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

Re: wm8960: add DAC Slope switch
by Charles Keepax 07 Dec '20

07 Dec '20

On Tue, Nov 24, 2020 at 06:23:13PM +0100, Lukas F. Hartmann wrote: > The WM8960 DAC has a "DAC Slope" switch that can toggle between two > different output filter curves. This patch adds support for it. > > Signed-off-by: Lukas F. Hartmann <lukas(a)mntre.com> > --- > +static int wm8960_put_dacslope(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); > + struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component); > + unsigned int val = ucontrol->value.integer.value[0]; > + > + if (val > 1) > + return -EINVAL; > + > + wm8960->dacslope = val; > + > + return snd_soc_component_update_bits(component, WM8960_DACCTL1, > + 0x2, val<<1); Is the DAC Slope switch not in DACCTL2? Thanks, Charles

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snd-rn-pci-acp3x - auto dmic detection unreliable
by Jaroslav Kysela 07 Dec '20

07 Dec '20

Hi, it seems that the ACPI auto-detection of the presence the digital microphone connected to AMD Renoir audio bridge is unreliable. It may be that hardware vendors do not provide the correct ACPI tables. Could you clarify the usage of _WOV integer value? It seems that this value is set on those plaforms which do not have the digital microphone (the microphone is connected to the HDA codec): Lenovo IdeaPad 5 (15ARE05) Lenovo Thinkpad X395 In any case, we should probably improve the auto-detection using DMI or other platform info. Jaroslav BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1892115 -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[PATCH] ALSA: hda/realtek - Enable headset mic of ASUS X430UN with ALC256
by Chris Chiu 07 Dec '20

07 Dec '20

The ASUS laptop X430UN with ALC256 can't detect the headset microphone until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> --- 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 f2398721ac1e..d5e4d0ba1008 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7827,6 +7827,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), -- 2.20.1

2 1

rt715: init: setup ADC07 to a proper volume
by GitHub pull_request - opened 07 Dec '20

07 Dec '20

alsa-project/alsa-ucm-conf pull request #65 was opened from libinyang: This patch initiates 'rt715 ADC 07 Capture Volume' to a proper volume. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/65 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/65.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: rawmidi: Access runtime->avail always in spinlock
by Takashi Iwai 06 Dec '20

06 Dec '20

The runtime->avail field may be accessed concurrently while some places refer to it without taking the runtime->lock spinlock, as detected by KCSAN. Usually this isn't a big problem, but for consistency and safety, we should take the spinlock at each place referencing this field. Reported-by: syzbot+a23a6f1215c84756577c(a)syzkaller.appspotmail.com Reported-by: syzbot+3d367d1df1d2b67f5c19(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/rawmidi.c | 49 +++++++++++++++++++++++++++++++------------- 1 file changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c78720a3299c..257ad5206240 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -95,11 +95,21 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file) } } -static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream) +static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime) +{ + return runtime->avail >= runtime->avail_min; +} + +static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + bool ready; - return runtime->avail >= runtime->avail_min; + spin_lock_irqsave(&runtime->lock, flags); + ready = __snd_rawmidi_ready(runtime); + spin_unlock_irqrestore(&runtime->lock, flags); + return ready; } static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream, @@ -1019,7 +1029,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, if (result > 0) { if (runtime->event) schedule_work(&runtime->event_work); - else if (snd_rawmidi_ready(substream)) + else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } spin_unlock_irqrestore(&runtime->lock, flags); @@ -1098,7 +1108,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun result = 0; while (count > 0) { spin_lock_irq(&runtime->lock); - while (!snd_rawmidi_ready(substream)) { + while (!__snd_rawmidi_ready(runtime)) { wait_queue_entry_t wait; if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { @@ -1115,9 +1125,11 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_read1(substream, @@ -1255,7 +1267,7 @@ int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int coun runtime->avail += count; substream->bytes += count; if (count > 0) { - if (runtime->drain || snd_rawmidi_ready(substream)) + if (runtime->drain || __snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } return count; @@ -1444,9 +1456,11 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail && !timeout) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail && !timeout) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count); @@ -1526,6 +1540,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; + unsigned long buffer_size, avail, xruns; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1544,13 +1559,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Mode : %s\n" " Buffer size : %lu\n" " Avail : %lu\n", runtime->oss ? "OSS compatible" : "native", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail); + buffer_size, avail); } } } @@ -1568,13 +1586,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + xruns = runtime->xruns; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Buffer size : %lu\n" " Avail : %lu\n" " Overruns : %lu\n", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail, - (unsigned long) runtime->xruns); + buffer_size, avail, xruns); } } } -- 2.26.2

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[PATCH] ALSA: seq: Use bool for snd_seq_queue internal flags
by Takashi Iwai 06 Dec '20

06 Dec '20

The snd_seq_queue struct contains various flags in the bit fields. Those are categorized to two different use cases, both of which are protected by different spinlocks. That implies that there are still potential risks of the bad operations for bit fields by concurrent accesses. For addressing the problem, this patch rearranges those flags to be a standard bool instead of a bit field. Reported-by: syzbot+63cbe31877bb80ef58f5(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_queue.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 1c3a8d3254d9..c69105dc1a10 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -26,10 +26,10 @@ struct snd_seq_queue { struct snd_seq_timer *timer; /* time keeper for this queue */ int owner; /* client that 'owns' the timer */ - unsigned int locked:1, /* timer is only accesibble by owner if set */ - klocked:1, /* kernel lock (after START) */ - check_again:1, - check_blocked:1; + bool locked; /* timer is only accesibble by owner if set */ + bool klocked; /* kernel lock (after START) */ + bool check_again; /* concurrent access happened during check */ + bool check_blocked; /* queue being checked */ unsigned int flags; /* status flags */ unsigned int info_flags; /* info for sync */ -- 2.26.2

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[PATCH 0/1] dt-bindings: eliminate yamllint warnings
by Zhen Lei 06 Dec '20

06 Dec '20

There're too many people, I just send to the maintainer, reviewer, supporter. Eliminate below warnings: ./Documentation/devicetree/bindings/clock/imx8qxp-lpcg.yaml:32:13: [warning] wrong indentation: expected 14 but found 12 (indentation) ./Documentation/devicetree/bindings/clock/imx8qxp-lpcg.yaml:35:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/display/intel,keembay-msscam.yaml:21:6: [warning] wrong indentation: expected 6 but found 5 (indentation) ./Documentation/devicetree/bindings/display/bridge/analogix,anx7625.yaml:52:9: [warning] wrong indentation: expected 6 but found 8 (indentation) ./Documentation/devicetree/bindings/display/bridge/intel,keembay-dsi.yaml:42:8: [warning] wrong indentation: expected 8 but found 7 (indentation) ./Documentation/devicetree/bindings/display/bridge/intel,keembay-dsi.yaml:45:8: [warning] wrong indentation: expected 8 but found 7 (indentation) ./Documentation/devicetree/bindings/display/panel/novatek,nt36672a.yaml:25:10: [warning] wrong indentation: expected 10 but found 9 (indentation) ./Documentation/devicetree/bindings/media/i2c/adv7604.yaml:24:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml:4:1: [error] missing document start "---" (document-start) ./Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml:29:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/media/i2c/mipi-ccs.yaml:32:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/media/i2c/ovti,ov772x.yaml:79:17: [warning] wrong indentation: expected 14 but found 16 (indentation) ./Documentation/devicetree/bindings/media/i2c/ovti,ov772x.yaml:88:17: [warning] wrong indentation: expected 14 but found 16 (indentation) ./Documentation/devicetree/bindings/media/i2c/sony,imx214.yaml:72:17: [warning] wrong indentation: expected 18 but found 16 (indentation) ./Documentation/devicetree/bindings/media/i2c/sony,imx214.yaml:75:17: [warning] wrong indentation: expected 18 but found 16 (indentation) ./Documentation/devicetree/bindings/mmc/mtk-sd.yaml:20:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/mmc/mtk-sd.yaml:30:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/mmc/mtk-sd.yaml:33:9: [warning] wrong indentation: expected 10 but found 8 (indentation) ./Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml:10:4: [warning] wrong indentation: expected 2 but found 3 (indentation) Zhen Lei (1): dt-bindings: eliminate yamllint warnings .../devicetree/bindings/clock/imx8qxp-lpcg.yaml | 20 ++++++++--------- .../bindings/display/bridge/analogix,anx7625.yaml | 4 ++-- .../bindings/display/bridge/intel,keembay-dsi.yaml | 4 ++-- .../bindings/display/intel,keembay-msscam.yaml | 4 ++-- .../bindings/display/panel/novatek,nt36672a.yaml | 2 +- .../devicetree/bindings/media/i2c/adv7604.yaml | 4 ++-- .../devicetree/bindings/media/i2c/mipi-ccs.yaml | 11 ++++----- .../devicetree/bindings/media/i2c/ovti,ov772x.yaml | 12 +++++----- .../devicetree/bindings/media/i2c/sony,imx214.yaml | 12 +++++----- Documentation/devicetree/bindings/mmc/mtk-sd.yaml | 26 +++++++++++----------- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 4 ++-- 11 files changed, 52 insertions(+), 51 deletions(-) -- 1.8.3

4 5

thinkpad t14s amd: unable to control built-in mic and headphones mic independently
by GitHub issues - opened 05 Dec '20

05 Dec '20

alsa-project/alsa-ucm-conf issue #64 was opened from tanuk: I did some investigation on a [pulseaudio bug report](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1067), and found that the UCM configuration for the internal mic and headset mic both use "Capture" as the CaptureMixerElem. This means that muting one device mutes also the other (and I guess volume control is shared too). In amixer output I don't see a separate control for the headset mic, but for the internal mic volume I guess "Digital" can be used (there's no switch, though). So it seems that these changes are needed: - don't set CaptureMixerElem, CaptureSwitch or CaptureVolume for the headset mic - set the internal mic CaptureMixerElem and CaptureVolume to Digital - you may want to also set CaptureMasterElem for both both mics to Capture Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/64 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: Behringer WING usb audio - cyclic xruns dependent on periods/buffers
by Takashi Iwai 05 Dec '20

05 Dec '20

On Thu, 03 Dec 2020 21:06:24 +0100, Ben Bell wrote: > > On Sat, Nov 28, 2020 at 09:36:00AM +0000, Ben Bell wrote: > > > In general you should avoid 44.1kHz if you want a small period size > > > for a realtime process on USB-audio. With 44.1kHz, the packet size > > > can't be fixed in integer, and the ISO transfer requires variable > > > packet sizes. OTOH, ALSA API requires the fixed period size, hence > > > it'll lead to inconsistencies occasionally. > > So changing to 48kHz had no appreciable effect, but I'm working at that > rate for now to eliminate the 44.1kHz weirdness from investigations. > Using a USB2 card, also had no effect. Testing with a different USB audio > interface (albeit a simple stereo one) didn't exhibit the behaviour, even > when I took buffer sizes right down. > > I'm not sure where to go to get more information or where's the best place > to ask for help. I'm happy to do the leg work, but I don't know enough > about the kernel, alsa or USB to figure it out without some help. > > Current question: what is the delay in /proc/asound/card1/pcm0c/sub0/status > actually measuring? I'm assuming it's measured in samples? I've written > something to scrape the stats out in a tight loop and report. > > What I see is a cycle where the delay rises and then a chunk equal to the > frames per period (or sometimes, earlier on fpp-48) is removed. That feels > like chunks being read out of a buffer. All fine. > > After a while though, the maximum delay we reach with each cycle is creeping > up. It increases by one every few cycles (usually two or three, or three or > four -- always oscillating between two values) but of it's still only being > emptied by a full period's worth of samples each cycle. So the overall effect > is the delay creeps up and up until it hits the buffer size and then we get > an xrun. > > Like I said in the initial email, it feels like some sort of clock drift > problem, where we're managing very slowly to collect more samples than > we're reading -- to the tune of about 1 extra every few cycles -- and > nothing on the consumer side is ever managing to compensate for that. > I'm not even sure how that sort of drift would be possible though. Seems > surprising. > > Does any of this sound suspicious, or for that matter completely normal? > Any suggestions where should I be looking next? At least you can try the latest patch set destined for 5.11, which should improve the cases for the implicit feedback. The patches are either in linux-next tree or the topic/usb-audio-refactoring branch of my sound.git tree. It's based on 5.10-rc, so should be cleanly mergeable to the latest Linus tree. Takashi

2 1

[PATCH] slimbus: qcom-ngd-ctrl: Fix error handling to keep reference balanced
by Zhang Qilong 05 Dec '20

05 Dec '20

In the error branch, we should call pm_runtime_put_noidle and pm_runtime_disable to decrease usage counter and increase the power disable depth to avoid reference leaks. Fixes: 917809e2280bb ("slimbus: ngd: Add qcom SLIMBus NGD driver") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index c054e83ab636..a9b4a1d95045 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1483,7 +1483,7 @@ static int qcom_slim_ngd_probe(struct platform_device *pdev) ret = qcom_slim_ngd_qmi_svc_event_init(ctrl); if (ret) { dev_err(&pdev->dev, "QMI service registration failed:%d", ret); - return ret; + goto pm_err; } INIT_WORK(&ctrl->m_work, qcom_slim_ngd_master_worker); @@ -1500,6 +1500,9 @@ static int qcom_slim_ngd_probe(struct platform_device *pdev) qcom_slim_ngd_qmi_svc_event_deinit(&ctrl->qmi); if (ctrl->mwq) destroy_workqueue(ctrl->mwq); +pm_err: + pm_runtime_put_noidle(dev); + pm_runtime_disable(dev); return ret; } -- 2.25.4

1 0

[PATCH] soundwire: intel: fix another unused-function warning
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> Without CONFIG_PM, there is another warning about an unused function: drivers/soundwire/intel.c:530:12: error: 'intel_link_power_down' defined but not used [-Werror=unused-function] After a previous fix, the driver already uses both an #ifdef and a __maybe_unused annotation but still gets it wrong. Remove the ifdef and instead use __maybe_unused consistently to avoid the problem for good. Fixes: f046b2334083 ("soundwire: intel: fix intel_suspend/resume defined but not used warning") Fixes: ebf878eddbb4 ("soundwire: intel: add pm_runtime support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- drivers/soundwire/intel.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 6a1e862b16c3..66adb258a425 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1585,8 +1585,6 @@ int intel_master_process_wakeen_event(struct platform_device *pdev) * PM calls */ -#ifdef CONFIG_PM - static int __maybe_unused intel_suspend(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); @@ -1641,7 +1639,7 @@ static int __maybe_unused intel_suspend(struct device *dev) return 0; } -static int intel_suspend_runtime(struct device *dev) +static int __maybe_unused intel_suspend_runtime(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); @@ -1796,7 +1794,7 @@ static int __maybe_unused intel_resume(struct device *dev) return ret; } -static int intel_resume_runtime(struct device *dev) +static int __maybe_unused intel_resume_runtime(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); @@ -1969,8 +1967,6 @@ static int intel_resume_runtime(struct device *dev) return ret; } -#endif - static const struct dev_pm_ops intel_pm = { SET_SYSTEM_SLEEP_PM_OPS(intel_suspend, intel_resume) SET_RUNTIME_PM_OPS(intel_suspend_runtime, intel_resume_runtime, NULL) -- 2.27.0

3 2

[PATCH v2] ALSA: hda/realtek: make bass spk volume adjustable on a yoga laptop
by Hui Wang 05 Dec '20

05 Dec '20

This change could fix 2 issues on this machine: - the bass speaker's output volume can't be adjusted, that is because the bass speaker is routed to the DAC (Nid 0x6) which has no volume control. - after plugging a headset with vol+, vol- and pause buttons on it, press those buttons, nothing happens, this means those buttons don't work at all. This machine has alc287 codec, need to add the codec id to the disable/enable_headset_jack_key(), then the headset button could work. The quirk of ALC285_FIXUP_THINKPAD_HEADSET_JACK could fix both of these 2 issues. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..5a905fa1b33a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3104,6 +3104,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -3130,6 +3131,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -8578,6 +8580,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_HEADSET_JACK, + {0x14, 0x90170110}, + {0x17, 0x90170111}, + {0x19, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x17, 0x90170110}, -- 2.25.1

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[PATCH] ALSA: hda/realtek: make bass spk volume adjustable on a yoga laptop
by Hui Wang 05 Dec '20

05 Dec '20

This also make the headset button work on this machine. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..5a905fa1b33a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3104,6 +3104,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -3130,6 +3131,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -8578,6 +8580,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_HEADSET_JACK, + {0x14, 0x90170110}, + {0x17, 0x90170111}, + {0x19, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x17, 0x90170110}, -- 2.25.1

2 2

[PATCH] ASoC: SOF: control: fix cppcheck warning in snd_sof_volume_info()
by Kai Vehmanen 05 Dec '20

05 Dec '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Fix cppcheck warning: sound/soc/sof/control.c:117:82: style:inconclusive: Function 'snd_sof_volume_info' argument 2 names different: declaration 'ucontrol' definition 'uinfo'. [funcArgNamesDifferent] Fixes: fca18e62984a ("ASoC: SOF: control: override volume info callback") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/sof-audio.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index d00a918b6f39..dc930fc2f4b5 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -125,7 +125,7 @@ int snd_sof_volume_get(struct snd_kcontrol *kcontrol, int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_sof_volume_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *ucontrol); + struct snd_ctl_elem_info *uinfo); int snd_sof_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_sof_switch_put(struct snd_kcontrol *kcontrol, -- 2.28.0

2 1

[PATCH] ASoC: q6afe-clocks: Add missing parent clock rate
by Srinivas Kandagatla 05 Dec '20

05 Dec '20

setting clock rate on child clocks without a parent clock rate will result in zero clk rate for child. This also means that when audio is started dsp will attempt to access registers without enabling clock resulting in board boot up. Fix this by adding the missing parent clock rate. Fixes: 520a1c396d196 ("ASoC: q6afe-clocks: add q6afe clock controller") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-clocks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/qdsp6/q6afe-clocks.c b/sound/soc/qcom/qdsp6/q6afe-clocks.c index 87e4633afe2c..f0362f061652 100644 --- a/sound/soc/qcom/qdsp6/q6afe-clocks.c +++ b/sound/soc/qcom/qdsp6/q6afe-clocks.c @@ -16,6 +16,7 @@ .afe_clk_id = Q6AFE_##id, \ .name = #id, \ .attributes = LPASS_CLK_ATTRIBUTE_COUPLE_NO, \ + .rate = 19200000, \ .hw.init = &(struct clk_init_data) { \ .ops = &clk_q6afe_ops, \ .name = #id, \ -- 2.21.0

2 1

[PATCH] ASoC: qcom: fix QDSP6 dependencies, attempt #3
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> The previous fix left another warning in randconfig builds: 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 [=y] && COMMON_CLK [=n] Selected by [y]: - SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && QCOM_APR [=y] Add one more dependency for this one. Fixes: 2bc8831b135c ("ASoC: qcom: fix SDM845 & QDSP6 dependencies more") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 27f93006be96..cc7c1de2f1d9 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -106,6 +106,7 @@ config SND_SOC_QDSP6 config SND_SOC_MSM8996 tristate "SoC Machine driver for MSM8996 and APQ8096 boards" depends on QCOM_APR + depends on COMMON_CLK select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON help -- 2.27.0

3 2

[PATCH] ASoC: cros_ec_codec: fix uninitialized memory read
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> gcc points out a memory area that is copied to a device but not initialized: sound/soc/codecs/cros_ec_codec.c: In function 'i2s_rx_event': arch/x86/include/asm/string_32.h:83:20: error: '*((void *)&p+4)' may be used uninitialized in this function [-Werror=maybe-uninitialized] 83 | *((int *)to + 1) = *((int *)from + 1); Initialize all the unused fields to zero. Fixes: 727f1c71c780 ("ASoC: cros_ec_codec: refactor I2S RX") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cros_ec_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 58894bf47514..f33a2a9654e7 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -332,7 +332,7 @@ static int i2s_rx_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_to_component(w->dapm); struct cros_ec_codec_priv *priv = snd_soc_component_get_drvdata(component); - struct ec_param_ec_codec_i2s_rx p; + struct ec_param_ec_codec_i2s_rx p = {}; switch (event) { case SND_SOC_DAPM_PRE_PMU: -- 2.27.0

3 4

[PATCH] ASoC: atmel: mchp-spdifrx needs COMMON_CLK
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> Compile-testing this driver on an older platform without CONFIG_COMMON_CLK fails with ERROR: modpost: "clk_set_min_rate" [sound/soc/atmel/snd-soc-mchp-spdifrx.ko] undefined! Make this is a strict dependency. Fixes: ef265c55c1ac ("ASoC: mchp-spdifrx: add driver for SPDIF RX") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/atmel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index bd8854bfd2ee..142373ec411a 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -148,6 +148,7 @@ config SND_MCHP_SOC_SPDIFTX config SND_MCHP_SOC_SPDIFRX tristate "Microchip ASoC driver for boards using S/PDIF RX" depends on OF && (ARCH_AT91 || COMPILE_TEST) + depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help -- 2.27.0

3 2

[PATCH] ASoC: fsl_aud2htx: mark PM functions as __maybe_unused
by Arnd Bergmann 05 Dec '20

05 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> When CONFIG_PM is disabled, we get a warning for unused functions: sound/soc/fsl/fsl_aud2htx.c:261:12: error: unused function 'fsl_aud2htx_runtime_suspend' [-Werror,-Wunused-function] static int fsl_aud2htx_runtime_suspend(struct device *dev) sound/soc/fsl/fsl_aud2htx.c:271:12: error: unused function 'fsl_aud2htx_runtime_resume' [-Werror,-Wunused-function] static int fsl_aud2htx_runtime_resume(struct device *dev) Mark these as __maybe_unused to avoid the warning without adding an #ifdef. Fixes: 8a24c834c053 ("ASoC: fsl_aud2htx: Add aud2htx module driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/fsl/fsl_aud2htx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 4091ccc7c3e9..d70d5e75a30c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -258,7 +258,7 @@ static int fsl_aud2htx_remove(struct platform_device *pdev) return 0; } -static int fsl_aud2htx_runtime_suspend(struct device *dev) +static int __maybe_unused fsl_aud2htx_runtime_suspend(struct device *dev) { struct fsl_aud2htx *aud2htx = dev_get_drvdata(dev); @@ -268,7 +268,7 @@ static int fsl_aud2htx_runtime_suspend(struct device *dev) return 0; } -static int fsl_aud2htx_runtime_resume(struct device *dev) +static int __maybe_unused fsl_aud2htx_runtime_resume(struct device *dev) { struct fsl_aud2htx *aud2htx = dev_get_drvdata(dev); int ret; -- 2.27.0

3 2

[PATCH v2] ASoC: intel: sof_rt5682: Add support for tgl_rt1011_rt5682
by Brent Lu 05 Dec '20

05 Dec '20

This patch adds the driver data for two rt1011 speaker amplifiers on SSP1 and rt5682 on SSP0 for TGL platform. DAI format for rt1011 is leveraged from cml_rt1011_rt5682 which is 4-slot tdm with 100fs bclk. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 138 ++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 24 +++ sound/soc/intel/boards/sof_rt5682.c | 26 +++- .../intel/common/soc-acpi-intel-tgl-match.c | 17 ++- 6 files changed, 201 insertions(+), 7 deletions(-) create mode 100644 sound/soc/intel/boards/sof_realtek_common.c create mode 100644 sound/soc/intel/boards/sof_realtek_common.h diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index dddb672a6d55..b58b9b60d37e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -443,6 +443,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT5682_I2C select SND_SOC_DMIC diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a58e4d22e9c8..5f03e484b215 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -18,7 +18,7 @@ snd-soc-sst-byt-cht-cx2072x-objs := bytcht_cx2072x.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o -snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o +snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o sof_realtek_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o hda_dsp_common.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c new file mode 100644 index 000000000000..f3cf73c620ba --- /dev/null +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -0,0 +1,138 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2020 Intel Corporation. All rights reserved. + +#include <linux/device.h> +#include <linux/kernel.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <uapi/sound/asound.h> +#include "../../codecs/rt1011.h" +#include "sof_realtek_common.h" + +/* + * Current only 2-amp configuration is supported for rt1011 + */ +static const struct snd_soc_dapm_route rt1011_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left SPO" }, + { "Right Spk", NULL, "Right SPO" }, +}; + +/* + * Make sure device's Unique ID follows this configuration: + * + * Two speakers: + * 0: left, 1: right + * Four speakers: + * 0: Woofer left, 1: Woofer right + * 2: Tweeter left, 3: Tweeter right + */ +static struct snd_soc_codec_conf rt1011_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1011_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1011_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1011_dai_link_components[] = { + { + .name = RT1011_DEV0_NAME, + .dai_name = RT1011_CODEC_DAI, + }, + { + .name = RT1011_DEV1_NAME, + .dai_name = RT1011_CODEC_DAI, + }, +}; + +static const struct { + unsigned int tx; + unsigned int rx; +} rt1011_tdm_mask[] = { + {.tx = 0x4, .rx = 0x1}, + {.tx = 0x8, .rx = 0x2}, +}; + +static int rt1011_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int srate, i, ret = 0; + + srate = params_rate(params); + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + /* 100 Fs to drive 24 bit data */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, + 100 * srate, 256 * srate); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set pll, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1011_FS_SYS_PRE_S_PLL1, + 256 * srate, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set sysclk, ret %d\n", + ret); + return ret; + } + + if (i >= ARRAY_SIZE(rt1011_tdm_mask)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", + i); + return -ENODEV; + } + + ret = snd_soc_dai_set_tdm_slot(codec_dai, rt1011_tdm_mask[i].tx, + rt1011_tdm_mask[i].rx, 4, + params_width(params)); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set tdm slot, ret %d\n", + ret); + return ret; + } + } + + return 0; +} + +static const struct snd_soc_ops rt1011_ops = { + .hw_params = rt1011_hw_params, +}; + +static int rt1011_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1011_dapm_routes, + ARRAY_SIZE(rt1011_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_rt1011_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = rt1011_dai_link_components; + link->num_codecs = ARRAY_SIZE(rt1011_dai_link_components); + link->init = rt1011_init; + link->ops = &rt1011_ops; +} + +void sof_rt1011_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = rt1011_codec_confs; + card->num_configs = ARRAY_SIZE(rt1011_codec_confs); +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h new file mode 100644 index 000000000000..87cb3812b926 --- /dev/null +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright(c) 2020 Intel Corporation. + */ + +/* + * This file defines data structures used in Machine Driver for Intel + * platforms with Realtek Codecs. + */ +#ifndef __SOF_REALTEK_COMMON_H +#define __SOF_REALTEK_COMMON_H + +#include <sound/soc.h> + +#define RT1011_CODEC_DAI "rt1011-aif" +#define RT1011_DEV0_NAME "i2c-10EC1011:00" +#define RT1011_DEV1_NAME "i2c-10EC1011:01" +#define RT1011_DEV2_NAME "i2c-10EC1011:02" +#define RT1011_DEV3_NAME "i2c-10EC1011:03" + +void sof_rt1011_dai_link(struct snd_soc_dai_link *link); +void sof_rt1011_codec_conf(struct snd_soc_card *card); + +#endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 891f908659f5..8b1ca2da9bb9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -24,6 +24,7 @@ #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" #include "sof_maxim_common.h" +#include "sof_realtek_common.h" #define NAME_SIZE 32 @@ -41,10 +42,11 @@ #define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) #define SOF_RT5682_NUM_HDMIDEV(quirk) \ ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) -#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(13) -#define SOF_RT1015_SPEAKER_AMP_100FS BIT(14) -#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_RT1011_SPEAKER_AMP_PRESENT BIT(13) +#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(14) +#define SOF_RT1015_SPEAKER_AMP_100FS BIT(15) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(17) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -741,6 +743,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].codecs = max98360a_component; links[id].num_codecs = ARRAY_SIZE(max98360a_component); links[id].init = speaker_codec_init; + } else if (sof_rt5682_quirk & + SOF_RT1011_SPEAKER_AMP_PRESENT) { + sof_rt1011_dai_link(&links[id]); } else { links[id].codecs = max98357a_component; links[id].num_codecs = ARRAY_SIZE(max98357a_component); @@ -851,6 +856,8 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) sof_max98373_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) + sof_rt1011_codec_conf(&sof_audio_card_rt5682); dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); @@ -931,6 +938,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT1015_SPEAKER_AMP_100FS | SOF_RT5682_SSP_AMP(1)), }, + { + .name = "tgl_rt1011_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1011_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; @@ -948,6 +964,7 @@ module_platform_driver(sof_audio) MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); +MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); MODULE_ALIAS("platform:tgl_max98357a_rt5682"); @@ -955,3 +972,4 @@ MODULE_ALIAS("platform:jsl_rt5682_rt1015"); MODULE_ALIAS("platform:tgl_max98373_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_max98360a"); MODULE_ALIAS("platform:cml_rt1015_rt5682"); +MODULE_ALIAS("platform:tgl_rt1011_rt5682"); diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 9f243e60b95c..98196e9fad62 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -9,7 +9,7 @@ #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> -static struct snd_soc_acpi_codecs tgl_codecs = { +static const struct snd_soc_acpi_codecs tgl_codecs = { .num_codecs = 1, .codecs = {"MX98357A"} }; @@ -305,11 +305,16 @@ static const struct snd_soc_acpi_link_adr tgl_3_in_1_sdca[] = { {} }; -static struct snd_soc_acpi_codecs tgl_max98373_amp = { +static const struct snd_soc_acpi_codecs tgl_max98373_amp = { .num_codecs = 1, .codecs = {"MX98373"} }; +static const struct snd_soc_acpi_codecs tgl_rt1011_amp = { + .num_codecs = 1, + .codecs = {"10EC1011"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC1308", @@ -335,6 +340,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { .sof_fw_filename = "sof-tgl.ri", .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "tgl_rt1011_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &tgl_rt1011_amp, + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt1011-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines); -- 2.17.1

3 2

[PATCH] ASoC: jz4740-i2s: add missed checks for clk_get()
by Chuhong Yuan 05 Dec '20

05 Dec '20

jz4740_i2s_set_sysclk() does not check the return values of clk_get(), while the file dereferences the pointers in clk_put(). Add the missed checks to fix it. Fixes: 11bd3dd1b7c2 ("ASoC: Add JZ4740 ASoC support") Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/jz4740/jz4740-i2s.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index c7bd20104b20..0793e284d0e7 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -312,10 +312,14 @@ static int jz4740_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, switch (clk_id) { case JZ4740_I2S_CLKSRC_EXT: parent = clk_get(NULL, "ext"); + if (IS_ERR(parent)) + return PTR_ERR(parent); clk_set_parent(i2s->clk_i2s, parent); break; case JZ4740_I2S_CLKSRC_PLL: parent = clk_get(NULL, "pll half"); + if (IS_ERR(parent)) + return PTR_ERR(parent); clk_set_parent(i2s->clk_i2s, parent); ret = clk_set_rate(i2s->clk_i2s, freq); break; -- 2.26.2

3 2

Re: [PATCH 1/4] ASoC/SoundWire: rt1316: Add RT1316 SDCA vendor-specific driver
by Mark Brown 04 Dec '20

04 Dec '20

On Fri, Dec 04, 2020 at 02:42:54AM +0000, Shuming [范書銘] wrote: > > > +/* RT1316 SDCA function topology */ > > > +#define FUN_SMART_AMP 0x04 > > Full marks for picking this constant! > Sorry, I don't get that point. Could you give me a hint? The name is funny - "fun smart amp".

2 1

[PATCH] [RFC] alsaloop: add feedback frequency control support for UAC2 gadgets
by Ruslan Bilovol 04 Dec '20

04 Dec '20

Add support of special "PCM Feedback Frequency Hz" UAC2 Gadget mixer control that is designed to notify host about real sampling frequency of the gadget so it can adjust number of samples that hosts sends to the gadget. This is useful if both host and gadget has its own internal freerunning clock, so host can adjust number of samples sent, preventing overrun/underrun conditions. This patch reuses logic of the "PCM Rate Shift 100000" control used in case of in-kernel ALSA loopback driver. The only difference is alsaloop reports not rate shift but frequency in Hz Signed-off-by: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> --- alsaloop/alsaloop.h | 1 + alsaloop/pcmjob.c | 35 +++++++++++++++++++++++++---------- 2 files changed, 26 insertions(+), 10 deletions(-) diff --git a/alsaloop/alsaloop.h b/alsaloop/alsaloop.h index c4aa618..9a50a42 100644 --- a/alsaloop/alsaloop.h +++ b/alsaloop/alsaloop.h @@ -122,6 +122,7 @@ struct loopback_handle { unsigned int ctl_pollfd_count; snd_ctl_elem_value_t *ctl_notify; snd_ctl_elem_value_t *ctl_rate_shift; + snd_ctl_elem_value_t *ctl_fback_freq; snd_ctl_elem_value_t *ctl_active; snd_ctl_elem_value_t *ctl_format; snd_ctl_elem_value_t *ctl_rate; diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 6a9aff4..b3802a8 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1058,15 +1058,22 @@ static int set_notify(struct loopback_handle *lhandle, int enable) static int set_rate_shift(struct loopback_handle *lhandle, double pitch) { - int err; + int err = 0; - if (lhandle->ctl_rate_shift == NULL) - return 0; - snd_ctl_elem_value_set_integer(lhandle->ctl_rate_shift, 0, pitch * 100000); - err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_rate_shift); - if (err < 0) { - logit(LOG_CRIT, "Cannot set PCM Rate Shift element for %s: %s\n", lhandle->id, snd_strerror(err)); - return err; + if (lhandle->ctl_rate_shift) { + snd_ctl_elem_value_set_integer(lhandle->ctl_rate_shift, 0, pitch * 100000); + err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_rate_shift); + if (err < 0) { + logit(LOG_CRIT, "Cannot set PCM Rate Shift element for %s: %s\n", lhandle->id, snd_strerror(err)); + return err; + } + } else if (lhandle->ctl_fback_freq) { + snd_ctl_elem_value_set_integer(lhandle->ctl_fback_freq, 0, lhandle->rate * (2.0 - pitch)); + err = snd_ctl_elem_write(lhandle->ctl, lhandle->ctl_fback_freq); + if (err < 0) { + logit(LOG_CRIT, "Cannot set PCM Feedback Frequency element for %s: %s\n", lhandle->id, snd_strerror(err)); + return err; + } } return 0; } @@ -1195,6 +1202,7 @@ static int openctl(struct loopback_handle *lhandle, int device, int subdevice) int err; lhandle->ctl_rate_shift = NULL; + lhandle->ctl_fback_freq = NULL; if (lhandle->loopback->play == lhandle) { if (lhandle->loopback->controls) goto __events; @@ -1204,6 +1212,8 @@ static int openctl(struct loopback_handle *lhandle, int device, int subdevice) &lhandle->ctl_notify); openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", &lhandle->ctl_rate_shift); + openctl_elem(lhandle, device, subdevice, "PCM Feedback Frequency Hz", + &lhandle->ctl_fback_freq); set_rate_shift(lhandle, 1); openctl_elem(lhandle, device, subdevice, "PCM Slave Active", &lhandle->ctl_active); @@ -1289,6 +1299,9 @@ static int closeit(struct loopback_handle *lhandle) if (lhandle->ctl_rate_shift) snd_ctl_elem_value_free(lhandle->ctl_rate_shift); lhandle->ctl_rate_shift = NULL; + if (lhandle->ctl_fback_freq) + snd_ctl_elem_value_free(lhandle->ctl_fback_freq); + lhandle->ctl_fback_freq = NULL; if (lhandle->ctl) err = snd_ctl_close(lhandle->ctl); lhandle->ctl = NULL; @@ -1334,9 +1347,11 @@ int pcmjob_init(struct loopback *loop) snprintf(id, sizeof(id), "%s/%s", loop->play->id, loop->capt->id); id[sizeof(id)-1] = '\0'; loop->id = strdup(id); - if (loop->sync == SYNC_TYPE_AUTO && loop->capt->ctl_rate_shift) + if (loop->sync == SYNC_TYPE_AUTO && (loop->capt->ctl_rate_shift || + loop->capt->ctl_fback_freq)) loop->sync = SYNC_TYPE_CAPTRATESHIFT; - if (loop->sync == SYNC_TYPE_AUTO && loop->play->ctl_rate_shift) + if (loop->sync == SYNC_TYPE_AUTO && (loop->play->ctl_rate_shift || + loop->play->ctl_fback_freq)) loop->sync = SYNC_TYPE_PLAYRATESHIFT; #ifdef USE_SAMPLERATE if (loop->sync == SYNC_TYPE_AUTO && loop->src_enable) -- 1.9.1

2 2

How to set channel balance in amixer?
by GitHub issues - opened 04 Dec '20

04 Dec '20

alsa-project/alsa-utils issue #73 was opened from quantum-booty: When I change volumn using keyboard media keys, the left and right channel becomes imbalanced. ![image](https://user-images.githubusercontent.com/28958399/101188833-23befc00-364e-11eb-8816-efd94b9514d4.png) I currently have this mapping in i3, I would like to add command to balance the channels after I change the volumn. bindsym XF86AudioRaiseVolume exec --no-startup-id "amixer set Master '5%+,5%+'" Issue URL : https://github.com/alsa-project/alsa-utils/issues/73 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] [v2] ASoC: cros_ec_codec: fix uninitialized memory read
by Arnd Bergmann 04 Dec '20

04 Dec '20

From: Arnd Bergmann <arnd(a)arndb.de> gcc points out a memory area that is copied to a device but not initialized: sound/soc/codecs/cros_ec_codec.c: In function 'i2s_rx_event': arch/x86/include/asm/string_32.h:83:20: error: '*((void *)&p+4)' may be used uninitialized in this function [-Werror=maybe-uninitialized] 83 | *((int *)to + 1) = *((int *)from + 1); Change the length of the command to only pass down the part of the structure that has been initialized, as Tzung-Bi Shih explains that only that member is meant to be used. Cc: Tzung-Bi Shih <tzungbi(a)google.com> Fixes: 727f1c71c780 ("ASoC: cros_ec_codec: refactor I2S RX") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cros_ec_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 58894bf47514..6ec673573c70 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -348,7 +348,7 @@ static int i2s_rx_event(struct snd_soc_dapm_widget *w, } return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, - (uint8_t *)&p, sizeof(p), NULL, 0); + &p.cmd, sizeof(p.cmd), NULL, 0); } static struct snd_soc_dapm_widget i2s_rx_dapm_widgets[] = { -- 2.27.0

2 2

Behringer WING usb audio - cyclic xruns dependent on periods/buffers
by Ben Bell 03 Dec '20

03 Dec '20

I've got a Behringer WING digital mixer which is equipped with a USB interface supporting 48 in, 48 out at 44.1 or 48kHz. It's plugged into a USB3 interface and for the most part it seems to work well as a class-compliant audio interface, but I'm struggling to eliminate xruns and it's starting to feel either driver or hardware quirk related issue. I've done all the usual things -- I'm running a PREEMPT_RT kernel, set up the realtime priorities of Jack and the relevant USB IRQ to no avail. I'm doing most of the debugging remotely over ssh with X shutdown and most other processes on the system stopped. There's nothing plugged into the computer beyond network, keyboard, mouse and the audio interface. I'm currently testing on kernel 5.10.0-rc5 with preempt-rt patches, but I've previously seen the same things on 5.4-rt (Debian backports) and 5.8 (self-compiled). The xruns are characterised by bursts of "retire_capture_urb" warnings in the kernel logs (and in particular the frame status for everything in those URBs is EXDEV, meaning I think that the frames haven't been consumed by the time it's retired). I've patched sound/driver/pcm.c to provide me with a bit more debugging information and it looks like usually it's almost always between 7-10 consecutive calls to retire_capture_urb that are affected. The bursts of retire_capture_urb warnings seem roughly cyclic, with a cycle time that is dependent on the combination of the frames/period setting and the number of periods per buffer, though it doesn't appear to be a strictly linear thing: 512/2 (~170s cycle); 256/2 (~94s); 128/3 (~65s); 64/13 (~270s). I can't immediately see any definitive smoking gun in the urb or interrupt counts, but there is some grouping in the timings between the xruns. Taking 64 frames, 5 periods, it's usually 50s, but sometimes 60s: 21:47:57 50s since xrun 4 21:48:47 50s since xrun 5 21:49:46 59s since xrun 6 21:50:46 60s since xrun 7 21:51:36 50s since xrun 8 21:52:36 60s since xrun 9 21:53:36 60s since xrun 10 21:54:37 61s since xrun 11 21:55:27 50s since xrun 12 21:56:27 60s since xrun 13 21:57:27 60s since xrun 14 21:58:17 50s since xrun 15 21:59:16 59s since xrun 16 22:00:06 50s since xrun 17 22:00:56 50s since xrun 18 ... If I count the URB numbers (sorry, I'm shaky on the terminology, but I'm counting each URB which is retired) I often (but not always) see the same number of URBs passing between xrun bursts. Currently at 64 frames/5 periods, it's often exactly 50042 or 49923 urbs between bursts. The delay between starting Jack and encountering the first burst is not predictable -- it's not just a whole cycle -- but stopping jack for e.g 30s then starting again delays the next burst by the same amount. So it does seem related to something in the audio streaming rather than anything else going on on the system interfering. It's as if something is slowly slipping out of sync between the Wing and the kernel until they need to resync, but I don't really know enough about USB to have any deeper insight. All the tests above happen to be at 44.1kHz with the Wing set to 48 in, 48 out, but changing to 48kHz or 2/2 IO doesn't seem to cure anything. One more data point which may not be relevant: if I switch off or unplug the Wing while Jack is running, the system gradually locks up over a period of a few seconds. I can get something out of the console briefly afterwards but within about ten seconds it's completely unresponsive. I imagine that pulling out a USB device from a realtime thread isn't a kind thing to do, but I don't recall this happening with other interfaces. Any help or insights into where I should be looking (I'm a newbie in kernel space) appreciated. If someone wants remote access to the box to investigate in realtime we could probably figure something out. bjb

2 3

[bug report] ASoC: mchp-spdifrx: add driver for SPDIF RX
by Dan Carpenter 03 Dec '20

03 Dec '20

Hello Codrin Ciubotariu, The patch ef265c55c1ac: "ASoC: mchp-spdifrx: add driver for SPDIF RX" from Oct 2, 2020, leads to the following static checker warning: sound/soc/atmel/mchp-spdifrx.c:468 mchp_spdifrx_hw_params() warn: 'dev->gclk' not released on lines: 468. sound/soc/atmel/mchp-spdifrx.c 442 params_format(params)); 443 return -EINVAL; 444 } 445 446 if (dev->gclk_enabled) { 447 clk_disable_unprepare(dev->gclk); 448 dev->gclk_enabled = 0; 449 } 450 ret = clk_set_min_rate(dev->gclk, params_rate(params) * 451 SPDIFRX_GCLK_RATIO_MIN + 1); 452 if (ret) { 453 dev_err(dev->dev, 454 "unable to set gclk min rate: rate %u * ratio %u + 1\n", 455 params_rate(params), SPDIFRX_GCLK_RATIO_MIN); 456 return ret; 457 } 458 ret = clk_prepare_enable(dev->gclk); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 459 if (ret) { 460 dev_err(dev->dev, "unable to enable gclk: %d\n", ret); 461 return ret; 462 } 463 dev->gclk_enabled = 1; 464 465 dev_dbg(dev->dev, "GCLK range min set to %d\n", 466 params_rate(params) * SPDIFRX_GCLK_RATIO_MIN + 1); 467 468 return regmap_write(dev->regmap, SPDIFRX_MR, mr); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Smatch is complaining that if the regmap_write() fails then we should disable and unprepare the "dev->gclk". 469 } regards, dan carpenter

2 2

[PATCH] ASoC: intel: sof_rt5682: Add support for tgl_rt1011_rt5682
by Brent Lu 03 Dec '20

03 Dec '20

This patch adds the driver data for two rt1011 speaker amplifiers on SSP1 and rt5682 on SSP0 for TGL platform. DAI format for rt1011 is leveraged from cml_rt1011_rt5682 which is 4-slot tdm with 100fs bclk. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 137 ++++++++++++++++++ sound/soc/intel/boards/sof_realtek_common.h | 24 +++ sound/soc/intel/boards/sof_rt5682.c | 26 +++- .../intel/common/soc-acpi-intel-tgl-match.c | 13 ++ 6 files changed, 198 insertions(+), 5 deletions(-) create mode 100644 sound/soc/intel/boards/sof_realtek_common.c create mode 100644 sound/soc/intel/boards/sof_realtek_common.h diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index dddb672a6d55..b58b9b60d37e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -443,6 +443,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT5682_I2C select SND_SOC_DMIC diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a58e4d22e9c8..5f03e484b215 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -18,7 +18,7 @@ snd-soc-sst-byt-cht-cx2072x-objs := bytcht_cx2072x.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o -snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o +snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o sof_realtek_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o hda_dsp_common.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c new file mode 100644 index 000000000000..b3bc41385436 --- /dev/null +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -0,0 +1,137 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2020 Intel Corporation. All rights reserved. + +#include <linux/kernel.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <uapi/sound/asound.h> +#include "../../codecs/rt1011.h" +#include "sof_realtek_common.h" + +/* + * Current only 2-amp configuration is supported for rt1011 + */ +static const struct snd_soc_dapm_route rt1011_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left SPO" }, + { "Right Spk", NULL, "Right SPO" }, +}; + +/* + * Make sure device's Unique ID follows this configuration: + * + * Two speakers: + * 0: left, 1: right + * Four speakers: + * 0: Woofer left, 1: Woofer right + * 2: Tweeter left, 3: Tweeter right + */ +static struct snd_soc_codec_conf rt1011_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1011_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(RT1011_DEV1_NAME), + .name_prefix = "Right", + }, +}; + +static struct snd_soc_dai_link_component rt1011_dai_link_components[] = { + { + .name = RT1011_DEV0_NAME, + .dai_name = RT1011_CODEC_DAI, + }, + { + .name = RT1011_DEV1_NAME, + .dai_name = RT1011_CODEC_DAI, + }, +}; + +struct { + unsigned int tx; + unsigned int rx; +} rt1011_tdm_mask[] = { + {.tx = 0x4, .rx = 0x1}, + {.tx = 0x8, .rx = 0x2}, +}; + +static int rt1011_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int srate, i, ret = 0; + + srate = params_rate(params); + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + /* 100 Fs to drive 24 bit data */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, + 100 * srate, 256 * srate); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set pll, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT1011_FS_SYS_PRE_S_PLL1, + 256 * srate, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set sysclk, ret %d\n", + ret); + return ret; + } + + if (i >= ARRAY_SIZE(rt1011_tdm_mask)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", + i); + return -ENODEV; + } + + ret = snd_soc_dai_set_tdm_slot(codec_dai, rt1011_tdm_mask[i].tx, + rt1011_tdm_mask[i].rx, 4, + params_width(params)); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set tdm slot, ret %d\n", + ret); + return ret; + } + } + + return 0; +} + +static const struct snd_soc_ops rt1011_ops = { + .hw_params = rt1011_hw_params, +}; + +static int rt1011_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1011_dapm_routes, + ARRAY_SIZE(rt1011_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_rt1011_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = rt1011_dai_link_components; + link->num_codecs = ARRAY_SIZE(rt1011_dai_link_components); + link->init = rt1011_init; + link->ops = &rt1011_ops; +} + +void sof_rt1011_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = rt1011_codec_confs; + card->num_configs = ARRAY_SIZE(rt1011_codec_confs); +} diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h new file mode 100644 index 000000000000..87cb3812b926 --- /dev/null +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright(c) 2020 Intel Corporation. + */ + +/* + * This file defines data structures used in Machine Driver for Intel + * platforms with Realtek Codecs. + */ +#ifndef __SOF_REALTEK_COMMON_H +#define __SOF_REALTEK_COMMON_H + +#include <sound/soc.h> + +#define RT1011_CODEC_DAI "rt1011-aif" +#define RT1011_DEV0_NAME "i2c-10EC1011:00" +#define RT1011_DEV1_NAME "i2c-10EC1011:01" +#define RT1011_DEV2_NAME "i2c-10EC1011:02" +#define RT1011_DEV3_NAME "i2c-10EC1011:03" + +void sof_rt1011_dai_link(struct snd_soc_dai_link *link); +void sof_rt1011_codec_conf(struct snd_soc_card *card); + +#endif /* __SOF_REALTEK_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 891f908659f5..8b1ca2da9bb9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -24,6 +24,7 @@ #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" #include "sof_maxim_common.h" +#include "sof_realtek_common.h" #define NAME_SIZE 32 @@ -41,10 +42,11 @@ #define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) #define SOF_RT5682_NUM_HDMIDEV(quirk) \ ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) -#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(13) -#define SOF_RT1015_SPEAKER_AMP_100FS BIT(14) -#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_RT1011_SPEAKER_AMP_PRESENT BIT(13) +#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(14) +#define SOF_RT1015_SPEAKER_AMP_100FS BIT(15) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(17) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -741,6 +743,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].codecs = max98360a_component; links[id].num_codecs = ARRAY_SIZE(max98360a_component); links[id].init = speaker_codec_init; + } else if (sof_rt5682_quirk & + SOF_RT1011_SPEAKER_AMP_PRESENT) { + sof_rt1011_dai_link(&links[id]); } else { links[id].codecs = max98357a_component; links[id].num_codecs = ARRAY_SIZE(max98357a_component); @@ -851,6 +856,8 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) sof_max98373_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) + sof_rt1011_codec_conf(&sof_audio_card_rt5682); dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); @@ -931,6 +938,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT1015_SPEAKER_AMP_100FS | SOF_RT5682_SSP_AMP(1)), }, + { + .name = "tgl_rt1011_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1011_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; @@ -948,6 +964,7 @@ module_platform_driver(sof_audio) MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); +MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); MODULE_ALIAS("platform:tgl_max98357a_rt5682"); @@ -955,3 +972,4 @@ MODULE_ALIAS("platform:jsl_rt5682_rt1015"); MODULE_ALIAS("platform:tgl_max98373_rt5682"); MODULE_ALIAS("platform:jsl_rt5682_max98360a"); MODULE_ALIAS("platform:cml_rt1015_rt5682"); +MODULE_ALIAS("platform:tgl_rt1011_rt5682"); diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 9f243e60b95c..84cc6326979a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -310,6 +310,11 @@ static struct snd_soc_acpi_codecs tgl_max98373_amp = { .codecs = {"MX98373"} }; +static struct snd_soc_acpi_codecs tgl_rt1011_amp = { + .num_codecs = 1, + .codecs = {"10EC1011"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC1308", @@ -335,6 +340,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { .sof_fw_filename = "sof-tgl.ri", .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "tgl_rt1011_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &tgl_rt1011_amp, + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt1011-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines); -- 2.17.1

3 2

[PATCH] ASoC: amd: add missed checks for clk_get()
by Chuhong Yuan 03 Dec '20

03 Dec '20

cz_da7219_init() does not check the return value of clk_get(), while da7219_clk_enable() calls clk_set_rate() to dereference the pointers. Add checks to fix the problems. Fixes: bb24a31ed584 ("ASoC: AMD: Configure wclk and bclk of master codec") Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/amd/acp-da7219-max98357a.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index a7702e64ec51..d5185cd3106b 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -74,7 +74,12 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) } da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); + if (IS_ERR(da7219_dai_wclk)) + return PTR_ERR(da7219_dai_wclk); + da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); + if (IS_ERR(da7219_dai_bclk)) + return PTR_ERR(da7219_dai_bclk); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | -- 2.26.2

2 1

[PATCH] ASoC: fsl_audmix: Remove unneeded data field
by Fabio Estevam 03 Dec '20

03 Dec '20

The .data field is only used to pass the string name to platform_device_register_data(). Pass the string name directly to make the code simpler. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/fsl_audmix.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 7ad5925772e8..8dc44dec7956 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -455,7 +455,6 @@ static const struct regmap_config fsl_audmix_regmap_config = { static const struct of_device_id fsl_audmix_ids[] = { { .compatible = "fsl,imx8qm-audmix", - .data = "imx-audmix", }, { /* sentinel */ } }; @@ -465,17 +464,9 @@ static int fsl_audmix_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct fsl_audmix *priv; - const char *mdrv; - const struct of_device_id *of_id; void __iomem *regs; int ret; - of_id = of_match_device(fsl_audmix_ids, dev); - if (!of_id || !of_id->data) - return -EINVAL; - - mdrv = of_id->data; - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; @@ -510,10 +501,10 @@ static int fsl_audmix_probe(struct platform_device *pdev) goto err_disable_pm; } - priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0); + priv->pdev = platform_device_register_data(dev, "imx-audmix", 0, NULL, 0); if (IS_ERR(priv->pdev)) { ret = PTR_ERR(priv->pdev); - dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret); + dev_err(dev, "failed to register platform: %d\n", ret); goto err_disable_pm; } -- 2.17.1

2 1

[PATCH 1/2] ASoC: fsl-asoc-card: Add support for si476x codec
by Shengjiu Wang 03 Dec '20

03 Dec '20

The si476x codec is used for FM radio function on i.MX6 auto board, it only supports recording function. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index a2dd3b6b7fec..f62f81ceab0d 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -131,6 +131,13 @@ static const struct snd_soc_dapm_route audio_map_tx[] = { {"CPU-Playback", NULL, "ASRC-Playback"}, }; +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* 1st half -- Normal DAPM routes */ + {"CPU-Capture", NULL, "Capture"}, + /* 2nd half -- ASRC DAPM routes */ + {"ASRC-Capture", NULL, "CPU-Capture"}, +}; + /* Add all possible widgets into here without being redundant */ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_LINE("Line Out Jack", NULL), @@ -653,6 +660,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.slot_width = 32; priv->card.dapm_routes = audio_map_tx; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); + } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) { + codec_dai_name = "si476x-codec"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + priv->card.dapm_routes = audio_map_rx; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx); } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -869,6 +881,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, { .compatible = "fsl,imx-audio-wm8524", }, + { .compatible = "fsl,imx-audio-si476x", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); -- 2.27.0

3 3

[PATCH 0/4] ASoC/SoundWire: add Realtek SoundWire SDCA codec
by Bard Liao 03 Dec '20

03 Dec '20

Add Realtek rt1316 and rt711-sdca vender-specific drivers plus some fixup for rt715-sdca driver. Jack Yu (2): ASoC/SoundWire: rt715-sdca: use rt715_sdca prefix for symbols ASoC/SoundWire: rt715-sdca: merge mute/volume operations Shuming Fan (2): ASoC/SoundWire: rt1316: Add RT1316 SDCA vendor-specific driver ASoC/SoundWire: rt711-sdca: Add RT711 SDCA vendor-specific driver sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt1316-sdw.c | 756 +++++++++++++++ sound/soc/codecs/rt1316-sdw.h | 115 +++ sound/soc/codecs/rt711-sdca-sdw.c | 424 +++++++++ sound/soc/codecs/rt711-sdca-sdw.h | 101 ++ sound/soc/codecs/rt711-sdca.c | 1482 +++++++++++++++++++++++++++++ sound/soc/codecs/rt711-sdca.h | 246 +++++ sound/soc/codecs/rt715-sdca-sdw.c | 12 +- sound/soc/codecs/rt715-sdca.c | 323 ++++--- sound/soc/codecs/rt715-sdca.h | 4 +- 11 files changed, 3316 insertions(+), 164 deletions(-) create mode 100644 sound/soc/codecs/rt1316-sdw.c create mode 100644 sound/soc/codecs/rt1316-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca-sdw.c create mode 100644 sound/soc/codecs/rt711-sdca-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca.c create mode 100644 sound/soc/codecs/rt711-sdca.h -- 2.17.1

2 5

[PATCH v3 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. CPU0 CPU1 dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 2.21.0

1 0

[PATCH v3 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

>Hi Gyeongtaek > >Thank you for your help > >> I tried to find out some change from the history. >> However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. >> So, I just fixed some miss from the commit comment and resend. >> If you have another suggestion or request, just let me know. > >OK, thanks. >I just thought it was related to recent (especially mine :) patch. > >BYW, your posted v2 still doesn't have alignment on log. >Is it mailer issue ? Sorry. I didn't fully understand what alignment means. I fixed the log by looking at some previous commits. Thank you so much. Gyeongtaek Lee > >> And I have a question. >> This patch can?t be applied to stable branches directly by conflict. >> I found out following commits are needed to apply this patch into the 5.4 stable. >> ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() >> ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code >> In this case, what should I do when stable maintainer requests resolving conflict to me? >> Should I send those patches to the maintainer? >> Or rewrite the patch on the latest of the each stable? > >Mark, please help him > >Thank you for your help !! > >Best regards >--- >Kuninori Morimoto > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

1 0

[PATCH v2 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

On 03 Dec 2020 07:33:03 +0900, Kuninori Morimoto wrote: >Hi Gyeongtaek > >Thank you for your patch > >> If stop by underrun and DPCM BE disconnection is run simultaneously, >> data abort can be occurred by the sequence below. >> >> /* In core X, running dpcm_be_dai_trigger() */ >> for_each_dpcm_be(fe, stream, dpcm) { >> /* In core Y, running dpcm_be_disconnect() */ >> spin_lock_irqsave(&fe->card->dpcm_lock, flags); >> list_del(&dpcm->list_be); >> list_del(&dpcm->list_fe); >> spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); >> kfree(dpcm); >> /* In core X, running dpcm_be_dai_trigger() */ >> struct snd_soc_pcm_runtime *be = dpcm->be; <== Accessing freed memory > >It is easy to read/understand if this code has alignment. > >> To prevent this situation, dpcm_lock should be acquired during >> iteration of dpcm list in dpcm_do_trigger(). >> >> Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> >> Cc: stable(a)vger.kernel.org >> --- > >Is this bugfix patch for some recent commit ? >If so, having Fixes: tag is helpful. > >Thank you for your help !! I tried to find out some change from the history. However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. So, I just fixed some miss from the commit comment and resend. If you have another suggestion or request, just let me know. And I have a question. This patch can’t be applied to stable branches directly by conflict. I found out following commits are needed to apply this patch into the 5.4 stable. ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code In this case, what should I do when stable maintainer requests resolving conflict to me? Should I send those patches to the maintainer? Or rewrite the patch on the latest of the each stable? Thank you! Gyeongtaek Lee > >Best regards >--- >Kuninori Morimoto > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

2 1

[PATCH v0] ASoC: rsnd: core: Check convert rate in rsnd_hw_params
by mdurnev＠gmail.com 03 Dec '20

03 Dec '20

From: Mikhail Durnev <mikhail_durnev(a)mentor.com> snd_pcm_hw_params_set_rate_near can return incorrect sample rate in some cases, e.g. when the backend output rate is set to some value higher than 48000 Hz and the input rate is 8000 Hz. So passing the value returned by snd_pcm_hw_params_set_rate_near to snd_pcm_hw_params will result in "FSO/FSI ratio error" and playing no audio at all while the userland is not properly notified about the issue. If SRC is unable to convert the requested sample rate to the sample rate the backend is using, then the requested sample rate should be adjusted in rsnd_hw_params. The userland will be notified about that change in the returned hw_params structure. Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> --- sound/soc/sh/rcar/core.c | 67 +++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 66 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6e670b3..8ca3fb7 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1428,8 +1428,73 @@ static int rsnd_hw_params(struct snd_soc_component *component, } if (io->converted_chan) dev_dbg(dev, "convert channels = %d\n", io->converted_chan); - if (io->converted_rate) + if (io->converted_rate) { dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + + /* + * SRC supports convert rates from params_rate(hw_params)/k_down + * to params_rate(hw_params)*k_up, where k_up is always 6, and + * k_down depends on number of channels and SRC unit. + * So all SRC units can upsample audio up to 6 times regardless + * its number of channels. And all SRC units can downsample + * 2 channel audio up to 6 times too. + */ + int k_up = 6; + int k_down = 6; + int channel; + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + + channel = io->converted_chan ? io->converted_chan : fe_channel; + + switch (rsnd_mod_id(src_mod)) { + /* + * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 4 channel audio + * up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio + * no more than twice. + */ + case 1: + case 3: + case 4: + if (channel > 4) { + k_down = 2; + break; + } + case 0: + if (channel > 2) + k_down = 4; + break; + + /* Other SRC units do not support more than 2 channels */ + default: + if (channel > 2) + return -EINVAL; + } + + if (params_rate(hw_params) > io->converted_rate * k_down) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + io->converted_rate * k_down; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + io->converted_rate * k_down; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } else if (params_rate(hw_params) * k_up < io->converted_rate) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + (io->converted_rate + k_up - 1) / k_up; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + (io->converted_rate + k_up - 1) / k_up; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } + + /* + * TBD: Max SRC input and output rates also depend on number + * of channels and SRC unit: + * SRC1, SRC3 and SRC4 do not support more than 128kHz + * for 6 channel and 96kHz for 8 channel audio. + * Perhaps this function should return EINVAL if the input or + * the output rate exceeds the limitation. + */ + } } return rsnd_dai_call(hw_params, io, substream, hw_params); -- 2.7.4

4 4

[PATCH v2 1/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. /* In core X, running dpcm_be_dai_trigger() */ for_each_dpcm_be(fe, stream, dpcm) { /* In core Y, running dpcm_be_disconnect() */ spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); /* In core X, running dpcm_be_dai_trigger() */ struct snd_soc_pcm_runtime *be = dpcm->be; <== Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_be_dai_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 2.21.0

1 0

[PATCH 1/2] ASoC: dt-bindings: imx-hdmi: Add binding doc for hdmi machine driver
by Shengjiu Wang 03 Dec '20

03 Dec '20

Imx-hdmi is a new added machine driver for supporting hdmi devices on i.MX platforms. There is HDMI IP or external HDMI modules connect with SAI or AUD2HTX interface. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../bindings/sound/imx-audio-hdmi.yaml | 52 +++++++++++++++++++ 1 file changed, 52 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml diff --git a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml new file mode 100644 index 000000000000..d5474f83ac2c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml @@ -0,0 +1,52 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audio-hdmi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP i.MX audio complex with HDMI + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + enum: + - fsl,imx-audio-hdmi + - fsl,imx-audio-sii902x + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + audio-cpu: + description: The phandle of an CPU DAI controller + + hdmi-out: + description: | + This is a boolean property. If present, the transmitting function + of HDMI will be enabled, indicating there's a physical HDMI out + connector or jack on the board or it's connecting to some other IP + block, such as an HDMI encoder or display-controller. + + hdmi-in: + description: | + This is a boolean property. If present, the receiving function of + HDMI will be enabled, indicating there is a physical HDMI in + connector/jack on the board. + +required: + - compatible + - model + - audio-cpu + +additionalProperties: false + +examples: + - | + sound-hdmi { + compatible = "fsl,imx-audio-hdmi"; + model = "audio-hdmi"; + audio-cpu = <&aud2htx>; + hdmi-out; + }; -- 2.27.0

3 3

[PATCH v2 0/5] regmap/SoundWire/ASoC: Add SoundWire SDCA support
by Bard Liao 03 Dec '20

03 Dec '20

The MIPI SoundWire Device Class standard will define audio functionality beyond the scope of the existing SoundWire 1.2 standard, which is limited to the bus and interface. The description is inspired by the USB Audio Class, with "functions", "entities", "control selectors", "audio clusters". The main difference with the USB Audio class is that the devices are typically on a motherboard and descriptors stored in platform firmware instead of being retrieved from the device. The current set of devices managed in this patchset are conformant with the SDCA 0.6 specification and require dedicated drivers since the descriptors and platform firmware specification is not complete at this time. They do however rely on the hierarchical addressing required by the SDCA standard. Future devices conformant with SDCA 1.0 should rely on a class driver. This series adds support for the hierarchical SDCA addressing and extends regmap. It then provides 3 codecs for RT711-sdca headset codec, RT1316 amplifier and RT715-scda microphone codec. Note that the release of this code before the formal adoption of the SDCA 1.0 specification was formally endorsed by the MIPI Board to make sure there is no delay for Linux-based support of this specification. v2: - rt715-sdca: Use rt715_sdca prefix to avoid compiling issue. - rt715-sdca: Merge multiple mute/volume operation into single mute/volume operation - rt711-sdca: Initial ret = 0 as it could be used uninitialized. Jack Yu (1): ASoC/SoundWire: rt715-sdca: First version of rt715 sdw sdca codec driver Pierre-Louis Bossart (2): soundwire: SDCA: add helper macro to access controls regmap/SoundWire: sdw: add support for SoundWire 1.2 MBQ Shuming Fan (2): ASoC/SoundWire: rt1316: Add RT1316 SDCA vendor-specific driver ASoC/SoundWire: rt711-sdca: Add RT711 SDCA vendor-specific driver drivers/base/regmap/Kconfig | 6 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw-mbq.c | 101 ++ include/linux/regmap.h | 35 + include/linux/soundwire/sdw_registers.h | 32 + sound/soc/codecs/Kconfig | 20 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/rt1316-sdw.c | 756 ++++++++++++ sound/soc/codecs/rt1316-sdw.h | 115 ++ sound/soc/codecs/rt711-sdca-sdw.c | 424 +++++++ sound/soc/codecs/rt711-sdca-sdw.h | 101 ++ sound/soc/codecs/rt711-sdca.c | 1482 +++++++++++++++++++++++ sound/soc/codecs/rt711-sdca.h | 246 ++++ sound/soc/codecs/rt715-sdca-sdw.c | 278 +++++ sound/soc/codecs/rt715-sdca-sdw.h | 170 +++ sound/soc/codecs/rt715-sdca.c | 947 +++++++++++++++ sound/soc/codecs/rt715-sdca.h | 124 ++ 17 files changed, 4843 insertions(+), 1 deletion(-) create mode 100644 drivers/base/regmap/regmap-sdw-mbq.c create mode 100644 sound/soc/codecs/rt1316-sdw.c create mode 100644 sound/soc/codecs/rt1316-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca-sdw.c create mode 100644 sound/soc/codecs/rt711-sdca-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca.c create mode 100644 sound/soc/codecs/rt711-sdca.h create mode 100644 sound/soc/codecs/rt715-sdca-sdw.c create mode 100644 sound/soc/codecs/rt715-sdca-sdw.h create mode 100644 sound/soc/codecs/rt715-sdca.c create mode 100644 sound/soc/codecs/rt715-sdca.h -- 2.17.1

5 15

[PATCH] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 03 Dec '20

03 Dec '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. /* In core X, running dpcm_be_dai_trigger() */ for_each_dpcm_be(fe, stream, dpcm) { /* In core Y, running dpcm_be_disconnect() */ spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); /* In core X, running dpcm_be_dai_trigger() */ struct snd_soc_pcm_runtime *be = dpcm->be; <== Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_do_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

2 2

[PATCH] ASoC: codecs: lpass-va-macro: remove some dead code
by Dan Carpenter 02 Dec '20

02 Dec '20

The "decimator" variable is in the 0-7 range and it's unsigned so there is no need to check for negative values. Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/lpass-va-macro.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 3e6bbef26dcb..91e6890d6efc 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -844,17 +844,10 @@ static int va_macro_hw_params(struct snd_pcm_substream *substream, for_each_set_bit(decimator, &va->active_ch_mask[dai->id], VA_MACRO_DEC_MAX) { - if (decimator >= 0) { - tx_fs_reg = CDC_VA_TX0_TX_PATH_CTL + - VA_MACRO_TX_PATH_OFFSET * decimator; - snd_soc_component_update_bits(component, tx_fs_reg, - 0x0F, tx_fs_rate); - } else { - dev_err(va_dev, - "%s: ERROR: Invalid decimator: %d\n", - __func__, decimator); - return -EINVAL; - } + tx_fs_reg = CDC_VA_TX0_TX_PATH_CTL + + VA_MACRO_TX_PATH_OFFSET * decimator; + snd_soc_component_update_bits(component, tx_fs_reg, 0x0F, + tx_fs_rate); } return 0; } -- 2.29.2

3 3

[PATCH] ASoC: rt715: remove unused parameter
by jack.yu＠realtek.com 02 Dec '20

02 Dec '20

From: Jack Yu <jack.yu(a)realtek.com> Remove unused parameter in rt715.h. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index d0d0fd2a6fdb..009a8266f606 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -207,7 +207,6 @@ struct sdw_stream_data { enum { RT715_AIF1, RT715_AIF2, - RT715_AIFS, }; #define RT715_POWER_UP_DELAY_MS 400 -- 2.29.0

2 1

[PATCH] ASoC: Intel: Boards: tgl_max98373: update TDM slot_width
by Ranjani Sridharan 02 Dec '20

02 Dec '20

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Speaker amp's SSP bclk configuration was changed in the topology file to be based on 12.288MHz and dai_ops->hw_params is based on s32le format. But, the TDM slot size remained set to 24 bits. This inconsistency created audible noises and needs to be corrected. This patch updates TDM slot width to 32. Fixes: bc7477fc2ab4 ("ASoC: Intel: Boards: tgl_max98373: Update TDM configuration in hw_params") Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/intel/boards/sof_maxim_common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index b6e63ea13d64..c2a9757181fe 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -49,11 +49,11 @@ static int max98373_hw_params(struct snd_pcm_substream *substream, for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAX_98373_DEV0_NAME)) { /* DEV0 tdm slot configuration */ - snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 8, 24); + snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 8, 32); } if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) { /* DEV1 tdm slot configuration */ - snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 8, 24); + snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 8, 32); } } return 0; -- 2.25.1

2 1

[RESEND PATCH 0/6] ASoC: mediatek: mt8192: support new machine rt1015p_rt5682
by Tzung-Bi Shih 02 Dec '20

02 Dec '20

The series supports new machines with rt1015p and rt5682. Reuses most of the code in mt8192-mt6359-rt1015-rt5682.c. The first 3 patches refactor the existing mt8192-mt6359-rt1015-rt5682.c. For easier to support newly added components later. The 4th patch fixes a typo in DT bindings document. The 5th patch proposes a new compatible string "mt8192_mt6359_rt1015p_rt5682" for machines with rt1015p and rt5682. The 6th patch reuses the existing machine driver to support machines with rt1015p and rt5682. Tzung-Bi Shih (6): ASoC: mediatek: mt8192: rename common symbols ASoC: mediatek: mt8192: extract rt1015_rt5682 specific DAI link ASoC: mediatek: mt8192: move rt1015_rt5682 specific data ASoC: dt-bindings: mt8192-mt6359: fix typo in the example ASoC: dt-bindings: mt8192-mt6359: add new compatible for using rt1015p ASoC: mediatek: mt8192: support rt1015p_rt5682 .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 6 +- sound/soc/mediatek/Kconfig | 1 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 171 +++++++++++++----- 3 files changed, 129 insertions(+), 49 deletions(-) -- 2.29.2.454.gaff20da3a2-goog

2 7

How to interperet snd_pcm_htimestamp available frames?
by Eric Buddington 02 Dec '20

02 Dec '20

I am recording from multiple (cheap USB) mics, and trying to synchronize them. I am assuming that there will be a common minimum latency (reflecting an uncontested USB bus), with occasional spikes up - but not down. In my loop, I am calling snd_pcm_mmap_begin(), snd_pcm_htimestamp(), and snd__mmap_commit() in that order. My question is about the 'avail' value returned from htimestamp. It is documented as "Number of available frames when timestamp was grabbed". Does this mean frames since the last frame consumed with _mmap_commit()? Is there a possibility of there being no timestamp, or one where 'avail' refers to some older state? Thanks for any insight. -Eric

2 1

[PATCH] pcm: set the snd_pcm_ioplug_status() tstamp field
by Jonas Holmberg 02 Dec '20

02 Dec '20

Set the status tstamp field so that it can be accessed with snd_pcm_status_get_htstamp(). Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_ioplug.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index a437ca32..9b1b8ac3 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -115,6 +115,7 @@ static int snd_pcm_ioplug_status(snd_pcm_t *pcm, snd_pcm_status_t * status) snd_pcm_ioplug_hw_ptr_update(pcm); status->state = io->data->state; status->trigger_tstamp = io->trigger_tstamp; + gettimestamp(&status->tstamp, pcm->tstamp_type); status->avail = snd_pcm_mmap_avail(pcm); status->avail_max = io->avail_max; return 0; -- 2.26.2

2 2

[PATCH v1] ASoC: Intel: catpt: Replace open coded variant of resource_intersection()
by Andy Shevchenko 02 Dec '20

02 Dec '20

Since we have resource_intersection() helper, let's utilize it here. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- Promised clean up for recently introduced helper. This has dependency to the patches currently in linux-pm tree. Other than that everything else is already in upstream. Hence, logically it's better to push thru Rafael's tree than wait one more cycle. sound/soc/intel/catpt/core.h | 11 ----------- sound/soc/intel/catpt/loader.c | 2 +- 2 files changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/soc/intel/catpt/core.h b/sound/soc/intel/catpt/core.h index 0f53a0d43254..a64a0a77dcb7 100644 --- a/sound/soc/intel/catpt/core.h +++ b/sound/soc/intel/catpt/core.h @@ -22,17 +22,6 @@ void catpt_sram_free(struct resource *sram); struct resource * catpt_request_region(struct resource *root, resource_size_t size); -static inline bool catpt_resource_overlapping(struct resource *r1, - struct resource *r2, - struct resource *ret) -{ - if (!resource_overlaps(r1, r2)) - return false; - ret->start = max(r1->start, r2->start); - ret->end = min(r1->end, r2->end); - return true; -} - struct catpt_ipc_msg { union { u32 header; diff --git a/sound/soc/intel/catpt/loader.c b/sound/soc/intel/catpt/loader.c index 40c22e4bb263..ff7b8f0d34ac 100644 --- a/sound/soc/intel/catpt/loader.c +++ b/sound/soc/intel/catpt/loader.c @@ -267,7 +267,7 @@ static int catpt_restore_fwimage(struct catpt_dev *cdev, r2.start = off; r2.end = r2.start + info->size - 1; - if (!catpt_resource_overlapping(&r2, &r1, &common)) + if (!resource_intersection(&r2, &r1, &common)) continue; /* calculate start offset of common data area */ off = common.start - r1.start; -- 2.29.2

4 8

[PATCH 1/2] ALSA: ppc: drop if block with always false condition
by Uwe Kleine-König 02 Dec '20

02 Dec '20

The remove callback is only called for devices that were probed successfully before. As the matching probe function cannot complete without error if dev->match_id != PS3_MATCH_ID_SOUND, we don't have to check this here. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/ppc/snd_ps3.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 58bb49fff184..6ab796a5d936 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -1053,8 +1053,6 @@ static int snd_ps3_driver_remove(struct ps3_system_bus_device *dev) { int ret; pr_info("%s:start id=%d\n", __func__, dev->match_id); - if (dev->match_id != PS3_MATCH_ID_SOUND) - return -ENXIO; /* * ctl and preallocate buffer will be freed in -- 2.29.2

6 15

[PATCH] soundwire: master: use pm_runtime_set_active() on add
by Bard Liao 02 Dec '20

02 Dec '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The 'master' device acts as a glue layer used during bus initialization only, and it needs to be 'transparent' for pm_runtime management. Its behavior should be that it becomes active when one of its children becomes active, and suspends when all of its children are suspended. In our tests on Intel platforms, we routinely see these sort of warnings on the initial boot: [ 21.447345] rt715 sdw:3:25d:715:0: runtime PM trying to activate child device sdw:3:25d:715:0 but parent (sdw-master-3) is not active This is root-caused to a missing setup to make the device 'active' on probe. Since we don't want the device to remain active forever after the probe, the autosuspend configuration is also enabled at the end of the probe - the device will actually autosuspend only in the case where there are no devices physically attached. In practice, the master device will suspend when all its children are no longer active. Fixes: bd84256e86ecf ('soundwire: master: enable pm runtime') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/master.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/drivers/soundwire/master.c b/drivers/soundwire/master.c index 3488bb824e84..9b05c9e25ebe 100644 --- a/drivers/soundwire/master.c +++ b/drivers/soundwire/master.c @@ -8,6 +8,15 @@ #include <linux/soundwire/sdw_type.h> #include "bus.h" +/* + * The 3s value for autosuspend will only be used if there are no + * devices physically attached on a bus segment. In practice enabling + * the bus operation will result in children devices become active and + * the master device will only suspend when all its children are no + * longer active. + */ +#define SDW_MASTER_SUSPEND_DELAY_MS 3000 + /* * The sysfs for properties reflects the MIPI description as given * in the MIPI DisCo spec @@ -154,7 +163,12 @@ int sdw_master_device_add(struct sdw_bus *bus, struct device *parent, bus->dev = &md->dev; bus->md = md; + pm_runtime_set_autosuspend_delay(&bus->md->dev, SDW_MASTER_SUSPEND_DELAY_MS); + pm_runtime_use_autosuspend(&bus->md->dev); + pm_runtime_mark_last_busy(&bus->md->dev); + pm_runtime_set_active(&bus->md->dev); pm_runtime_enable(&bus->md->dev); + pm_runtime_idle(&bus->md->dev); device_register_err: return ret; } -- 2.17.1

3 5

Sound BlasterX AE-5 not working
by Oliver Z. 02 Dec '20

02 Dec '20

Since a couple of month i am trying to get that sound card working. First it seemed to be a pulseaudio problem but now it looks like its an alsa/kernel issue. with some help from #alsa we figured out its the same problem as mentioned here: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/935 with everything connected: > jacks_do card=0 card=1 numid=51,iface=CARD,name='Line Out CLFE Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=49,iface=CARD,name='Line Out Front Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=50,iface=CARD,name='Line Out Surround Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=48,iface=CARD,name='Line Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=46,iface=CARD,name='Mic Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=47,iface=CARD,name='Mic Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on numid=54,iface=CARD,name='SPDIF In Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=52,iface=CARD,name='SPDIF Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=53,iface=CARD,name='SPDIF Phantom Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on card=2 numid=25,iface=CARD,name='HDMI/DP,pcm=10 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=31,iface=CARD,name='HDMI/DP,pcm=11 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=37,iface=CARD,name='HDMI/DP,pcm=12 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=1,iface=CARD,name='HDMI/DP,pcm=3 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off line out jack removed > jacks_do card=0 card=1 numid=51,iface=CARD,name='Line Out CLFE Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=49,iface=CARD,name='Line Out Front Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=50,iface=CARD,name='Line Out Surround Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=48,iface=CARD,name='Line Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=46,iface=CARD,name='Mic Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=47,iface=CARD,name='Mic Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=off numid=54,iface=CARD,name='SPDIF In Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=52,iface=CARD,name='SPDIF Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=53,iface=CARD,name='SPDIF Phantom Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on card=2 numid=25,iface=CARD,name='HDMI/DP,pcm=10 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=31,iface=CARD,name='HDMI/DP,pcm=11 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=37,iface=CARD,name='HDMI/DP,pcm=12 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=1,iface=CARD,name='HDMI/DP,pcm=3 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off and line out plugged in again > jacks_do card=0 card=1 numid=51,iface=CARD,name='Line Out CLFE Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=49,iface=CARD,name='Line Out Front Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=50,iface=CARD,name='Line Out Surround Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=48,iface=CARD,name='Line Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=46,iface=CARD,name='Mic Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=47,iface=CARD,name='Mic Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on numid=54,iface=CARD,name='SPDIF In Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=52,iface=CARD,name='SPDIF Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=53,iface=CARD,name='SPDIF Phantom Jack',index=1 ; type=BOOLEAN,access=r-------,values=1 : values=on card=2 numid=25,iface=CARD,name='HDMI/DP,pcm=10 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=31,iface=CARD,name='HDMI/DP,pcm=11 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=37,iface=CARD,name='HDMI/DP,pcm=12 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=1,iface=CARD,name='HDMI/DP,pcm=3 Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off i would love to have this soundcard working. i had to buy a cheap usb soundcard now to get my headset working on linux (fedora 33). on windows it works as expected

1 0