iocost.c | 2 +- block/blk-iolatency.c | 2 +- block/kyber-iosched.c | 2 +- drivers/acpi/apei/ghes.c | 2 +- drivers/atm/idt77252.c | 6 +- drivers/block/drbd/drbd_main.c | 2 +- drivers/block/loop.c | 2 +- drivers/bluetooth/hci_bcsp.c | 2 +- drivers/bluetooth/hci_qca.c | 4 +- drivers/clocksource/arm_arch_timer.c | 12 ++-- drivers/clocksource/timer-sp804.c | 6 +- drivers/gpu/drm/i915/i915_sw_fence.c | 2 +- drivers/hid/hid-wiimote-core.c | 2 +- drivers/input/keyboard/locomokbd.c | 2 +- drivers/input/keyboard/omap-keypad.c | 2 +- drivers/input/mouse/alps.c | 2 +- drivers/isdn/mISDN/l1oip_core.c | 4 +- drivers/isdn/mISDN/timerdev.c | 4 +- drivers/leds/trigger/ledtrig-activity.c | 2 +- drivers/leds/trigger/ledtrig-heartbeat.c | 2 +- drivers/leds/trigger/ledtrig-pattern.c | 2 +- drivers/leds/trigger/ledtrig-transient.c | 2 +- drivers/media/pci/ivtv/ivtv-driver.c | 2 +- drivers/media/usb/pvrusb2/pvrusb2-hdw.c | 16 +++--- drivers/media/usb/s2255/s2255drv.c | 4 +- drivers/net/ethernet/intel/i40e/i40e_main.c | 6 +- drivers/net/ethernet/marvell/sky2.c | 2 +- drivers/net/ethernet/sun/sunvnet.c | 2 +- drivers/net/usb/sierra_net.c | 2 +- .../wireless/broadcom/brcm80211/brcmfmac/btcoex.c | 2 +- drivers/net/wireless/intel/iwlwifi/iwl-dbg-tlv.c | 2 +- drivers/net/wireless/intel/iwlwifi/mvm/sta.c | 2 +- drivers/net/wireless/intersil/hostap/hostap_ap.c | 2 +- drivers/net/wireless/marvell/mwifiex/main.c | 2 +- drivers/net/wireless/microchip/wilc1000/hif.c | 6 +- drivers/nfc/pn533/pn533.c | 2 +- drivers/nfc/pn533/uart.c | 2 +- drivers/pcmcia/bcm63xx_pcmcia.c | 2 +- drivers/pcmcia/electra_cf.c | 2 +- drivers/pcmcia/omap_cf.c | 2 +- drivers/pcmcia/pd6729.c | 4 +- drivers/pcmcia/yenta_socket.c | 4 +- drivers/scsi/qla2xxx/qla_edif.c | 4 +- drivers/staging/media/atomisp/i2c/atomisp-lm3554.c | 2 +- drivers/tty/n_gsm.c | 2 +- drivers/tty/sysrq.c | 2 +- drivers/usb/gadget/udc/m66592-udc.c | 2 +- drivers/usb/serial/garmin_gps.c | 2 +- drivers/usb/serial/mos7840.c | 4 +- fs/ext4/super.c | 2 +- fs/nilfs2/segment.c | 2 +- include/linux/timer.h | 62 +++++++++++++++++++-- kernel/time/timer.c | 64 ++++++++++++---------- net/802/garp.c | 2 +- net/802/mrp.c | 4 +- net/bridge/br_multicast.c | 8 +-- net/bridge/br_multicast_eht.c | 4 +- net/core/gen_estimator.c | 2 +- net/ipv4/ipmr.c | 2 +- net/ipv6/ip6mr.c | 2 +- net/mac80211/mesh_pathtbl.c | 2 +- net/netfilter/ipset/ip_set_list_set.c | 2 +- net/netfilter/ipvs/ip_vs_lblc.c | 2 +- net/netfilter/ipvs/ip_vs_lblcr.c | 2 +- net/netfilter/xt_IDLETIMER.c | 4 +- net/netfilter/xt_LED.c | 2 +- net/rxrpc/conn_object.c | 2 +- net/sched/cls_flow.c | 2 +- net/sunrpc/svc.c | 2 +- net/tipc/discover.c | 2 +- net/tipc/monitor.c | 2 +- sound/i2c/other/ak4117.c | 2 +- sound/synth/emux/emux.c | 2 +- 78 files changed, 207 insertions(+), 148 deletions(-) ---------------------------

2 1

[PATCH] ASoC: wm8962: Wait for updated value of WM8962_CLOCKING1 register
by Chancel Liu 07 Nov '22

07 Nov '22

DSPCLK_DIV field in WM8962_CLOCKING1 register is used to generate correct frequency of LRCLK and BCLK. Sometimes the read-only value can't be updated timely after enabling SYSCLK. This results in wrong calculation values. Delay is introduced here to wait for newest value from register. The time of the delay should be at least 500~1000us according to test. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/codecs/wm8962.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b4b4355c6728..563843565f68 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2503,6 +2503,14 @@ static void wm8962_configure_bclk(struct snd_soc_component *component) snd_soc_component_update_bits(component, WM8962_CLOCKING2, WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA); + /* DSPCLK_DIV field in WM8962_CLOCKING1 register is used to generate + * correct frequency of LRCLK and BCLK. Sometimes the read-only value + * can't be updated timely after enabling SYSCLK. This results in wrong + * calculation values. Delay is introduced here to wait for newest + * value from register. The time of the delay should be at least + * 500~1000us according to test. + */ + msleep(1); dspclk = snd_soc_component_read(component, WM8962_CLOCKING1); if (snd_soc_component_get_bias_level(component) != SND_SOC_BIAS_ON) -- 2.25.1

2 1

Question regarding delayed trigger in dpcm_set_fe_update_state()
by Cezary Rojewski 07 Nov '22

07 Nov '22

Hello, Writing with question regarding dpcm_set_fe_update_state() function which is part of sound/soc/soc-pcm.c file and has been introduced with commit "ASoC: dpcm: Fix race between FE/BE updates and trigger" [1]. The part that concerns me is the invocation of dpcm_fe_dai_do_trigger() regardless of the actual state given DPCM is in (actual state, not the DPCM_UPDATE_XX). The conditional invocation of said _trigger() and addition of .trigger_pending field are here to address a race where PCM state is being modified from multiple locations simultaneously, at least judging by the commit's description. Note that the dpcm_set_fe_update_state() is called from all the dai-ops i.e.: startup, shutdown, hw_params, prepare and hw_free. Now, given that knowledge, we could end up in scenario where dpcm_fe_dai_do_trigger() is invoked as a part of dpcm_fe_dai_hw_free(). dpcm_set_fe_update_state() is called there twice, once with DPCM_UPDATE_FE and once with DPCM_UPDATE_NO. The second case is the more interesting one since it's called **after** ->hw_free() callback is invoked for all the DAIs. dpcm_fe_dai_hw_free() dpcm_set_fe_update_state(DPCM_UPDATE_FE) // fine (...) dpcm_be_dai_hw_free() // data allocated in hw_params // is freed here (...) dpcm_set_fe_update_state(DPCM_UPDATE_NO) // not fine The last is *not fine* if the .trigger_pending is not a zero, and can lead to panic as code used during ->trigger() is often manipulating data allocated during ->hw_params() but that data has just been freed with ->hw_free(). Is what I'm looking at a bug? Or, perhaps there's something I'm missing in the picture. From my study, it seems that only dpcm_fe_dai_prepare() is a safe place for calling dpcm_fe_dai_do_trigger() - once .trigger_pending is taken into account. Any input is much appreciated. [1]: https://lore.kernel.org/all/1415116348-11792-1-git-send-email-tiwai@suse.de… Regards, Czarek

2 3

回复: Re: 回复: Re: 回复: Re: [PATCH] ALSA: usb-audio: fix Hamedal C20 usb camero disconnect issue
by 艾超 07 Nov '22

07 Nov '22

I tested the hotplug ãrecord and catch the picture, this interface reset doesn't cause any problem. In snd_usb_endpoint_configure, It set false to interface to deselect altsetting at first. But , if skip the interface clear (ex. ep->cur_rate = 0;), there is no sound output. ---- ä¸»ãé¢ï¼Re: åå¤: Re: åå¤: Re: [PATCH] ALSA: usb-audio: fix Hamedal C20 usb camero disconnect issue æ¥ãæï¼2022-11-07 16:15 åä»¶äººï¼Takashi Iwai æ¶ä»¶äººï¼è¾è¶ On Mon, 07 Nov 2022 02:33:57 +0100, è¾è¶ wrote: > > Yes, the audio device is a UAC1 device, and it used set_sample_rate_v1 to > set sample. > > So, I add a quirk to keep up the USB interface up even if the audio stream > is closed. Well, my question is also about the parameter changes. This interface reset doesn't cause any problem? Or we should skip the interface clear there, too? Takashi > > ---- > > ä¸»ãé¢ï¼Re: åå¤: Re: [PATCH] ALSA: usb-audio: fix Hamedal C20 usb camero > disconnect issue > æ¥ãæï¼2022-11-04 17:59 > åä»¶äººï¼Takashi Iwai > æ¶ä»¶äººï¼è¾è¶ > > On Fri, 04 Nov 2022 10:50:48 +0100, > è¾è¶ wrote: > > > > I 'm sorry, I don't know whether you receiver my mail for 'lsusb -v' to > > Hamedal C20 > > I received your previous mails including lsusb -v output. But the > post was filtered out by some reason? > > So the audio device looks like a UAC1 device. And this one requires > to keep up the USB interface up even if the audio stream is closed > (not about the parameter changes)? > > Takashi > >

1 0

[PATCH] ALSA: usb-audio: Add quirks for MacroSilicon MS2100/MS2106 devices (5.10)
by John Veness 07 Nov '22

07 Nov '22

[ Upstream commit 6e2c9105e0b743c92a157389d40f00b81bdd09fe] Backported to 5.10. Treat the claimed 96kHz 1ch in the descriptors as 48kHz 2ch, so that the audio stream doesn't sound mono. Also fix initial stream alignment, so that left and right channels are in the correct order. Signed-off-by: John Veness <john-linux(a)pelago.org.uk> Cc: <stable(a)vger.kernel.org> # 5.10 --- sound/usb/quirks-table.h | 52 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 1 + 2 files changed, 53 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 1ac91c46da3c..a51591f68ae6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3656,6 +3656,58 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* + * MacroSilicon MS2100/MS2106 based AV capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. + */ +{ + USB_AUDIO_DEVICE(0x534d, 0x0021), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS210x", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, + /* * MacroSilicon MS2109 based HDMI capture cards * diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 41f5d8242478..04a691bc560c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1508,6 +1508,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ pioneer_djm_set_format_quirk(subs); break; + case USB_ID(0x534d, 0x0021): /* MacroSilicon MS2100/MS2106 */ case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; -- 2.34.1

2 1