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[PATCH v2 00/11] Support for CS42L83 on Apple machines
by Martin Povišer 20 Sep '22

20 Sep '22

[Changes for v2: Collected ack on compatible. Addressed Richard's comments pertaining to error handling. Rebased. Switched to dev_err_probe() in few places (added patch #11). Fixed authorship of one of Richard's patches.] Hi all, there's a CS42L83 headphone jack codec found in Apple computers (in the recent 'Apple Silicon' ones as well as in earlier models, one example [1]). The part isn't publicly documented, but it appears almost identical to CS42L42, for which we have a driver in kernel. This series adapts the CS42L42 driver to the new part, and makes one change in anticipation of a machine driver for the Apple computers. Patch 1 adds new compatible to the cs42l42 schema. Patches 2 to 7 are taken from Richard's recent series [2] adding soundwire support to cs42l42. They are useful refactorings to build on in the later patches, and also this way our work doesn't diverge. (I fixed missing free_irq path in cs42l42_init, did s/Soundwire/SoundWire/ in changelogs, rebased.) Patch 8 exports some regmap-related symbols from cs42l42.c so they can be used to create cs42l83 regmap in cs42l83-i2c.c later. Patch 9 is the cs42l83 support proper. Patch 10 implements 'set_bclk_ratio' on the cs42l42 core. This will be called by the upcoming ASoC machine driver for 'Apple Silicon' Macs. (We have touched on this change to be made in earlier discussion, see [3] and replies.) Patch 11 brings cs42l42-i2c.c in sync with cs42l83-i2c.c on dev_err_probe() usage. Best, Martin [1] https://www.ifixit.com/Teardown/MacBook+Pro+13-Inch+Touch+Bar+2018+Teardown… [2] https://lore.kernel.org/alsa-devel/20220819125230.42731-1-rf@opensource.cir… [3] https://lore.kernel.org/asahi/8961DDD2-93FF-4A18-BCA2-90FCE298F517@cutebit.… Martin Povišer (5): ASoC: dt-bindings: cs42l42: Add 'cs42l83' compatible ASoC: cs42l42: Export regmap elements to core namespace ASoC: cs42l83: Extend CS42L42 support to new part ASoC: cs42l42: Implement 'set_bclk_ratio' ASoC: cs42l42: Switch to dev_err_probe() helper Richard Fitzgerald (6): ASoC: cs42l42: Add bitclock frequency argument to cs42l42_pll_config() ASoC: cs42l42: Use cs42l42->dev instead of &i2c_client->dev ASoC: cs42l42: Split probe() and remove() into stages ASoC: cs42l42: Split cs42l42_resume into two functions ASoC: cs42l42: Pass component and dai defs into common probe ASoC: cs42l42: Split I2C identity into separate module .../bindings/sound/cirrus,cs42l42.yaml | 1 + MAINTAINERS | 1 + include/sound/cs42l42.h | 1 + sound/soc/codecs/Kconfig | 15 +- sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/cs42l42-i2c.c | 106 +++++++ sound/soc/codecs/cs42l42.c | 259 +++++++++--------- sound/soc/codecs/cs42l42.h | 24 +- sound/soc/codecs/cs42l83-i2c.c | 242 ++++++++++++++++ 9 files changed, 529 insertions(+), 126 deletions(-) create mode 100644 sound/soc/codecs/cs42l42-i2c.c create mode 100644 sound/soc/codecs/cs42l83-i2c.c -- 2.33.0

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"no soundcard" for GeminiLake High Definition Audio (rev 06)
by Linas Vepstas 20 Sep '22

20 Sep '22

Kernel reports "no soundcard". Presumably, this is why I don't have sound. Let me dive right in with details: cat /proc/asound/cards --- no soundcards --- lspci: 00:0e.0 Multimedia audio controller: Intel Corporation Celeron/Pentium Silver Processor High Definition Audio (rev 06) 00:0e.0 0401: 8086:3198 (rev 06) DeviceName: Onboard - Sound Subsystem: Device 02f3:f000 Kernel driver in use: sof-audio-pci-intel-apl Kernel modules: snd_hda_intel, snd_soc_skl, snd_sof_pci_intel_apl dmesg |grep -iE 'snd|sof|audio' [ 0.201569] ACPI: Added _OSI(Linux-Lenovo-NV-HDMI-Audio) [ 4.715400] snd_hda_intel 0000:00:0e.0: DSP detected with PCI class/subclass/prog-if info 0x040100 [ 4.938181] snd_soc_skl 0000:00:0e.0: DSP detected with PCI class/subclass/prog-if info 0x040100 [ 5.324959] sof-audio-pci-intel-apl 0000:00:0e.0: DSP detected with PCI class/subclass/prog-if info 0x040100 [ 5.325268] sof-audio-pci-intel-apl 0000:00:0e.0: DSP detected with PCI class/subclass/prog-if 0x040100 [ 5.325465] sof-audio-pci-intel-apl 0000:00:0e.0: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [ 5.357483] sof-audio-pci-intel-apl 0000:00:0e.0: use msi interrupt mode [ 5.393933] sof-audio-pci-intel-apl 0000:00:0e.0: NHLT_DEVICE_I2S detected, ssp_mask 0x5 [ 5.393945] sof-audio-pci-intel-apl 0000:00:0e.0: hda codecs found, mask 4 [ 5.399557] sof-audio-pci-intel-apl 0000:00:0e.0: Firmware info: version 2:1:1-3964a [ 5.399567] sof-audio-pci-intel-apl 0000:00:0e.0: Firmware: ABI 3:21:0 Kernel ABI 3:23:0 [ 5.399579] sof-audio-pci-intel-apl 0000:00:0e.0: unknown sof_ext_man header type 3 size 0x30 [ 5.439596] sof-audio-pci-intel-apl 0000:00:0e.0: Firmware info: version 2:1:1-3964a [ 5.439604] sof-audio-pci-intel-apl 0000:00:0e.0: Firmware: ABI 3:21:0 Kernel ABI 3:23:0 cat /proc/version Linux version 6.0.0-rc5 (root@yippie) (gcc (Debian 12.2.0-1) 12.2.0, GNU ld (GNU Binutils for Debian) 2.38.90.20220713) #3 SMP PREEMPT_DYNAMIC Thu Sep 15 04:35:50 CDT 2022 I built this a few days ago from `make oldconfig` on the current Debian testing 5.19.15 kernel. I attempted debugging, and I found this: sound/soc/sof/intel/pci-apl.c: { PCI_DEVICE(0x8086, 0x3198), /* GeminiLake */ .driver_data = (unsigned long)&glk_desc}, Yay! That's my pci id! Adding printk's to both int snd_card_register(struct snd_card *card) int snd_soc_register_card(struct snd_soc_card *card) shows that neither of these are ever called. I was unable to figure out where they should have been called from. It's as if some driver is missing ??? There's nothing in sound/soc/intel/boards/* that seems to apply to my situation??? FWIW, more about this hardware: -- It's a cheap laptop, from newegg, Ipason MaxBook P1X, 4-core Intel Celeron, 12GB RAM, great price. -- lspci 00:00.0 Host bridge: Intel Corporation Gemini Lake Host Bridge (rev 06) 00:00.1 Signal processing controller: Intel Corporation Celeron/Pentium Silver Processor Dynamic Platform and Thermal Framework Processor Participant (rev 06) 00:02.0 VGA compatible controller: Intel Corporation GeminiLake [UHD Graphics 600] (rev 06) 00:0e.0 Multimedia audio controller: Intel Corporation Celeron/Pentium Silver Processor High Definition Audio (rev 06) 00:0f.0 Communication controller: Intel Corporation Celeron/Pentium Silver Processor Trusted Execution Engine Interface (rev 06) 00:12.0 SATA controller: Intel Corporation Celeron/Pentium Silver Processor SATA Controller (rev 06) 00:13.0 PCI bridge: Intel Corporation Gemini Lake PCI Express Root Port (rev f6) ... 00:15.0 USB controller: Intel Corporation Celeron/Pentium Silver Processor USB 3.0 xHCI Controller (rev 06) 00:16.0 Signal processing controller: Intel Corporation Celeron/Pentium Silver Processor I2C 0 (rev 06) ... 00:18.0 Signal processing controller: Intel Corporation Celeron/Pentium Silver Processor Serial IO UART Host Controller (rev 06) ... 00:1c.0 SD Host controller: Intel Corporation Celeron/Pentium Silver Processor SDA Standard Compliant SD Host Controller (rev 06) 00:1e.0 SD Host controller: Intel Corporation Device 31d0 (rev 06) 00:1f.0 ISA bridge: Intel Corporation Celeron/Pentium Silver Processor LPC Controller (rev 06) 00:1f.1 SMBus: Intel Corporation Celeron/Pentium Silver Processor Gaussian Mixture Model (rev 06) 01:00.0 Network controller: Realtek Semiconductor Co., Ltd. RTL8821CE 802.11ac PCIe Wireless Network Adapter So I guess there's "no soundcard" because the thing is wired directly into the PCI bus ?? I used to know PCI intimately, but that was a decade ago. I've forgotten all of that. Advice on how to proceed would be appreciated. I could maybe even write a device driver if someone holds my hand (I've written or worked on drivers for ethernet, scsi before) -- Linas -- Patrick: Are they laughing at us? Sponge Bob: No, Patrick, they are laughing next to us.

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[PATCH] ALSA: usb-audio: Properly refcounting clock rate
by Takashi Iwai 20 Sep '22

20 Sep '22

We fixed the bug introduced by the patch for managing the shared clocks at the commit 809f44a0cc5a ("ALSA: usb-audio: Clear fixed clock rate at closing EP"), but it was merely a workaround. By this change, the clock reference rate is cleared at each EP close, hence the still remaining EP may need a re-setup of rate unnecessarily. This patch introduces the proper refcounting for the clock reference object so that the clock setup is done only when needed. Fixes: 809f44a0cc5a ("ALSA: usb-audio: Clear fixed clock rate at closing EP") Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0c94ebc98e90..b2d0b42b581f 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -39,6 +39,7 @@ struct snd_usb_iface_ref { struct snd_usb_clock_ref { unsigned char clock; atomic_t locked; + int opened; int rate; bool need_setup; struct list_head list; @@ -803,6 +804,7 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep = NULL; goto unlock; } + ep->clock_ref->opened++; } ep->cur_audiofmt = fp; @@ -926,8 +928,10 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, endpoint_set_interface(chip, ep, false); if (!--ep->opened) { - if (ep->clock_ref && !atomic_read(&ep->clock_ref->locked)) - ep->clock_ref->rate = 0; + if (ep->clock_ref) { + if (!--ep->clock_ref->opened) + ep->clock_ref->rate = 0; + } ep->iface = 0; ep->altsetting = 0; ep->cur_audiofmt = NULL; @@ -1649,8 +1653,7 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending) WRITE_ONCE(ep->sync_source->sync_sink, NULL); stop_urbs(ep, false, keep_pending); if (ep->clock_ref) - if (!atomic_dec_return(&ep->clock_ref->locked)) - ep->clock_ref->rate = 0; + atomic_dec(&ep->clock_ref->locked); } } -- 2.35.3

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ACP6x: UCM not working
by GitHub issues - opened 20 Sep '22

20 Sep '22

alsa-project/alsa-ucm-conf issue #215 was opened from sebastianst: My Ryzen 6000 laptop has an AMD audio coprocessor driving the microphone with the ACP6x kernel module. #138 added initial support for this, but me and some other people on different Ryzen 6000 laptops have issues with alsa-ucm not configuring the microphone properly. We have to disable `alsa-ucm` to get any sound input or output working with pulseaudio, see https://bbs.archlinux.org/viewtopic.php?pid=2058087#p2058087 for the relevant discussion. Note that we first had to add our laptops to some quirk table of the kernel module to get it working at all. Any idea how to fix this? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/215 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH -next] slimbus: qcom-ngd: Add error handling in of_qcom_slim_ngd_register
by Lin Yujun 20 Sep '22

20 Sep '22

No error handling is performed when platform_device_add() return fails. Refer to the error handling of driver_set_override(), add error handling for platform_device_add(). Fixes: 917809e2280b ("slimbus: ngd: Add qcom SLIMBus NGD driver") Signed-off-by: Lin Yujun <linyujun809(a)huawei.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 0aa8408464ad..fb591990ddad 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1470,7 +1470,13 @@ static int of_qcom_slim_ngd_register(struct device *parent, ngd->pdev->dev.of_node = node; ctrl->ngd = ngd; - platform_device_add(ngd->pdev); + ret = platform_device_add(ngd->pdev); + if (ret) { + platform_device_put(ngd->pdev); + kfree(ngd); + of_node_put(node); + return ret; + } ngd->base = ctrl->base + ngd->id * data->offset + (ngd->id - 1) * data->size; -- 2.17.1

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[PATCH] Revert "ALSA: usb-audio: Split endpoint setups for hw_params and prepare"
by Takashi Iwai 20 Sep '22

20 Sep '22

This reverts commit ff878b408a03bef5d610b7e2302702e16a53636e. Unfortunately the recent fix seems bringing another regressions with PulseAudio / pipewire, at least for Steinberg and MOTU devices. As a temporary solution, do a straight revert. The issue for Android will be revisited again later by another different fix (if any). Fixes: ff878b408a03 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216500 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 23 ++++++++++++++--------- sound/usb/endpoint.h | 6 ++---- sound/usb/pcm.c | 14 ++++---------- 3 files changed, 20 insertions(+), 23 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 8c8f9a851f89..eb71df9da831 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -758,8 +758,7 @@ bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, * The endpoint needs to be closed via snd_usb_endpoint_close() later. * * Note that this function doesn't configure the endpoint. The substream - * needs to set it up later via snd_usb_endpoint_set_params() and - * snd_usb_endpoint_prepare(). + * needs to set it up later via snd_usb_endpoint_configure(). */ struct snd_usb_endpoint * snd_usb_endpoint_open(struct snd_usb_audio *chip, @@ -1293,13 +1292,12 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep) /* * snd_usb_endpoint_set_params: configure an snd_usb_endpoint * - * It's called either from hw_params callback. * Determine the number of URBs to be used on this endpoint. * An endpoint must be configured before it can be started. * An endpoint that is already running can not be reconfigured. */ -int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { const struct audioformat *fmt = ep->cur_audiofmt; int err; @@ -1382,18 +1380,18 @@ static int init_sample_rate(struct snd_usb_audio *chip, } /* - * snd_usb_endpoint_prepare: Prepare the endpoint + * snd_usb_endpoint_configure: Configure the endpoint * * This function sets up the EP to be fully usable state. - * It's called either from prepare callback. + * It's called either from hw_params or prepare callback. * The function checks need_setup flag, and performs nothing unless needed, * so it's safe to call this multiple times. * * This returns zero if unchanged, 1 if the configuration has changed, * or a negative error code. */ -int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_configure(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { bool iface_first; int err = 0; @@ -1414,6 +1412,9 @@ int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, if (err < 0) goto unlock; } + err = snd_usb_endpoint_set_params(chip, ep); + if (err < 0) + goto unlock; goto done; } @@ -1441,6 +1442,10 @@ int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, if (err < 0) goto unlock; + err = snd_usb_endpoint_set_params(chip, ep); + if (err < 0) + goto unlock; + err = snd_usb_select_mode_quirk(chip, ep->cur_audiofmt); if (err < 0) goto unlock; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index e67ea28faa54..6a9af04cf175 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -17,10 +17,8 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, bool is_sync_ep); void snd_usb_endpoint_close(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); -int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep); -int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep); +int snd_usb_endpoint_configure(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock); bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b604f7e95e82..d45d1d7e6664 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -443,17 +443,17 @@ static int configure_endpoints(struct snd_usb_audio *chip, if (stop_endpoints(subs, false)) sync_pending_stops(subs); if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); if (err < 0) return err; } - err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); + err = snd_usb_endpoint_configure(chip, subs->data_endpoint); if (err < 0) return err; snd_usb_set_format_quirk(subs, subs->cur_audiofmt); } else { if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); if (err < 0) return err; } @@ -551,13 +551,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, subs->cur_audiofmt = fmt; mutex_unlock(&chip->mutex); - if (subs->sync_endpoint) { - ret = snd_usb_endpoint_set_params(chip, subs->sync_endpoint); - if (ret < 0) - goto unlock; - } - - ret = snd_usb_endpoint_set_params(chip, subs->data_endpoint); + ret = configure_endpoints(chip, subs); unlock: if (ret < 0) -- 2.35.3

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[PATCH linux-next] ALSA: es18xx: Remove the unneeded result variable
by cgel.zte＠gmail.com 20 Sep '22

20 Sep '22

From: ye xingchen <ye.xingchen(a)zte.com.cn> Return the value inb() directly instead of storing it in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: ye xingchen <ye.xingchen(a)zte.com.cn> --- sound/isa/es18xx.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 3fcd168480b6..0a32845b1017 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1344,11 +1344,8 @@ ES18XX_SINGLE("GPO1 Switch", 0, ES18XX_PM, 1, 1, ES18XX_FL_PMPORT), static int snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { - int data; - outb(reg, chip->ctrl_port); - data = inb(chip->ctrl_port + 1); - return data; + return inb(chip->ctrl_port + 1); } static void snd_es18xx_config_write(struct snd_es18xx *chip, -- 2.25.1

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[PATCH] soundwire: bus: Fix wrong port number in sdw_handle_slave_alerts()
by Richard Fitzgerald 20 Sep '22

20 Sep '22

for_each_set_bit() gives the bit-number counting from 0 (LSbit==0). When processing INTSTAT2, bit 0 is DP4 so the port number is (bit + 4). Likewise for INTSTAT3 bit 0 is DP11 so port number is (bit + 11). Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 8eded1a55227..df0ae869ee51 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1622,7 +1622,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) port = buf2[0] & SDW_SCP_INTSTAT2_PORT4_10; for_each_set_bit(bit, &port, 8) { /* scp2 ports start from 4 */ - port_num = bit + 3; + port_num = bit + 4; sdw_handle_port_interrupt(slave, port_num, &port_status[port_num]); @@ -1634,7 +1634,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) port = buf2[1] & SDW_SCP_INTSTAT3_PORT11_14; for_each_set_bit(bit, &port, 8) { /* scp3 ports start from 11 */ - port_num = bit + 10; + port_num = bit + 11; sdw_handle_port_interrupt(slave, port_num, &port_status[port_num]); -- 2.30.2

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HDA HDMI PCM device allocation
by Jaroslav Kysela 20 Sep '22

20 Sep '22

Hi, I am trying to fix some issues in UCM for the HDA HDMI devices [1][2]. I would like to summary the current situation at first (correct me, if I miss something): We have two methods to map the PINs in the HDA HDMI driver to the PCM devices (legacy/static - 1:1 mapping, dynamic - used for new devices with the MST capability). There is also set of converters in each HDMI codec and the number of simultaneously used PCM devices cannot go beyond this count of converters in hardware (otherwise -EBUSY error is returned). The count of converters is 3 or 4 depending on the hardware. Things to discuss: It seems quite straight to limit the count of created PCMs to the count of converters. We cannot use more anyway and it does not help, if more PCM devices are allocated (and Jacks reported) to applications when they cannot be used simultaneously. Legacy/static mapping should be converted to dynamic (unless the count of created PCM devices is equal to the count of codec converters). There may be 1:1 mapping between the converter and the PCM device to make things easier. There is a corner case, when more HDMI devices are connected than the count of converters. In this case, an extra method (a module parameter and/or a control element and/or procfs) may be used to filter unwanted HDMI devices. It may be a bit difficult to select the proper filtering key - it may be the PIN/MST device hash or so. The driver may report this key in eld#* files (procfs). Impact to applications: Those days, pulseaudio or pipewire servers are mostly used on the current hardware. Both servers share the legacy probe code for HDMI devices - they are trying to open PCM devices sequentially and check for the error code. There should not be a problem when the connected HDMI devices do not go beyond the count of converters. A minor issue is that the name of the used sink/port may be different (users may be forced to reselect the output path). For other applications, the PCM device assigned to the connected HDMI device may be different (available in a different ALSA device name). I do not think that it's a big issue. It should be easy solvable with an updated software configuration. Let me know about your opinion about this. Thank you, Jaroslav [1] https://github.com/alsa-project/alsa-lib/issues/245 [2] https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2481 -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[PATCH 0/8] ALSA: hda: document/simplify hdac_ext handling
by Pierre-Louis Bossart 20 Sep '22

20 Sep '22

This sound/hda/ext library is a confusing magic black box that very few understand. It needn't be that way, we can document/simplify and make the code clearer. Pierre-Louis Bossart (8): ALSA: hda: make snd_hdac_stream_clear() static ALSA: hda: document state machine for hdac_streams ALSA: hda: ext: make snd_hdac_ext_stream_init() static ALSA: hda: Use hdac_ext prefix in snd_hdac_stream_free_all() for clarity ALSA: hda: add snd_hdac_stop_streams() helper ALSA: hda: ext: simplify logic for stream assignment ALSA: hda: ext: fix locking in stream_release ALSA: hda: ext: remove always-true conditions on host and link release include/sound/hdaudio.h | 3 +- include/sound/hdaudio_ext.h | 5 +-- sound/hda/ext/hdac_ext_stream.c | 34 +++++++-------- sound/hda/hdac_stream.c | 74 +++++++++++++++++++++++++++++---- sound/pci/hda/hda_controller.c | 4 +- sound/soc/intel/avs/core.c | 4 +- sound/soc/intel/skylake/skl.c | 2 +- 7 files changed, 87 insertions(+), 39 deletions(-) -- 2.34.1

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