October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v4 0/2] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 02 Jan '20

02 Jan '20

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the for-next. v4 changes: 1. Fixed wrong signed-off as correct mail suffix. 2. Fixed patch subject that add patch version. v3 changes: 1. https://patchwork.kernel.org/patch/11164283/ and https://patchwork.kernel.org/patch/11164305/ has been merged. 2. Change the name of mtk_wdt_compatible to mtk_wdt_data. 3. Remove toprgu_reset struct and use mtk_wdt_dev instead. 4. Get the value of sw_rst_num from .h file. 5. Adddd mt2712-resets.h for mt2712. 6. Improve commit message. v2 changes: 1. remove "WIP" that in the title of patches 2. add hyper link for the patch that depends on 3. patchwork list: https://patchwork.kernel.org/cover/11164285/ https://patchwork.kernel.org/patch/11164295/ https://patchwork.kernel.org/patch/11164299/ https://patchwork.kernel.org/patch/11164283/ https://patchwork.kernel.org/patch/11164305/ v1 changes: 1. patchwork list: https://patchwork.kernel.org/cover/11164173/ https://patchwork.kernel.org/patch/11164181/ https://patchwork.kernel.org/patch/11164185/ https://patchwork.kernel.org/patch/11164187/ https://patchwork.kernel.org/patch/11164175/ yong.liang (2): dt-bindings: mediatek: mt8183: Add #reset-cells watchdog: mtk_wdt: mt8183: Add reset controller .../devicetree/bindings/watchdog/mtk-wdt.txt | 10 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 111 +++++++++++++++++- .../reset-controller/mt2712-resets.h | 22 ++++ .../reset-controller/mt8183-resets.h | 15 +++ 5 files changed, 155 insertions(+), 4 deletions(-) create mode 100644 include/dt-bindings/reset-controller/mt2712-resets.h -- 2.18.0

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[alsa-devel] [PATCH v2 0/2] ALSA: usb-audio: fix PCM device order
by Takashi Iwai 30 Dec '19

30 Dec '19

Hi, this is a rewrite of the old USB-audio fix patch I obviously overlooked to adapt to 5.3 kernel with a preliminary cleanup patch. Totally untested, so any test / feedback would be appreciated. thanks, Takashi === Alexander Tsoy (1): ALSA: usb-audio: fix PCM device order Takashi Iwai (1): ALSA: usb-audio: Unify audioformat release code sound/usb/stream.c | 64 +++++++++++++++++++++++++++++++++++++++--------------- 1 file changed, 47 insertions(+), 17 deletions(-) -- 2.16.4

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Re: [alsa-devel] snd-bebob : from kernel 4.13 to 5.3.19 and .20
by Takashi Sakamoto 05 Dec '19

05 Dec '19

Hi, On Wed, Oct 30, 2019 at 10:40:18AM +0100, Jean-Paul Argudo wrote: > > I'll install Eoan kernel and test my devices. > > (but I don't have Saffire and Saffire LE...) I'd like you to wait > > for my test report. > > What I can say is that with kernel 4.13 under Ubunto Disco everything > was fine :-( I install Eoan (linux-image-5.3.0-19-generic:5.3.0-19.20) and test ALSA bebob driver with my test devices: - Yamaha GO46 (DM1000 ASIC) - M-Audio FireWire Solo (DM1000 ASIC) - Behringer FCA610 (DM1500 ASIC) - Focusrite SaffirePro 26 I/O (DM1000E) However I don't find similar discontinuities. This means that you encounter model-specific issue as regression I didn't realized. For my information, would you please get output from nodes on procfs? * /proc/asound/cardX/firewire/clock * /proc/asound/cardX/firewire/firmware * /proc/asound/cardX/firewire/formation Additionally, please install 'gir1.2-hinawa-2.0' package, then clone below remote repository: * https://github.com/takaswie/hinawa-utils This repository includes two scripts to get information from the device: * hinawa-bebob-parser * hinawa-bebob-connection-cui When your device is detected as sound card '2' and your device is also detected as '/dev/fw1', execute below: $ ./hinawa-bebob-connection-cui 2 graph-connections $ ./hinawa-bebob-parser 1 2 > At startup it lights green ok, but no sound is playable, then the > lights turn orange (like it is when it's not working), I hear a > "relay sound" (a electric clic of a relay), then, the Saffire LE > disapears from the sound menu in Ubuntu sound menu. Once your device is detected and ALSA bebob driver bound to it, pulseaudio detects it and start PCM substream to get capabilities of the device. In your case, the PCM substream is corrupted by the discontinuity. Then pulseaudio give up to use the device. As a result, sound menu drops entry of the device. I guess that the color of LED corresponds to the packet streaming. On Wed, Oct 30, 2019 at 11:12:01AM +0100, Jean-Paul Argudo wrote: > But something new and maybe interesting for you : > > Oct 30 09:25:25 deiphobe pulseaudio[2589]: E: [alsa-sink-BeBoB] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. > Oct 30 09:25:25 deiphobe pulseaudio[2589]: E: [alsa-sink-BeBoB] alsa- sink.c: Most likely this is a bug in the ALSA driver 'snd_bebob'. Please report this issue to the ALSA developers. > Oct 30 09:25:25 deiphobe pulseaudio[2589]: E: [alsa-sink-BeBoB] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Practically these logs can be ignored. (I work to suppress the above lines in development period for Linux kernel v5.5.) > **And in kern.log** : at the beggining in the boot sequence (?), > between FS-cache lines and usb ones: > > Oct 27 21:54:36 deiphobe kernel: [ 8.742105] FS-Cache: N-key=[8] '020001bdc0a800fe' > > Oct 27 21:54:40 deiphobe kernel: [ 12.768168] usb 3-4.1: reset high- speed USB device number 5 using xhci_hcd > Oct 27 21:54:45 deiphobe kernel: [ 18.294126] snd-bebob fw1.0: transaction failed: no ack > Oct 27 21:54:45 deiphobe kernel: [ 18.294167] snd-bebob fw1.0: fail to get section type for isoc out plug 0: -5 > Oct 27 21:54:45 deiphobe kernel: [ 18.294172] snd-bebob fw1.0: fail to run AMDTP slave stream:-5 > Oct 27 21:54:45 deiphobe kernel: [ 18.306399] firewire_ohci 0000:05:01.0: DMA context still active (0x00000411) > Oct 27 21:54:45 deiphobe kernel: [ 18.318425] firewire_ohci 0000:05:01.0: DMA context still active (0x00000411) > Oct 27 21:54:47 deiphobe kernel: [ 20.311440] irq_handler: 20 callbacks suppressed > Oct 27 21:54:47 deiphobe kernel: [ 20.311441] firewire_ohci 0000:05:01.0: isochronous cycle inconsistent > Oct 27 21:54:48 deiphobe kernel: [ 21.322730] firewire_core 0000:05:01.0: phy config: new root=ffc1, gap_count=5 > > Oct 27 21:54:50 deiphobe kernel: [ 22.970251] rfkill: input handler disabled > Oct 27 20:55:10 deiphobe kernel: [ 42.697284] usb 3-4.1: reset high- speed USB device number 5 using xhci_hcd IEEE 1394 bus 'bus-reset' state. Just before/after the bus-reset, any functionality easily fails. It's better to plug-in the device enough after bootup. Regards Takashi Sakamoto

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[alsa-devel] Surround speaker connection on Acer 8951G
by Sergey 'Jin' Bostandzhyan 28 Nov '19

28 Nov '19

Hi, I am not sure if this is the right place to ask or if I should just submit a bugreport, but you guys helped me out with a similar problem about 10 years ago, so I'll try here first. If I'm wrong here, then I apologize, just let me know. I have an Acer Aspire Ethos 8951G notebook which has 6 built in speakers for 5.1 surround sound, it's a Realtek ALC669X. The problem is that only two speakers are recognized by default. I tried fiddling around with hdajackretast and it does make a difference, but I somehow could not find a step-by-step approach on how to figure out what is what and how to map everything correctly, so some guidance would be very much appreciated. I even ended up in situations where I got channels supporting only mute which would influence muting of other channels, I really got lost there.. Here's my alsa-info output: http://alsa-project.org/db/?f=3adde87164f5fc349c3c78b211ee63e416ebf10b It would be great if we could figure it out and make it work by default. Basically that's what we did 10 years ago, back then I had an Acer Aspire 8920G with a ALC889 which had the same problem that got fixed by you and these days it just works out of the box :) Kind regards, Jin

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[alsa-devel] Focusrite Scarlett 6i6 gen1 - input handling fix
by Jens Verwiebe 19 Nov '19

19 Nov '19

The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage. This patch introduces this functionality as to be seen in the mac or windows scarlett control. The correct address could already be found in the dump info, but was never used. Cheers ... Jens Signed-off-by: Jens Verwiebe <info(a)jensverwiebe.de> diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 7438e7c..f3bdac3 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -152,6 +152,7 @@ enum { SCARLETT_OUTPUTS, SCARLETT_SWITCH_IMPEDANCE, SCARLETT_SWITCH_PAD, + SCARLETT_SWITCH_GAIN, }; enum { @@ -202,6 +203,15 @@ struct scarlett_device_info { } }; +static const struct scarlett_mixer_elem_enum_info opt_gain = { + .start = 0, + .len = 2, + .offsets = {}, + .names = (char const * const []){ + "Lo", "Hi" + } +}; + static const struct scarlett_mixer_elem_enum_info opt_impedance = { .start = 0, .len = 2, @@ -664,8 +674,8 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, { .num = 1, .type = SCARLETT_SWITCH_PAD, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, }, .matrix_mux_init = { @@ -895,6 +905,15 @@ static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, if (err < 0) return err; break; + case SCARLETT_SWITCH_GAIN: + sprintf(mx, "Input %d Gain Switch", ctl->num); + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, + scarlett_ctl_enum_resume, 0x01, + 0x08, ctl->num, USB_MIXER_S16, 1, mx, + &opt_gain, &elem); + if (err < 0) + return err; + break; } } -- Jens Verwiebe Allerskehre 44 - 22309 Hamburg Tel.: +49 40 68 78 50 mobile: +49 172 400 49 07 mailto: info(a)jensverwiebe.de web: http://www.jensverwiebe.de

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[alsa-devel] parameter for pulse device?
by frederik＠ofb.net 16 Nov '19

16 Nov '19

Dear ALSA, In my ~/.asoundrc, I have lines like this: pcm.mic { type pulse; device "mic" } pcm.monitor { type pulse; device "monitor"; } pcm.music { type pulse; device "music"; } to make it easier for ALSA-aware programs to input and output via PulseAudio, e.g.: ecasound -i alsa,mic -o alsa,monitor -etd:... However, I would like to simplify this and not have to update ~/.asoundrc every time I create a new PulseAudio device. Since ALSA has the ability for PCMs to take a parameter, I thought this might work with the "pulse" PCM and the PulseAudio device name. But I get an "error: Invalid argument" when trying to pass the device name as an argument to the "pulse" PCM: $ ecasound -o alsa,pulse:music -i some.wav ... ALSA lib conf.c:5014:(snd_config_expand) Unknown parameters music ALSA lib pcm.c:2564:(snd_pcm_open_noupdate) Unknown PCM pulse:music ERROR: Connecting chainsetup failed: "Enabling chainsetup: AUDIOIO-ALSA: ... Unable to open ALSA-device for playback; error: Invalid argument" Is there some magic with macros that I can use to accomplish this syntax, or can we add the ability for the "pulse" PCM to take a parameter naming the device? Thanks, Frederick

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[alsa-devel] [PATCH 0/4] soundwire: update ASoC interfaces
by Pierre-Louis Bossart 14 Nov '19

14 Nov '19

We need new fields in existing structures to a) deal with race conditions on codec probe/enumeration b) allow for multi-step ACPI scan/probe/startup on Intel plaforms To avoid conflicts between ASoC and Soundwire trees, these 4 patches are provided out-of-order, before the functionality enabled in these header files is added in follow-up patch series which can be applied separately in the ASoC and Soundwire trees (of course after Vinod and Mark sync-up so that these patches are present in both trees). Pierre-Louis Bossart (3): soundwire: sdw_slave: add new fields to track probe status soundwire: add enumeration_complete structure soundwire: intel: update interfaces between ASoC and SoundWire Rander Wang (1): soundwire: intel: update stream callbacks for hwparams/free stream operations drivers/soundwire/intel.c | 20 ++++-- drivers/soundwire/intel.h | 13 ++-- drivers/soundwire/intel_init.c | 31 +++------ include/linux/soundwire/sdw.h | 7 ++ include/linux/soundwire/sdw_intel.h | 103 +++++++++++++++++++++++++--- 5 files changed, 131 insertions(+), 43 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH v2] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint
by Yu-Hsuan Hsu 11 Nov '19

11 Nov '19

On KBL platform, the microphone is attached to external codec(rt5514) instead of PCH. However, TDM slot between PCH and codec is 16 bits only. In order to avoid setting wrong format, we should add a constraint to force to use 16 bits format forever. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- I have updated the commit message. Please see whether it is clear enough. Thanks. sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a01..67b276a65a8d2d 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -400,6 +400,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -- 2.23.0.351.gc4317032e6-goog

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[alsa-devel] [PATCH v3 00/11] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 11 Nov '19

11 Nov '19

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://patchwork.kernel.org/cover/11185769/ This patchset has been tested on SDM845 based DragonBoard DB845c and Lenovo Yoga C630 laptop with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. Pin Controller patch does not have any link dependency, it can go by its own. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. If anyone is interested to try, here are working set of patches on top of rc3. https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=audio/v5… alsa ucm files: https://git.linaro.org/people/srinivas.kandagatla/alsa-lib.git/log/?h=DB845c Thanks, srini Changes since v2: - Updated mfd driver as suggested by Lee. - Updated bindings as suggested by Rob. - Addressed various comments by Cezary Rojewski - Cleaned up code a bit. Srinivas Kandagatla (10): ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec mfd: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings dt-bindings: pinctrl: qcom-wcd934x: Add bindings for gpio ASoC: qcom: dt-bindings: Add compatible for DB845c and Lenovo Yoga ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga Yeleswarapu Nagaradhesh (1): pinctrl: qcom-wcd934x: Add support to wcd934x pinctrl driver. .../pinctrl/qcom,wcd934x-pinctrl.yaml | 52 + .../devicetree/bindings/sound/qcom,sdm845.txt | 5 +- .../bindings/sound/qcom,wcd934x.yaml | 162 + drivers/mfd/Kconfig | 12 + drivers/mfd/Makefile | 1 + drivers/mfd/wcd934x.c | 306 + drivers/pinctrl/qcom/Kconfig | 7 + drivers/pinctrl/qcom/Makefile | 1 + drivers/pinctrl/qcom/pinctrl-wcd934x-gpio.c | 365 ++ include/linux/mfd/wcd934x/registers.h | 529 ++ include/linux/mfd/wcd934x/wcd934x.h | 31 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5084 +++++++++++++++++ sound/soc/qcom/sdm845.c | 86 +- 15 files changed, 6651 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,wcd934x-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml create mode 100644 drivers/mfd/wcd934x.c create mode 100644 drivers/pinctrl/qcom/pinctrl-wcd934x-gpio.c create mode 100644 include/linux/mfd/wcd934x/registers.h create mode 100644 include/linux/mfd/wcd934x/wcd934x.h create mode 100644 sound/soc/codecs/wcd934x.c -- 2.21.0

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[alsa-devel] KMSAN: uninit-value in get_term_name
by syzbot 11 Nov '19

11 Nov '19

Hello, syzbot found the following crash on: HEAD commit: d86c1556 kmsan: add printk_test() git tree: https://github.com/google/kmsan.git master console output: https://syzkaller.appspot.com/x/log.txt?x=16c4ef54e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=c07a3d4f8a59e198 dashboard link: https://syzkaller.appspot.com/bug?extid=8f2612936028bfd28f28 compiler: clang version 9.0.0 (/home/glider/llvm/clang 80fee25776c2fb61e74c1ecb1a523375c2500b69) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=11b96c4ce00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=11a51ca8e00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+8f2612936028bfd28f28(a)syzkaller.appspotmail.com usb 1-1: New USB device found, idVendor=1d6b, idProduct=0101, bcdDevice= 0.40 usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ===================================================== BUG: KMSAN: uninit-value in get_term_name+0x1b7/0xad0 sound/usb/mixer.c:652 CPU: 1 PID: 30 Comm: kworker/1:1 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x191/0x1f0 lib/dump_stack.c:113 kmsan_report+0x128/0x220 mm/kmsan/kmsan_report.c:108 __msan_warning+0x73/0xe0 mm/kmsan/kmsan_instr.c:245 get_term_name+0x1b7/0xad0 sound/usb/mixer.c:652 __build_feature_ctl+0x11f1/0x29e0 sound/usb/mixer.c:1628 build_feature_ctl sound/usb/mixer.c:1709 [inline] parse_audio_feature_unit sound/usb/mixer.c:1921 [inline] parse_audio_unit+0x2308/0x7490 sound/usb/mixer.c:2753 snd_usb_mixer_controls sound/usb/mixer.c:3095 [inline] snd_usb_create_mixer+0x1d7c/0x4070 sound/usb/mixer.c:3445 usb_audio_probe+0x286b/0x3eb0 sound/usb/card.c:653 usb_probe_interface+0xd19/0x1310 drivers/usb/core/driver.c:361 really_probe+0xd91/0x1f90 drivers/base/dd.c:552 driver_probe_device+0x1ba/0x510 drivers/base/dd.c:721 __device_attach_driver+0x5b8/0x790 drivers/base/dd.c:828 bus_for_each_drv+0x28e/0x3b0 drivers/base/bus.c:430 __device_attach+0x489/0x750 drivers/base/dd.c:894 device_initial_probe+0x4a/0x60 drivers/base/dd.c:941 bus_probe_device+0x131/0x390 drivers/base/bus.c:490 device_add+0x25b5/0x2df0 drivers/base/core.c:2201 usb_set_configuration+0x309f/0x3710 drivers/usb/core/message.c:2027 generic_probe+0xe7/0x280 drivers/usb/core/generic.c:210 usb_probe_device+0x146/0x200 drivers/usb/core/driver.c:266 really_probe+0xd91/0x1f90 drivers/base/dd.c:552 driver_probe_device+0x1ba/0x510 drivers/base/dd.c:721 __device_attach_driver+0x5b8/0x790 drivers/base/dd.c:828 bus_for_each_drv+0x28e/0x3b0 drivers/base/bus.c:430 __device_attach+0x489/0x750 drivers/base/dd.c:894 device_initial_probe+0x4a/0x60 drivers/base/dd.c:941 bus_probe_device+0x131/0x390 drivers/base/bus.c:490 device_add+0x25b5/0x2df0 drivers/base/core.c:2201 usb_new_device+0x23e5/0x2fb0 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5098 [inline] hub_port_connect_change drivers/usb/core/hub.c:5213 [inline] port_event drivers/usb/core/hub.c:5359 [inline] hub_event+0x581d/0x72f0 drivers/usb/core/hub.c:5441 process_one_work+0x1572/0x1ef0 kernel/workqueue.c:2269 worker_thread+0x111b/0x2460 kernel/workqueue.c:2415 kthread+0x4b5/0x4f0 kernel/kthread.c:256 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:355 Uninit was stored to memory at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:151 [inline] kmsan_internal_chain_origin+0xbd/0x180 mm/kmsan/kmsan.c:319 __msan_chain_origin+0x6b/0xd0 mm/kmsan/kmsan_instr.c:179 parse_term_proc_unit+0x73d/0x7e0 sound/usb/mixer.c:896 __check_input_term+0x13ef/0x2360 sound/usb/mixer.c:989 check_input_term sound/usb/mixer.c:1008 [inline] parse_audio_feature_unit sound/usb/mixer.c:1875 [inline] parse_audio_unit+0x1478/0x7490 sound/usb/mixer.c:2753 snd_usb_mixer_controls sound/usb/mixer.c:3095 [inline] snd_usb_create_mixer+0x1d7c/0x4070 sound/usb/mixer.c:3445 usb_audio_probe+0x286b/0x3eb0 sound/usb/card.c:653 usb_probe_interface+0xd19/0x1310 drivers/usb/core/driver.c:361 really_probe+0xd91/0x1f90 drivers/base/dd.c:552 driver_probe_device+0x1ba/0x510 drivers/base/dd.c:721 __device_attach_driver+0x5b8/0x790 drivers/base/dd.c:828 bus_for_each_drv+0x28e/0x3b0 drivers/base/bus.c:430 __device_attach+0x489/0x750 drivers/base/dd.c:894 device_initial_probe+0x4a/0x60 drivers/base/dd.c:941 bus_probe_device+0x131/0x390 drivers/base/bus.c:490 device_add+0x25b5/0x2df0 drivers/base/core.c:2201 usb_set_configuration+0x309f/0x3710 drivers/usb/core/message.c:2027 generic_probe+0xe7/0x280 drivers/usb/core/generic.c:210 usb_probe_device+0x146/0x200 drivers/usb/core/driver.c:266 really_probe+0xd91/0x1f90 drivers/base/dd.c:552 driver_probe_device+0x1ba/0x510 drivers/base/dd.c:721 __device_attach_driver+0x5b8/0x790 drivers/base/dd.c:828 bus_for_each_drv+0x28e/0x3b0 drivers/base/bus.c:430 __device_attach+0x489/0x750 drivers/base/dd.c:894 device_initial_probe+0x4a/0x60 drivers/base/dd.c:941 bus_probe_device+0x131/0x390 drivers/base/bus.c:490 device_add+0x25b5/0x2df0 drivers/base/core.c:2201 usb_new_device+0x23e5/0x2fb0 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5098 [inline] hub_port_connect_change drivers/usb/core/hub.c:5213 [inline] port_event drivers/usb/core/hub.c:5359 [inline] hub_event+0x581d/0x72f0 drivers/usb/core/hub.c:5441 process_one_work+0x1572/0x1ef0 kernel/workqueue.c:2269 worker_thread+0x111b/0x2460 kernel/workqueue.c:2415 kthread+0x4b5/0x4f0 kernel/kthread.c:256 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:355 Uninit was created at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:151 [inline] kmsan_internal_poison_shadow+0x60/0x120 mm/kmsan/kmsan.c:134 kmsan_slab_alloc+0xaa/0x120 mm/kmsan/kmsan_hooks.c:88 slab_alloc_node mm/slub.c:2792 [inline] slab_alloc mm/slub.c:2801 [inline] __kmalloc+0x28e/0x430 mm/slub.c:3832 kmalloc include/linux/slab.h:561 [inline] usb_get_configuration+0x50d/0x76a0 drivers/usb/core/config.c:857 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0x224/0x2fb0 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5098 [inline] hub_port_connect_change drivers/usb/core/hub.c:5213 [inline] port_event drivers/usb/core/hub.c:5359 [inline] hub_event+0x581d/0x72f0 drivers/usb/core/hub.c:5441 process_one_work+0x1572/0x1ef0 kernel/workqueue.c:2269 worker_thread+0x111b/0x2460 kernel/workqueue.c:2415 kthread+0x4b5/0x4f0 kernel/kthread.c:256 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:355 ===================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

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[alsa-devel] [ANNOUNCE] Schedule for 2019 Linux Audio miniconference
by Mark Brown 11 Nov '19

11 Nov '19

Hi, As in previous years we're going to have an audio miniconference so we can get together and talk through issues, especially design decisions, face to face. This year's event will be held on October 31st in Lyon, France, the day after ELC-E. This will be held at the Lyon Convention Center (the ELC-E venue) and will be free of charge to attend, generously sponsored by Intel. The plan is to gather at 9am and start once enough people are there and ready. I look forward to seeing you all tomorrow! The agenda is looking pretty full now, what we've got so far is: - Introductions (me) - Use case management enhancements (Curtis) - DSP framework integration (Liam) - SOF integration with ACPI and DT (Daniel) - SOF support for non-modular drivers (Daniel) - Gapless compressed playback (Patrick) - Soundwire status (Liam?) - Multi-CPU support - Mixing with HDA/I2S in a single system - PCM issues (Patrick) - PCM device volume control - Attaching timestamps to buffers - kcontrol issues (Takashi) - Standardization - Confirmance testing - Virtualization (Liam) - User experiences (Jerome) - Unified graph (Patrick) - Componentisation status/plans (Me) Thanks again to Intel for supporting this event. Thanks, Mark

2 2

[alsa-devel] [PATCH 0/3] soundwire: use UniqueID only when relevant
by Pierre-Louis Bossart 09 Nov '19

09 Nov '19

The hardware UniqueID, typically enabled with pin-strapping, is required during enumeration to avoid conflicts between devices of the same type. When there are no devices of the same type, using the UniqueID is overkill and results in a lot of probe errors due to mismatches between ACPI tables and hardware capabilities. For example it's not uncommon for BIOS vendors to copy/paste the same settings between platforms but the hardware pin-strapping is different. This is perfectly legit and permitted by MIPI specs. With this patchset, the UniqueID is only used when multiple devices of the same type are detected. The loop to detect multiple identical devices is not super efficient but with typically fewer than 4 devices per link there's no real incentive to be smarter. This change is only implemented for ACPI platforms, for DeviceTree there is no change. Pierre-Louis Bossart (3): soundwire: remove bitfield for unique_id, use u8 soundwire: slave: add helper to extract slave ID soundwire: ignore uniqueID when irrelevant drivers/soundwire/bus.c | 7 +-- drivers/soundwire/slave.c | 98 +++++++++++++++++++++++++++-------- include/linux/soundwire/sdw.h | 4 +- 3 files changed, 84 insertions(+), 25 deletions(-) -- 2.20.1

2 9

[alsa-devel] [PATCH] soundwire: intel: fix PDI/stream mapping for Bulk
by Pierre-Louis Bossart 09 Nov '19

09 Nov '19

The previous formula is incorrect for PDI0/1, the mapping is not linear but has a discontinuity between PDI1 and PDI2. This change has no effect on PCM PDIs (same mapping). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/intel.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index b403ccc832b6..c984261fcc33 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -480,7 +480,10 @@ intel_pdi_shim_configure(struct sdw_intel *sdw, struct sdw_cdns_pdi *pdi) unsigned int link_id = sdw->instance; int pdi_conf = 0; - pdi->intel_alh_id = (link_id * 16) + pdi->num + 5; + /* the Bulk and PCM streams are not contiguous */ + pdi->intel_alh_id = (link_id * 16) + pdi->num + 3; + if (pdi->num >= 2) + pdi->intel_alh_id += 2; /* * Program stream parameters to stream SHIM register @@ -509,7 +512,10 @@ intel_pdi_alh_configure(struct sdw_intel *sdw, struct sdw_cdns_pdi *pdi) unsigned int link_id = sdw->instance; unsigned int conf; - pdi->intel_alh_id = (link_id * 16) + pdi->num + 5; + /* the Bulk and PCM streams are not contiguous */ + pdi->intel_alh_id = (link_id * 16) + pdi->num + 3; + if (pdi->num >= 2) + pdi->intel_alh_id += 2; /* Program Stream config ALH register */ conf = intel_readl(alh, SDW_ALH_STRMZCFG(pdi->intel_alh_id)); -- 2.20.1

2 3

[alsa-devel] [PATCH 00/14] soundwire: intel: implement new ASoC interfaces
by Pierre-Louis Bossart 08 Nov '19

08 Nov '19

this patchset applies on top of the series "[PATCH 0/4] soundwire: update ASoC interfaces" and previously submitted cleanups "[PATCH v3 0/5] soundwire: intel/cadence: better initialization" The changes are essentially a removal of the platform devices and the implementation of the new interfaces required to scan the ACPI tables, probe the links and start them. The missing prepare, trigger and setup ASoC callbacks are also implemented. The hw_params and free callbacks use the new interfaces as well. While there are quite a few lines of code changed, this is mostly about interface changes. The next series will contain more functional changes and deal with race conditions on probe, enumeration and suspend/resume issues. Bard Liao (1): soundwire: add device driver to sdw_md_driver Pierre-Louis Bossart (10): soundwire: renames to prepare support for master drivers/devices soundwire: rename dev_to_sdw_dev macro soundwire: rename drv_to_sdw_slave_driver macro soundwire: bus_type: rename sdw_drv_ to sdw_slave_drv soundwire: intel: rename res field as link_res soundwire: add support for sdw_slave_type soundwire: add initial definitions for sdw_master_device soundwire: intel: remove platform devices and provide new interface soundwire: intel: free all resources on hw_free() soundwire: intel_init: add implementation of sdw_intel_enable_irq() Rander Wang (3): soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback drivers/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus_type.c | 60 +++--- drivers/soundwire/intel.c | 280 ++++++++++++++++++++++----- drivers/soundwire/intel.h | 8 +- drivers/soundwire/intel_init.c | 300 ++++++++++++++++++++++------- drivers/soundwire/master.c | 64 ++++++ drivers/soundwire/slave.c | 9 +- include/linux/soundwire/sdw.h | 39 +++- include/linux/soundwire/sdw_type.h | 34 +++- 11 files changed, 642 insertions(+), 160 deletions(-) create mode 100644 drivers/soundwire/master.c -- 2.20.1

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[alsa-devel] [PATCH] ASoC: fsi: switch to yaml base Documentation
by Kuninori Morimoto 07 Nov '19

07 Nov '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document for FSI. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,fsi.txt | 31 --------- .../devicetree/bindings/sound/renesas,fsi.yaml | 76 ++++++++++++++++++++++ 2 files changed, 76 insertions(+), 31 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/renesas,fsi.txt create mode 100644 Documentation/devicetree/bindings/sound/renesas,fsi.yaml diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.txt b/Documentation/devicetree/bindings/sound/renesas,fsi.txt deleted file mode 100644 index 0cf0f81..0000000 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.txt +++ /dev/null @@ -1,31 +0,0 @@ -Renesas FSI - -Required properties: -- compatible : "renesas,fsi2-<soctype>", - "renesas,sh_fsi2" or "renesas,sh_fsi" as - fallback. - Examples with soctypes are: - - "renesas,fsi2-r8a7740" (R-Mobile A1) - - "renesas,fsi2-sh73a0" (SH-Mobile AG5) -- reg : Should contain the register physical address and length -- interrupts : Should contain FSI interrupt - -- fsia,spdif-connection : FSI is connected by S/PDIF -- fsia,stream-mode-support : FSI supports 16bit stream mode. -- fsia,use-internal-clock : FSI uses internal clock when master mode. - -- fsib,spdif-connection : same as fsia -- fsib,stream-mode-support : same as fsia -- fsib,use-internal-clock : same as fsia - -Example: - -sh_fsi2: sh_fsi2@ec230000 { - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupts = <0 146 0x4>; - - fsia,spdif-connection; - fsia,stream-mode-support; - fsia,use-internal-clock; -}; diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml new file mode 100644 index 0000000..140a37f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,fsi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas FSI Sound Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for FSI2 SoC + - items: + - enum: + - renesas,fsi2-sh73a0 + - renesas,fsi2-r8a7740 + - enum: + - renesas,sh_fsi2 + # for Generic + - items: + - enum: + - renesas,sh_fsi + - renesas,sh_fsi2 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + fsia,spdif-connection: + $ref: /schemas/types.yaml#/definitions/flag + description: FSI is connected by S/PDIF + + fsia,stream-mode-support: + $ref: /schemas/types.yaml#/definitions/flag + description: FSI supports 16bit stream mode + + fsia,use-internal-clock: + $ref: /schemas/types.yaml#/definitions/flag + description: FSI uses internal clock when master mode + + fsib,spdif-connection: + $ref: /schemas/types.yaml#/definitions/flag + description: same as fsia + + fsib,stream-mode-support: + $ref: /schemas/types.yaml#/definitions/flag + description: same as fsia + + fsib,use-internal-clock: + $ref: /schemas/types.yaml#/definitions/flag + description: same as fsia + +required: + - compatible + - reg + - interrupts + +examples: + - | + sh_fsi2: sound@ec230000 { + compatible = "renesas,fsi2-r8a7740", "renesas,sh_fsi2"; + reg = <0xec230000 0x400>; + interrupts = <0 146 0x4>; + + fsia,spdif-connection; + fsia,stream-mode-support; + fsia,use-internal-clock; + }; -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: wm_adsp: Expose mixer control API
by David Rhodes 07 Nov '19

07 Nov '19

From: Li Xu <li.xu(a)cirrus.com> Expose mixer control API for reading and writing. The exposed API can be used by codec driver for interacting with mixer control in kernel space. This allows codec driver to implement more involved interactions with DSP firmware, such as Fast Use Case Switching. Signed-off-by: Li Xu <li.xu(a)cirrus.com> Signed-off-by: David Rhodes <david.rhodes(a)cirrus.com> --- sound/soc/codecs/wm_adsp.c | 70 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 4 +++ 2 files changed, 73 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index aa74fd1..77b8aea 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -635,6 +635,9 @@ struct wm_coeff_ctl_ops { struct wm_coeff_ctl { const char *name; const char *fw_name; + /* Subname is needed to match with firmware */ + const char *subname; + unsigned int subname_len; struct wm_adsp_alg_region alg_region; struct wm_coeff_ctl_ops ops; struct wm_adsp *dsp; @@ -1520,6 +1523,7 @@ static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl) { kfree(ctl->cache); kfree(ctl->name); + kfree(ctl->subname); kfree(ctl); } @@ -1597,6 +1601,15 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ret = -ENOMEM; goto err_ctl; } + if (subname) { + ctl->subname_len = subname_len; + ctl->subname = kmemdup(subname, + strlen(subname) + 1, GFP_KERNEL); + if (!ctl->subname) { + ret = -ENOMEM; + goto err_ctl_name; + } + } ctl->enabled = 1; ctl->set = 0; ctl->ops.xget = wm_coeff_get; @@ -1610,7 +1623,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl->cache = kzalloc(ctl->len, GFP_KERNEL); if (!ctl->cache) { ret = -ENOMEM; - goto err_ctl_name; + goto err_ctl_subname; } list_add(&ctl->list, &dsp->ctl_list); @@ -1633,6 +1646,8 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, err_ctl_cache: kfree(ctl->cache); +err_ctl_subname: + kfree(ctl->subname); err_ctl_name: kfree(ctl->name); err_ctl: @@ -2194,6 +2209,59 @@ static int wm_adsp_load(struct wm_adsp *dsp) return ret; } +/* + * Find wm_coeff_ctl with input name as its subname + * If not found, return NULL + */ +static struct wm_coeff_ctl *wm_adsp_get_ctl(struct wm_adsp *dsp, + const char *name) +{ + struct wm_coeff_ctl *pos, *rslt = NULL; + + list_for_each_entry(pos, &dsp->ctl_list, list) { + if (!pos->subname) + continue; + if (strncmp(pos->subname, name, pos->subname_len) == 0) { + rslt = pos; + break; + } + } + + return rslt; +} + +int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, const void *buf, + size_t len) +{ + struct wm_coeff_ctl *ctl; + + ctl = wm_adsp_get_ctl(dsp, name); + if (!ctl) + return -EINVAL; + + if (len > ctl->len) + return -EINVAL; + + return wm_coeff_write_control(ctl, buf, len); +} +EXPORT_SYMBOL_GPL(wm_adsp_write_ctl); + +int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, void *buf, + size_t len) +{ + struct wm_coeff_ctl *ctl; + + ctl = wm_adsp_get_ctl(dsp, name); + if (!ctl) + return -EINVAL; + + if (len > ctl->len) + return -EINVAL; + + return wm_coeff_read_control(ctl, buf, len); +} +EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); + static void wm_adsp_ctl_fixup_base(struct wm_adsp *dsp, const struct wm_adsp_alg_region *alg_region) { diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 5275a49..2a8cd09 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -223,5 +223,9 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); int wm_adsp_compr_copy(struct snd_compr_stream *stream, char __user *buf, size_t count); +int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, const void *buf, + size_t len); +int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, void *buf, + size_t len); #endif -- 1.9.1

4 3

[alsa-devel] [PATCH 0/5] ASoC: SOF: Intel: Soundwire integration
by Pierre-Louis Bossart 06 Nov '19

06 Nov '19

This patchset applies on top of the series "[PATCH 0/4] soundwire: update ASoC interfaces". The SOF/Intel code makes use of the interfaces defined for initialization. Build support for SoundWire is not provided for now, all Soundwire-related code will be handled with a dummy fallback. We will enable SoundWire interfaces in the Kconfigs when the functionality is enabled in the soundwire tree. In short, if the interfaces are agreed on, there is no risk with the integration of these patches on the ASoC side. Pierre-Louis Bossart (5): ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend include/sound/sof/dai-intel.h | 18 +-- sound/soc/sof/intel/hda-dsp.c | 2 + sound/soc/sof/intel/hda-loader.c | 13 ++ sound/soc/sof/intel/hda.c | 230 ++++++++++++++++++++++++++++++- sound/soc/sof/intel/hda.h | 44 ++++++ 5 files changed, 295 insertions(+), 12 deletions(-) -- 2.20.1

1 6

[alsa-devel] [PATCH v2 00/11] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 06 Nov '19

06 Nov '19

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://patchwork.kernel.org/cover/11185769/ This patchset has been tested on SDM845 based DragonBoard DB845c and Lenovo Yoga C630 laptop with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. Pin Controller patch does not have any link dependency, it can go by its own. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. If anyone is interested to try, here are working set of patches on top of rc3. https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=audio/v5… alsa ucm files: https://git.linaro.org/people/srinivas.kandagatla/alsa-lib.git/log/?h=DB845c Thanks, srini Changes since v1: - Code cleanup and convert to proper dpcm widgets where possible. - converted to mfd driver. - added pinctrl driver to this series - added dts changes in this series - bindings converted to yaml. Srinivas Kandagatla (10): ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec mfd: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings dt-bindings: pinctrl: qcom-wcd934x: Add bindings for gpio ASoC: dt-bindings: Add compatible for DB845c and Lenovo Yoga ASoC: qcom: sdm845: add support to DB845c and Lenovo Yoga Yeleswarapu Nagaradhesh (1): pinctrl: qcom-wcd934x: Add support to wcd934x pinctrl driver. .../pinctrl/qcom,wcd934x-pinctrl.yaml | 51 + .../devicetree/bindings/sound/qcom,sdm845.txt | 5 +- .../bindings/sound/qcom,wcd934x.yaml | 169 + drivers/mfd/Kconfig | 8 + drivers/mfd/Makefile | 1 + drivers/mfd/wcd934x.c | 330 ++ drivers/pinctrl/qcom/Kconfig | 7 + drivers/pinctrl/qcom/Makefile | 1 + drivers/pinctrl/qcom/pinctrl-wcd934x-gpio.c | 365 ++ include/linux/mfd/wcd934x/registers.h | 529 ++ include/linux/mfd/wcd934x/wcd934x.h | 24 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5218 +++++++++++++++++ sound/soc/qcom/sdm845.c | 71 +- 15 files changed, 6789 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,wcd934x-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml create mode 100644 drivers/mfd/wcd934x.c create mode 100644 drivers/pinctrl/qcom/pinctrl-wcd934x-gpio.c create mode 100644 include/linux/mfd/wcd934x/registers.h create mode 100644 include/linux/mfd/wcd934x/wcd934x.h create mode 100644 sound/soc/codecs/wcd934x.c -- 2.21.0

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[alsa-devel] [PATCH 1/7] ASoC: amd: Create multiple I2S platform device endpoints.
by Ravulapati Vishnu vardhan rao 06 Nov '19

06 Nov '19

Creates Platform Device endpoints for multiple I2S instances: SP and BT endpoints device. Pass PCI resources like MMIO, irq to the platform devices. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 5 +++ sound/soc/amd/raven/pci-acp3x.c | 83 +++++++++++++++++++++++++++-------------- 2 files changed, 60 insertions(+), 28 deletions(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 4f2cadd..2f15fe1 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -7,10 +7,15 @@ #include "chip_offset_byte.h" +#define ACP3x_DEVS 3 #define ACP3x_PHY_BASE_ADDRESS 0x1240000 #define ACP3x_I2S_MODE 0 #define ACP3x_REG_START 0x1240000 #define ACP3x_REG_END 0x1250200 +#define ACP3x_I2STDM_REG_START 0x1242400 +#define ACP3x_I2STDM_REG_END 0x1242410 +#define ACP3x_BT_TDM_REG_START 0x1242800 +#define ACP3x_BT_TDM_REG_END 0x1242810 #define I2S_MODE 0x04 #define BT_TX_THRESHOLD 26 #define BT_RX_THRESHOLD 25 diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index facec24..7f435b3 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -16,16 +16,16 @@ struct acp3x_dev_data { void __iomem *acp3x_base; bool acp3x_audio_mode; struct resource *res; - struct platform_device *pdev; + struct platform_device *pdev[ACP3x_DEVS]; }; static int snd_acp3x_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { int ret; - u32 addr, val; + u32 addr, val, i; struct acp3x_dev_data *adata; - struct platform_device_info pdevinfo; + struct platform_device_info pdevinfo[ACP3x_DEVS]; unsigned int irqflags; if (pci_enable_device(pci)) { @@ -68,7 +68,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, switch (val) { case I2S_MODE: adata->res = devm_kzalloc(&pci->dev, - sizeof(struct resource) * 2, + sizeof(struct resource) * 4, GFP_KERNEL); if (!adata->res) { ret = -ENOMEM; @@ -80,39 +80,61 @@ static int snd_acp3x_probe(struct pci_dev *pci, adata->res[0].start = addr; adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); - adata->res[1].name = "acp3x_i2s_irq"; - adata->res[1].flags = IORESOURCE_IRQ; - adata->res[1].start = pci->irq; - adata->res[1].end = pci->irq; + adata->res[1].name = "acp3x_i2s_sp"; + adata->res[1].flags = IORESOURCE_MEM; + adata->res[1].start = addr + ACP3x_I2STDM_REG_START; + adata->res[1].end = addr + ACP3x_I2STDM_REG_END; + + adata->res[2].name = "acp3x_i2s_bt"; + adata->res[2].flags = IORESOURCE_MEM; + adata->res[2].start = addr + ACP3x_BT_TDM_REG_START; + adata->res[2].end = addr + ACP3x_BT_TDM_REG_END; + + adata->res[3].name = "acp3x_i2s_irq"; + adata->res[3].flags = IORESOURCE_IRQ; + adata->res[3].start = pci->irq; + adata->res[3].end = adata->res[3].start; adata->acp3x_audio_mode = ACP3x_I2S_MODE; memset(&pdevinfo, 0, sizeof(pdevinfo)); - pdevinfo.name = "acp3x_rv_i2s"; - pdevinfo.id = 0; - pdevinfo.parent = &pci->dev; - pdevinfo.num_res = 2; - pdevinfo.res = adata->res; - pdevinfo.data = &irqflags; - pdevinfo.size_data = sizeof(irqflags); - - adata->pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(adata->pdev)) { - dev_err(&pci->dev, "cannot register %s device\n", - pdevinfo.name); - ret = PTR_ERR(adata->pdev); - goto unmap_mmio; + pdevinfo[0].name = "acp3x_rv_i2s_dma"; + pdevinfo[0].id = 0; + pdevinfo[0].parent = &pci->dev; + pdevinfo[0].num_res = 4; + pdevinfo[0].res = &adata->res[0]; + pdevinfo[0].data = &irqflags; + pdevinfo[0].size_data = sizeof(irqflags); + + pdevinfo[1].name = "acp3x_i2s_playcap"; + pdevinfo[1].id = 0; + pdevinfo[1].parent = &pci->dev; + pdevinfo[1].num_res = 1; + pdevinfo[1].res = &adata->res[1]; + + pdevinfo[2].name = "acp3x_i2s_playcap"; + pdevinfo[2].id = 1; + pdevinfo[2].parent = &pci->dev; + pdevinfo[2].num_res = 1; + pdevinfo[2].res = &adata->res[2]; + for (i = 0; i < ACP3x_DEVS ; i++) { + adata->pdev[i] = + platform_device_register_full(&pdevinfo[i]); + if (adata->pdev[i] == NULL) { + dev_err(&pci->dev, "cannot register %s device\n", + pdevinfo[i].name); + ret = -ENODEV; + goto unmap_mmio; + } } break; - default: - dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); - ret = -ENODEV; - goto unmap_mmio; } return 0; unmap_mmio: - pci_disable_msi(pci); + for (i = 0 ; i < ACP3x_DEVS ; i++) + platform_device_unregister(adata->pdev[i]); + kfree(adata->res); iounmap(adata->acp3x_base); release_regions: pci_release_regions(pci); @@ -124,9 +146,13 @@ static int snd_acp3x_probe(struct pci_dev *pci, static void snd_acp3x_remove(struct pci_dev *pci) { + int i; struct acp3x_dev_data *adata = pci_get_drvdata(pci); - platform_device_unregister(adata->pdev); + if (adata->acp3x_audio_mode == ACP3x_I2S_MODE) { + for (i = 0 ; i < ACP3x_DEVS ; i++) + platform_device_unregister(adata->pdev[i]); + } iounmap(adata->acp3x_base); pci_disable_msi(pci); @@ -151,6 +177,7 @@ static struct pci_driver acp3x_driver = { module_pci_driver(acp3x_driver); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); MODULE_DESCRIPTION("AMD ACP3x PCI driver"); MODULE_LICENSE("GPL v2"); -- 2.7.4

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[alsa-devel] [PATCH V2 1/2] ASoC: dt-bindings: fsl_asrc: add compatible string for imx8qm
by Shengjiu Wang 06 Nov '19

06 Nov '19

In order to support the two asrc modules in imx8qm, we need to add compatible string "fsl,imx8qm-asrc0" and "fsl,imx8qm-asrc1" Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,asrc.txt | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index 1d4d9f938689..cd2bd3daa7e1 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -8,7 +8,8 @@ three substreams within totally 10 channels. Required properties: - - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + - compatible : Contains "fsl,imx35-asrc", "fsl,imx53-asrc", + "fsl,imx8qm-asrc0" or "fsl,imx8qm-asrc1". - reg : Offset and length of the register set for the device. -- 2.21.0

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[alsa-devel] [PATCH] ASoC: bdw-rt5677: enable runtime channel merge
by Brent Lu 05 Nov '19

05 Nov '19

In the DAI link "Capture PCM", the FE DAI "Capture Pin" supports 4-channel capture but the BE DAI supports only 2-channel capture. To fix the channel mismatch, we need to enable the runtime channel merge for this DAI link. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5677.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 4a4d335..8d07038 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -273,6 +273,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { }, .dpcm_capture = 1, .dpcm_playback = 1, + .dpcm_merged_chan = 1, SND_SOC_DAILINK_REG(fe, dummy, platform), }, -- 2.7.4

3 4

[alsa-devel] tegra30 tdm support
by Ben Dooks 05 Nov '19

05 Nov '19

New series for TDM audio support on tegra30 v2: - moved edge-control and data-offset to the set_fmt callbacks - fixed dev_dbg in set_tdm callback - fixed dev_dbg message contents in set_tdm callback - changed fsync width to be permanently 1 clock v3: - cleanup fsync patch - fix rebase issue with tdm patch v4: - fix comment style issues - change tdm-a to data-offset 1

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[alsa-devel] [PATCH v2 resend 00/18] ASoC: soc-core cleanup - step 4
by Kuninori Morimoto 05 Nov '19

05 Nov '19

Hi Mark These are resend of soc-core cleanup step4. These are based on mark/for-5.5 branch. I already got report from Intel that these had no issue on Intel CI machine. Kuninori Morimoto (18): ASoC: soc-core: remove unneeded snd_soc_tplg_component_remove() ASoC: soc-core: move soc_init_dai_link() ASoC: soc-core: tidyup soc_init_dai_link() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: call soc_bind_dai_link() under snd_soc_add_dai_link() ASoC: soc-core: add soc_unbind_dai_link() ASoC: soc-core: move snd_soc_lookup_component() ASoC: soc-core: add snd_soc_del_component_unlocked() ASoC: soc-core: remove snd_soc_component_add/del() ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() ASoC: soc-core: move snd_soc_register_dai() ASoC: soc-core: move snd_soc_unregister_dais() ASoC: soc-core: add snd_soc_unregister_dai() ASoC: soc-core: have legacy_dai_naming at snd_soc_register_dai() ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() ASoC: soc-core: call snd_soc_register_dai() from snd_soc_register_dais() ASoC: soc-core: remove topology specific operation ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY include/sound/soc.h | 15 +- sound/soc/soc-core.c | 536 +++++++++++++++++++++-------------------------- sound/soc/soc-topology.c | 17 +- 3 files changed, 263 insertions(+), 305 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v4 0/2] soundwire: Add support to Qualcomm SoundWire master
by Srinivas Kandagatla 05 Nov '19

05 Nov '19

Thanks for reviewing the v3 patchset. Here is new patchset addressing all the comments from v3 This patchset adds support for Qualcomm SoundWire Master Controller found in most of Qualcomm SoCs and WCD audio codecs. This driver along with WCD934x codec and WSA881x Class-D Smart Speaker Amplifier drivers is tested on on DragonBoard DB845c based of SDM845 SoC and Lenovo YOGA C630 Laptop based on SDM850. SoundWire controller on SDM845 is integrated in WCD934x audio codec via SlimBus interface. Currently this driver is very minimal and only supports PDM. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Test and add PCM support. Thanks, srini Changes since v3: - Updated bindings as suggested by Rob. - Fixed sdw_disable/unprepare order. - removed setting stream_name as suggested by Pierre - removed v1.5.0 controller support for now, will be added after testing. Srinivas Kandagatla (2): dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller .../bindings/soundwire/qcom,sdw.txt | 179 ++++ drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/qcom.c | 904 ++++++++++++++++++ 4 files changed, 1096 insertions(+) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c -- 2.21.0

3 8

[alsa-devel] [PATCH] dt-bindings: sound: adau7118: Fix example warning
by Maxime Ripard 04 Nov '19

04 Nov '19

The ADAU7118 has an example where the codec has an i2c address of 14, and the unit address set to 14 as well. However, while the address is expressed in decimal, the unit-address is supposed to be in hexadecimal, which ends up with two different addresses that trigger a DTC warning. Fix this by setting the unit address to (0x)e. Cc: Nuno Sá <nuno.sa(a)analog.com> Fixes: 969d49b2cdc8 ("dt-bindings: asoc: Add ADAU7118 documentation") Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- Documentation/devicetree/bindings/sound/adi,adau7118.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml index c3f10afbdd6f..65f6844a0c6d 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -65,7 +65,7 @@ examples: /* example with i2c support */ #address-cells = <1>; #size-cells = <0>; - adau7118_codec: audio-codec@14 { + adau7118_codec: audio-codec@e { compatible = "adi,adau7118"; reg = <14>; #sound-dai-cells = <0>; -- 2.23.0

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