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

[PATCH] Use card->dev in replace of the &pdev->dev argument in the dev_err function
by Liu Jing 18 Oct '24

18 Oct '24

Because card->dev = &pdev->dev is already defined in the rx51_soc_probe function, and then &pdev->dev is still used. Signed-off-by: Liu Jing <liujing(a)cmss.chinamobile.com> --- sound/soc/ti/rx51.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c index 77296237575a..d9900c69e536 100644 --- a/sound/soc/ti/rx51.c +++ b/sound/soc/ti/rx51.c @@ -371,7 +371,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,cpu-dai", 0); if (!dai_node) { - dev_err(&pdev->dev, "McBSP node is not provided\n"); + dev_err(card->dev, "McBSP node is not provided\n"); return -EINVAL; } rx51_dai[0].cpus->dai_name = NULL; @@ -381,7 +381,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,audio-codec", 0); if (!dai_node) { - dev_err(&pdev->dev, "Codec node is not provided\n"); + dev_err(card->dev, "Codec node is not provided\n"); return -EINVAL; } rx51_dai[0].codecs->name = NULL; @@ -389,7 +389,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,audio-codec", 1); if (!dai_node) { - dev_err(&pdev->dev, "Auxiliary Codec node is not provided\n"); + dev_err(card->dev, "Auxiliary Codec node is not provided\n"); return -EINVAL; } rx51_aux_dev[0].dlc.name = NULL; @@ -399,7 +399,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,headphone-amplifier", 0); if (!dai_node) { - dev_err(&pdev->dev, "Headphone amplifier node is not provided\n"); + dev_err(card->dev, "Headphone amplifier node is not provided\n"); return -EINVAL; } rx51_aux_dev[1].dlc.name = NULL; @@ -408,7 +408,7 @@ static int rx51_soc_probe(struct platform_device *pdev) rx51_codec_conf[1].dlc.of_node = dai_node; } - pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); + pdata = devm_kzalloc(card->dev, sizeof(*pdata), GFP_KERNEL); if (pdata == NULL) return -ENOMEM; @@ -439,7 +439,7 @@ static int rx51_soc_probe(struct platform_device *pdev) err = devm_snd_soc_register_card(card->dev, card); if (err) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err); + dev_err(card->dev, "snd_soc_register_card failed (%d)\n", err); return err; } -- 2.27.0

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[PATCH 5.10 0/1] soundwire: cadence: Fix error check in cdns_xfer_msg()
by Daniil Dulov 18 Oct '24

18 Oct '24

Svacer reports redundant comparison in cdns_xfer_msg(). The problem is present in 5.10 stable release and can be fixed by the following upstream patch that can be cleanly applied to 5.10 stable branch.

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[PATCH alsa-ucm-conf v5 1/2] sof-soundwire: Change map control names to make them unique and user friendly
by Maciej Strozek 18 Oct '24

18 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v5: Change PlaybackMixerElem to just "Speaker" v4: Add "CaptureMixer" to headset config v3: Make Headset Microphone more user-friendly and contain the word "Capture" v2: Rename to "Speaker Playback" instead of "Speaker Combined" --- ucm2/codecs/cs42l43/init.conf | 10 ++++++++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- ucm2/sof-soundwire/cs42l43.conf | 3 ++- 4 files changed, 14 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..b2796e2 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,13 +2,19 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } + "name='cs42l43 Headset Microphone Capture Switch'" { + "name='cs42l43 Decimator 1 Switch'".vindex.0 0 + } + "name='cs42l43 Headset Microphone Capture Volume'" { + "name='cs42l43 Decimator 1 Volume'".vindex.0 0 + } } } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..176f880 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Playback Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Playback Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 1702cf0..ddca67a 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -40,7 +40,8 @@ SectionDevice."Headset" { Value { CapturePriority 200 CapturePCM "hw:${CardId},4" - CaptureMixerElem "cs42l43 Decimator 1" + CaptureMixer "default:${CardId}" + CaptureMixerElem "cs42l43 Headset Microphone" JackControl "Headset Mic Jack" } } -- 2.39.5

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Qualcomm: fix codec initialisation
by GitHub pull_request - opened 18 Oct '24

18 Oct '24

alsa-project/alsa-ucm-conf pull request #454 was opened from jhovold: Fix the Qualcomm configurations that are using the same identifier for lazy includes of codec initialisation files which means that only the last one is actually included. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/454 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/454.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Revert "ucm2: Qualcomm: x1e80100: add USB DisplayPort playback"
by GitHub pull_request - opened 18 Oct '24

18 Oct '24

alsa-project/alsa-ucm-conf pull request #453 was opened from jhovold: This reverts commit 1dcb58e4fac989bafe0705e49a7f029727207fb3. The recently added DisplayPort support depends on kernel changes that never made it upstream and has some usability issues that remain to be resolved. Revert for now to avoid breaking existing setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/453 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/453.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument.
by GitHub issues - opened 17 Oct '24

17 Oct '24

alsa-project/alsa-plugins issue #62 was opened from linuxandos2: Playing a flac with a samplerate higer than 96000 with audacious gives: > _ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument._ Output device for audacious is hw:1.0 - USB Audio the device is a soudbaster blasterx G6. output of _cat /proc/asound/card1/stream0_ > Creative Technology Ltd Sound BlasterX G6 at usb-0000:00:14.0-4.1, high speed : > USB Audio > > Playback: > Status: Stop > Interface 1 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 1 > Implicit Feedback Mode: No > Interface 1 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 2 > Implicit Feedback Mode: No > > Capture: > Status: Stop > Interface 2 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Interface 2 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > linux: gentoo kernel: 6.9.9-gentoo-x86_64 alsa-lib-1.2.12 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.12 alsa-plugins-1.2.12 alsa-utils-1.2.12 selecting > pulse - PulseAudio Sound Server in audacious results in no sound and audacious hangs. Pipewire and wireplumber are used. On an old gentoo- system on another partition all samplerates play successfully. Using the same kernel: linux-6.8.1-gentoo as in the old system on the new one does not solve the problem. the versions of alsa are: alsa-lib-1.2.10-r2 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.10-r1 alsa-plugins-1.2.7.1-r1 alsa-utils-1.2.10-r1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/62 Repository URL: https://github.com/alsa-project/alsa-plugins

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acppdmmach: add support for ACP 7.0
by GitHub pull_request - opened 17 Oct '24

17 Oct '24

alsa-project/alsa-ucm-conf pull request #452 was opened from superm1: Starting with ACP 7.0 there is a generic 'acppdmmach' driver that will be utilized. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/452 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/452.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH alsa-ucm-conf v4 1/2] sof-soundwire: Change map control names to make them unique and user friendly
by Maciej Strozek 17 Oct '24

17 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v4: Add "CaptureMixer" to headset config v3: Make Headset Microphone more user-friendly and contain the word "Capture" v2: Rename to "Speaker Playback" instead of "Speaker Combined" --- ucm2/codecs/cs42l43/init.conf | 10 ++++++++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- ucm2/sof-soundwire/cs42l43.conf | 3 ++- 4 files changed, 14 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..b2796e2 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,13 +2,19 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } + "name='cs42l43 Headset Microphone Capture Switch'" { + "name='cs42l43 Decimator 1 Switch'".vindex.0 0 + } + "name='cs42l43 Headset Microphone Capture Volume'" { + "name='cs42l43 Decimator 1 Volume'".vindex.0 0 + } } } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..a3cf6ad 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Playback Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Playback Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker Playback" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 1702cf0..ddca67a 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -40,7 +40,8 @@ SectionDevice."Headset" { Value { CapturePriority 200 CapturePCM "hw:${CardId},4" - CaptureMixerElem "cs42l43 Decimator 1" + CaptureMixer "default:${CardId}" + CaptureMixerElem "cs42l43 Headset Microphone" JackControl "Headset Mic Jack" } } -- 2.39.5

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MUSL static build - conf.c still insists on using dynamic link libraries
by GitHub issues - opened 16 Oct '24

16 Oct '24

alsa-project/alsa-lib issue #411 was opened from pietervandermeer: alsa-lib v 1.2.12 configured for musl static build like so: ```CC=arm-linux-musleabihf-gcc ./configure --enable-shared=no --enable-static=yes --with-pic --host=armv7-unknown-linux-musleabihf --prefix=/home/vagrant/alsa-lib-1.2.12/build --libdir=/usr/lib/arm-linux-musleabihf --includedir=/usr/include/arm-linux-musleabihf/``` This builds and links fine against my executable. But when I run it, there's an issue: ``` INFO opening alsa audio playback device [hw:CARD=sndrpihifiberry,DEV=0] ALSA lib conf.c:3558:(snd_config_hooks_call) Cannot open shared library (null) ((null): Dynamic loading not supported) ALSA lib conf.c:4026:(snd_config_update_r) hooks failed, removing configuration WARN Not found playback device, error ALSA function 'snd_pcm_open' failed with error 'No such file or directory (2)'. Trying next.. ``` Note the "cannot open shared library". Well, yes, because you should be configured as 100% static, as per configure arguments ```--enable-shared=no --enable-static=yes``` Upon inspecting the code in ```conf.c```, there are various functions like ```snd_config_hooks_call()``` that just call ```snd_dlopen()``` and ```snd_dlsym()```. Right now I'm just replacing these with ```strcmp``` on the function_name followed by a static function call to the respective function. But I don't know how much work I'll need, and of course it is really hacky. It would be great if a 100% static build was supported. Issue URL : https://github.com/alsa-project/alsa-lib/issues/411 Repository URL: https://github.com/alsa-project/alsa-lib

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Working on audio for qcom apq8098 / msm8998
by Marc Gonzalez 15 Oct '24

15 Oct '24

Hello everyone, I've been trying to get audio working on a qcom apq8098 (cousin of msm8998) board. Code is mostly based on work from here: https://gitlab.com/msm8998-mainline/linux This is the diff I have on top of v6.12-rc1 https://paste.debian.net/1331666/ Kernel log after init starts: [ 13.311201] random: crng init done [ 14.724797] atl1c 0000:01:00.0 enp1s0: renamed from eth0 (while UP) [ 14.895978] Detected VIPT I-cache on CPU1 [ 14.896041] GICv3: CPU1: found redistributor 1 region 0:0x0000000017b20000 [ 14.896113] CPU1: Booted secondary processor 0x0000000001 [0x51af8014] [ 14.949128] Detected VIPT I-cache on CPU2 [ 14.949198] GICv3: CPU2: found redistributor 2 region 0:0x0000000017b40000 [ 14.949272] CPU2: Booted secondary processor 0x0000000002 [0x51af8014] [ 15.025271] Detected VIPT I-cache on CPU3 [ 15.025349] GICv3: CPU3: found redistributor 3 region 0:0x0000000017b60000 [ 15.025423] CPU3: Booted secondary processor 0x0000000003 [0x51af8014] [ 15.106412] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:3 [ 15.109935] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:4 [ 15.119041] qcom-q6afe aprsvc:apr-service:4:4: YO: q6afe_probe:1759 [ 15.134672] q6afe-dai 17300000.remoteproc:glink-edge:apr:apr-service@4:dais: YOA: q6afe_dai_dev_probe [ 15.137948] q6afe-dai 17300000.remoteproc:glink-edge:apr:apr-service@4:dais: YOA: of_q6afe_parse_dai_data [ 15.146461] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:7 [ 15.153563] remoteproc remoteproc0: powering up 4080000.remoteproc [ 15.166371] remoteproc remoteproc0: Booting fw image mba.mbn, size 234152 [ 15.169747] q6asm-dai 17300000.remoteproc:glink-edge:apr:apr-service@7:dais: Adding to iommu group 5 [ 15.170559] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:8 [ 15.179732] q6afe-dai 17300000.remoteproc:glink-edge:apr:apr-service@4:dais: YOA: msm_dai_q6_dai_probe [ 15.199345] qcom-q6afe aprsvc:apr-service:4:4: YO: q6afe_find_port:860 [ 15.208834] msm-snd-msm8998 sound: ASoC: registered pcm #0 MultiMedia1 (*) [ 15.216194] msm-snd-msm8998 sound: snd-soc-dummy-dai <-> MultiMedia1 mapping ok [ 15.221994] msm-snd-msm8998 sound: ASoC: registered pcm #1 MultiMedia2 (*) [ 15.229582] msm-snd-msm8998 sound: snd-soc-dummy-dai <-> MultiMedia2 mapping ok [ 15.236045] msm-snd-msm8998 sound: ASoC: registered pcm #2 (HDMI) [ 15.243211] msm-snd-msm8998 sound: i2s-hifi <-> HDMI mapping ok [ 15.285770] input: batfish Headset Jack as /devices/platform/sound/sound/card0/input2 [ 15.298010] qcom-q6v5-mss 4080000.remoteproc: MBA booted without debug policy, loading mpss [ 15.542937] remoteproc remoteproc0: remote processor 4080000.remoteproc is now up [ 17.727724] ath10k_snoc 18800000.wifi: qmi chip_id 0x30214 chip_family 0x4001 board_id 0xff soc_id 0x40010002 [ 17.727923] ath10k_snoc 18800000.wifi: qmi fw_version 0x100204b2 fw_build_timestamp 2019-09-04 03:01 fw_build_id QC_IMAGE_VERSION_STRING=WLAN.HL.1.0-01202-QCAHLSWMTPLZ-1.221523.2 [ 17.736928] ath10k_snoc 18800000.wifi: qmi not waiting for msa_ready indicator [ 17.903361] platform a800000.usb: deferred probe pending: dwc3: failed to initialize core [ 21.512409] arm-smmu 16c0000.iommu: Unhandled context fault: fsr=0x402, iova=0x00000000, fsynr=0x1, cbfrsynra=0x1900, cb=0 [ 21.512619] arm-smmu 16c0000.iommu: FSR = 00000402 [Format=2 TF], SID=0x1900 [ 21.522450] arm-smmu 16c0000.iommu: FSYNR0 = 00000001 [S1CBNDX=0 PLVL=1] [ 21.599761] ath10k_snoc 18800000.wifi: wcn3990 hw1.0 target 0x00000008 chip_id 0x00000000 sub 0000:0000 [ 21.599935] ath10k_snoc 18800000.wifi: kconfig debug 1 debugfs 1 tracing 0 dfs 0 testmode 0 [ 21.608221] ath10k_snoc 18800000.wifi: firmware ver api 5 features wowlan,mgmt-tx-by-reference,non-bmi crc32 3ec1ac4b [ 21.657531] ath10k_snoc 18800000.wifi: htt-ver 3.44 wmi-op 4 htt-op 3 cal file max-sta 32 raw 0 hwcrypto 1 [ 21.713363] arm-smmu 16c0000.iommu: Unhandled context fault: fsr=0x402, iova=0x00000000, fsynr=0x1, cbfrsynra=0x1900, cb=0 [ 21.713577] arm-smmu 16c0000.iommu: FSR = 00000402 [Format=2 TF], SID=0x1900 [ 21.723404] arm-smmu 16c0000.iommu: FSYNR0 = 00000001 [S1CBNDX=0 PLVL=1] [ 21.732374] ath10k_snoc 18800000.wifi: invalid MAC address; choosing random [ 21.737662] ath: EEPROM regdomain: 0x0 [ 21.744356] ath: EEPROM indicates default country code should be used [ 21.748148] ath: doing EEPROM country->regdmn map search [ 21.754648] ath: country maps to regdmn code: 0x3a [ 21.759986] ath: Country alpha2 being used: US [ 21.764605] ath: Regpair used: 0x3a Does anyone see something obvious I might have missed? Regards

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