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

[PATCH v3 00/10] ASoC: Clean-up W=1 build warnings - part2
by Pierre-Louis Bossart 10 Jul '20

10 Jul '20

Both Lee Jones and I submitted separate series, this is the second part of the merged result, for which no feedback was provided. I picked Lee's patches for rt5659 and ak4458 and added the pxa and ux500 that I didn't fix. The rest is largely identical between our respective series, with the exception of the sunxi which I documented and Lee removed. I don't have any specific preference and will go with the flow on this. Lee Jones (4): ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and 'reg_val_TSTDR' ASoC: codecs: rt5659: Remove many unused const variables ASoC: codecs: ak4458: Remove set but never checked variable 'ret' Pierre-Louis Bossart (6): ASoC: qcom: q6asm: fix kernel-doc ASoC: sunxi: sun4i-i2s: fix kernel-doc ASoC: sunxi: sun4i-spdif: fix kernel-doc ASoC: sti: uniperif: fix 'defined by not used' warning ASoC: codecs: rt5631: fix kernel-doc ASoC: codecs: tlv320aic26: fix kernel-doc warning sound/soc/codecs/ak4458.c | 6 +++--- sound/soc/codecs/rt5631.c | 8 +++++-- sound/soc/codecs/rt5659.c | 37 --------------------------------- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sunxi/sun4i-i2s.c | 10 ++++++++- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 8 +++---- 10 files changed, 27 insertions(+), 52 deletions(-) -- 2.25.1

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Weird jack detection on ASRock B450M Pro4
by Arun Raghavan 10 Jul '20

10 Jul '20

Hi folks, I ran into some weird behaviour on this motherboard. First, alsa-info is here: http://alsa-project.org/db/?f=f48105f391d848b34b3779f27369130ca024f417 The weird behaviour is that the HDA card is picked up fine, but jack detection is broken on boot. However, this gets "fixed" after you run alsa-info. The before/after codec info is at: https://arunraghavan.net/temp/alsa-codecs-before.txt https://arunraghavan.net/temp/alsa-codecs-after.txt The key delta is: ---- @@ -133,24 +133,24 @@ Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC892 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 - Amp-Out vals: [0x34 0x34] - Converter: stream=0, channel=0 + Amp-Out vals: [0x29 0x29] + Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 - Amp-Out vals: [0x34 0x34] - Converter: stream=0, channel=0 + Amp-Out vals: [0x29 0x29] + Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS @@ -159,12 +159,12 @@ Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 - Amp-Out vals: [0x34 0x34] - Converter: stream=0, channel=0 + Amp-Out vals: [0x29 0x29] + Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS @@ -172,11 +172,11 @@ Node 0x05 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] - Converter: stream=0, channel=0 + Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS ---- Any thoughts on what's changing that, and how this should be fixed? Cheers, Arun

1 1

ASoC: rt5670/cht_bsw_rt5672: Some bugfixes and cleanups
by Hans de Goede 10 Jul '20

10 Jul '20

Hi All, This series is mainly the result of me working on fixing the extra set of speakers in the Lenovo Miix 2 10 kbd dock not working with Linux (patches 1 and 3) while working on this I noticed some other bugs (patches 2 and 4) and I also ended up doing some generic cleanups (patches 5 and 6). For patches 1-3 or 1-4 it would be nice it they can go to a future 5.8-rc# 5-6 are mostly -next material. Regards, Hans

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Re: KASAN: use-after-free Read in line6_submit_audio_in_all_urbs
by Alan Stern 10 Jul '20

10 Jul '20

On Fri, Jun 26, 2020 at 10:18:12AM -0700, syzbot wrote: > Hello, > > syzbot found the following crash on: > > HEAD commit: fb574682 usbip: tools: fix module name in man page > git tree: https://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb.git usb-testing > console output: https://syzkaller.appspot.com/x/log.txt?x=156560b1100000 > kernel config: https://syzkaller.appspot.com/x/.config?x=63b40b2ae167bad6 > dashboard link: https://syzkaller.appspot.com/bug?extid=145012a46658ac00fc9e > compiler: gcc (GCC) 10.1.0-syz 20200507 > > Unfortunately, I don't have any reproducer for this crash yet. > > IMPORTANT: if you fix the bug, please add the following tag to the commit: > Reported-by: syzbot+145012a46658ac00fc9e(a)syzkaller.appspotmail.com > > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > ================================================================== > BUG: KASAN: use-after-free in line6_submit_audio_in_all_urbs+0x10b/0x120 sound/usb/line6/capture.c:72 > Read of size 8 at addr ffff8881cffb1800 by task kworker/1:5/3257 > > CPU: 1 PID: 3257 Comm: kworker/1:5 Not tainted 5.8.0-rc1-syzkaller #0 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 > Workqueue: events line6_startup_work > Call Trace: > __dump_stack lib/dump_stack.c:77 [inline] > dump_stack+0xf6/0x16e lib/dump_stack.c:118 > print_address_description.constprop.0+0x1a/0x210 mm/kasan/report.c:383 > __kasan_report mm/kasan/report.c:513 [inline] > kasan_report.cold+0x37/0x7c mm/kasan/report.c:530 > line6_submit_audio_in_all_urbs+0x10b/0x120 sound/usb/line6/capture.c:72 > line6_stream_start+0x207/0x230 sound/usb/line6/pcm.c:197 > line6_pcm_acquire+0x161/0x210 sound/usb/line6/pcm.c:318 > line6_startup_work+0x42/0x50 sound/usb/line6/driver.c:734 > process_one_work+0x94c/0x15f0 kernel/workqueue.c:2269 > worker_thread+0x64c/0x1120 kernel/workqueue.c:2415 > kthread+0x392/0x470 kernel/kthread.c:291 > ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:293 > > Allocated by task 76: > save_stack+0x1b/0x40 mm/kasan/common.c:48 > set_track mm/kasan/common.c:56 [inline] > __kasan_kmalloc.constprop.0+0xc2/0xd0 mm/kasan/common.c:494 > kmalloc include/linux/slab.h:555 [inline] > kzalloc include/linux/slab.h:669 [inline] > line6_init_pcm+0x2a7/0x9e0 sound/usb/line6/pcm.c:533 > toneport_init+0xdd/0x6b0 sound/usb/line6/toneport.c:419 > line6_probe+0xaa0/0x1330 sound/usb/line6/driver.c:809 It look like the cancel_delayed_work() in line6_disconnect() needs to be cancel_delayed_work_sync(). Unfortunately we can't test this until syzbot is able to reproduce the bug. Alan Stern

2 1

Re: 回复: WARNING in submit_audio_out_urb/usb_submit_urb
by Takashi Iwai 10 Jul '20

10 Jul '20

On Fri, 10 Jul 2020 10:12:43 +0200, Zhang, Qiang wrote: > > > > ________________________________________ > 发件人: linux-kernel-owner(a)vger.kernel.org <linux-kernel-owner(a)vger.kernel.org> 代表 syzbot <syzbot+c190f6858a04ea7fbc52(a)syzkaller.appspotmail.com> > 发送时间: 2020年7月9日 21:34 > 收件人: andreyknvl(a)google.com; gregkh(a)linuxfoundation.org; ingrassia(a)epigenesys.com; linux-kernel(a)vger.kernel.org; linux-usb(a)vger.kernel.org; syzkaller-bugs(a)googlegroups.com > 主题: WARNING in submit_audio_out_urb/usb_submit_urb > > Hello, > > syzbot found the following crash on: > > HEAD commit: 768a0741 usb: dwc2: gadget: Remove assigned but never used.. > git tree: https://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb.git usb-testing > console output: https://syzkaller.appspot.com/x/log.txt?x=1568d11f100000 > kernel config: https://syzkaller.appspot.com/x/.config?x=999be4eb2478ffa5 > dashboard link: https://syzkaller.appspot.com/bug?extid=c190f6858a04ea7fbc52 > compiler: gcc (GCC) 10.1.0-syz 20200507 > syz repro: https://syzkaller.appspot.com/x/repro.syz?x=123aa2fb100000 > > IMPORTANT: if you fix the bug, please add the following tag to the commit: > Reported-by: syzbot+c190f6858a04ea7fbc52(a)syzkaller.appspotmail.com > > usb 1-1: send failed (error -32) > snd_usb_toneport 1-1:0.0: Line 6 TonePort GX now attached > ------------[ cut here ]------------ > usb 1-1: BOGUS urb xfer, pipe 0 != type 3 > WARNING: CPU: 0 PID: 12 at drivers/usb/core/urb.c:478 usb_submit_urb+0xa17/0x13e0 drivers/usb/core/urb.c:478 > Kernel panic - not syncing: panic_on_warn set ... > CPU: 0 PID: 12 Comm: kworker/0:1 Not tainted 5.8.0-rc3-syzkaller #0 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 > Workqueue: events line6_startup_work > Call Trace: > __dump_stack lib/dump_stack.c:77 [inline] > dump_stack+0xf6/0x16e lib/dump_stack.c:118 > panic+0x2aa/0x6e1 kernel/panic.c:231 > __warn.cold+0x20/0x50 kernel/panic.c:600 > report_bug+0x1bd/0x210 lib/bug.c:198 > handle_bug+0x41/0x80 arch/x86/kernel/traps.c:235 > exc_invalid_op+0x13/0x40 arch/x86/kernel/traps.c:255 > asm_exc_invalid_op+0x12/0x20 arch/x86/include/asm/idtentry.h:563 > RIP: 0010:usb_submit_urb+0xa17/0x13e0 drivers/usb/core/urb.c:478 > Code: 84 e7 04 00 00 e8 a9 10 ca fd 4c 89 ef e8 41 79 12 ff 41 89 d8 44 89 e1 4c 89 f2 48 89 c6 48 c7 c7 80 a0 5d 86 e8 db 77 9e fd <0f> 0b e8 82 10 ca fd 0f b6 6c 24 08 48 c7 c6 e0 a1 5d 86 48 89 ef > RSP: 0018:ffff8881da227b10 EFLAGS: 00010086 > RAX: 0000000000000000 RBX: 0000000000000003 RCX: 0000000000000000 > RDX: ffff8881da211900 RSI: ffffffff8129b4e3 RDI: ffffed103b444f54 > RBP: 0000000000000030 R08: 0000000000000001 R09: ffff8881db21fe8b > R10: 0000000000000000 R11: 0000000000000004 R12: 0000000000000000 > R13: ffff8881d6ecd0a0 R14: ffff8881d3d8c690 R15: ffff8881d54c4000 > submit_audio_out_urb+0x6d6/0x1a00 sound/usb/line6/playback.c:271 > line6_submit_audio_out_all_urbs+0xc9/0x120 sound/usb/line6/playback.c:291 > line6_stream_start+0x187/0x230 sound/usb/line6/pcm.c:195 > line6_pcm_acquire+0x137/0x210 sound/usb/line6/pcm.c:318 > line6_startup_work+0x42/0x50 sound/usb/line6/driver.c:734 > process_one_work+0x94c/0x15f0 kernel/workqueue.c:2269 > worker_thread+0x64c/0x1120 kernel/workqueue.c:2415 > kthread+0x392/0x470 kernel/kthread.c:291 > ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:293 > > It's like Alan Stern's reply to the email titled "KASAN: use-after-free Read in line6_submit_audio_in_all_urbs." It's also like a problem with asynchronous operations. can replace "cancel_delayed_work" with" cancel_delayed_work_sync" > in "line6_disconnect" func？ No, this looks a different problem. It's just a matter of the missing URB type check that leads to a USB warning. A typical issue with the USB fuzzing. I'll check it later. thanks, Takashi

1 1

Feature request: volume parameter for aplay
by GitHub issues - opened 10 Jul '20

10 Jul '20

alsa-project/alsa-utils issue #45 was opened from flatsiedatsie: I am currently in a situation where I'm trying to play a notification sound when a privacy friendly voice control system hears its wake word. Specifically, it's the [Voco addon](https://github.com/createcandle/voco) for the [Mozilla WebThings Gateway](https://webthingsgateway.com/), an open source smart home controller. Up till now I used `aplay` for this. But I need/want to support a ReSpeaker USB microphone, and that device doesn't have a simple volume control. Unfortunately addons are not allowed to make changes to asound.conf (which could add softvol) or install something like Pulse audio to mitigate this situation, as those changes could affect other addons of the system. My current solution has been to switch to a different available audio player, `ffplay`. It has a `-volume` parameter. Having a `-volume` option actually works out quite nicely, as it allows sounds to be played at relative volume to each other. For example, I can play a music stream at 40% volume, while setting the audio feedback from the voice control to be louder so it can always be heard over the music. The downside of ffplay is that it's slow to start up. It can take up to a second or two. Aplay's responses were pretty much instant, which I now realise was really valuable and important. I'd prefer to switch back to aplay, but I'd need to somehow get aplay to output audio at a relative volume. Is this possible somehow? And if not, would it be an idea to add this feature to aplay? Issue URL : https://github.com/alsa-project/alsa-utils/issues/45 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH AUTOSEL 5.7 15/53] ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H
by Sasha Levin 10 Jul '20

10 Jul '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit c8d2e2bfaeffa0f914330e8b4e45b986c8d30b58 ] Usually the DSP is not traditionally enabled on H skews but this might be used moving forward. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200617164755.18104-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/sof-pci-dev.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 7b1846aeadd59..3a71f813fb563 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -411,8 +411,11 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&cfl_desc}, #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) - { PCI_DEVICE(0x8086, 0x34C8), + { PCI_DEVICE(0x8086, 0x34C8), /* ICL-LP */ .driver_data = (unsigned long)&icl_desc}, + { PCI_DEVICE(0x8086, 0x3dc8), /* ICL-H */ + .driver_data = (unsigned long)&icl_desc}, + #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_JASPERLAKE) { PCI_DEVICE(0x8086, 0x38c8), @@ -431,8 +434,11 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&cml_desc}, #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) - { PCI_DEVICE(0x8086, 0xa0c8), + { PCI_DEVICE(0x8086, 0xa0c8), /* TGL-LP */ .driver_data = (unsigned long)&tgl_desc}, + { PCI_DEVICE(0x8086, 0x43c8), /* TGL-H */ + .driver_data = (unsigned long)&tgl_desc}, + #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) { PCI_DEVICE(0x8086, 0x4b55), -- 2.25.1

3 4

[PATCH] ASoC: atmel-pdmic: remove codec component
by Codrin Ciubotariu 10 Jul '20

10 Jul '20

The CPU and the codec both are represented now as components, so for PDMIC we are registering two componenets with the same name. Since there is no actual codec, we will merge the codec component into the CPU one and use a dummy codec instead, for the DAI link. As a bonus, debugfs will no longer report an error when will try to create entries for both componenets with the same name. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/atmel-pdmic.c | 110 +++++++++++++--------------------- 1 file changed, 41 insertions(+), 69 deletions(-) diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index 5245826cd99d..c2b639928c69 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -147,32 +147,26 @@ static int atmel_pdmic_cpu_dai_prepare(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = cpu_dai->component; u32 val; + int ret; /* Clean the PDMIC Converted Data Register */ - return regmap_read(dd->regmap, PDMIC_CDR, &val); -} - -static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = { - .startup = atmel_pdmic_cpu_dai_startup, - .shutdown = atmel_pdmic_cpu_dai_shutdown, - .prepare = atmel_pdmic_cpu_dai_prepare, -}; + ret = regmap_read(dd->regmap, PDMIC_CDR, &val); + if (ret < 0) + return 0; -#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + ret = snd_soc_component_update_bits(component, PDMIC_CR, + PDMIC_CR_ENPDM_MASK, + PDMIC_CR_ENPDM_DIS << + PDMIC_CR_ENPDM_SHIFT); + if (ret < 0) + return ret; -static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = { - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_KNOT, - .formats = ATMEL_PDMIC_FORMATS,}, - .ops = &atmel_pdmic_cpu_dai_ops, -}; + return 0; +} -static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = { - .name = "atmel-pdmic", -}; +#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) /* platform */ #define ATMEL_PDMIC_MAX_BUF_SIZE (64 * 1024) @@ -355,27 +349,16 @@ static int atmel_pdmic_component_probe(struct snd_soc_component *component) return 0; } -static struct snd_soc_component_driver soc_component_dev_pdmic = { - .probe = atmel_pdmic_component_probe, - .controls = atmel_pdmic_snd_controls, - .num_controls = ARRAY_SIZE(atmel_pdmic_snd_controls), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -/* codec dai component */ #define PDMIC_MR_PRESCAL_MAX_VAL 127 static int -atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *codec_dai) +atmel_pdmic_cpu_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; unsigned int rate_min = substream->runtime->hw.rate_min; unsigned int rate_max = substream->runtime->hw.rate_max; int fs = params_rate(params); @@ -445,21 +428,10 @@ atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream, return 0; } -static int atmel_pdmic_codec_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) -{ - struct snd_soc_component *component = codec_dai->component; - - snd_soc_component_update_bits(component, PDMIC_CR, PDMIC_CR_ENPDM_MASK, - PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT); - - return 0; -} - -static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) +static int atmel_pdmic_cpu_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 val; switch (cmd) { @@ -482,16 +454,16 @@ static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops atmel_pdmic_codec_dai_ops = { - .hw_params = atmel_pdmic_codec_dai_hw_params, - .prepare = atmel_pdmic_codec_dai_prepare, - .trigger = atmel_pdmic_codec_dai_trigger, +static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = { + .startup = atmel_pdmic_cpu_dai_startup, + .shutdown = atmel_pdmic_cpu_dai_shutdown, + .prepare = atmel_pdmic_cpu_dai_prepare, + .hw_params = atmel_pdmic_cpu_dai_hw_params, + .trigger = atmel_pdmic_cpu_dai_trigger, }; -#define ATMEL_PDMIC_CODEC_DAI_NAME "atmel-pdmic-hifi" -static struct snd_soc_dai_driver atmel_pdmic_codec_dai = { - .name = ATMEL_PDMIC_CODEC_DAI_NAME, +static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = { .capture = { .stream_name = "Capture", .channels_min = 1, @@ -499,7 +471,17 @@ static struct snd_soc_dai_driver atmel_pdmic_codec_dai = { .rates = SNDRV_PCM_RATE_KNOT, .formats = ATMEL_PDMIC_FORMATS, }, - .ops = &atmel_pdmic_codec_dai_ops, + .ops = &atmel_pdmic_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = { + .name = "atmel-pdmic", + .probe = atmel_pdmic_component_probe, + .controls = atmel_pdmic_snd_controls, + .num_controls = ARRAY_SIZE(atmel_pdmic_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, }; /* ASoC sound card */ @@ -528,9 +510,9 @@ static int atmel_pdmic_asoc_card_init(struct device *dev, dai_link->name = "PDMIC"; dai_link->stream_name = "PDMIC PCM"; - dai_link->codecs->dai_name = ATMEL_PDMIC_CODEC_DAI_NAME; + dai_link->codecs->dai_name = "snd-soc-dummy-dai"; dai_link->cpus->dai_name = dev_name(dev); - dai_link->codecs->name = dev_name(dev); + dai_link->codecs->name = "snd-soc-dummy"; dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; @@ -684,16 +666,6 @@ static int atmel_pdmic_probe(struct platform_device *pdev) return ret; } - /* register codec and codec dai */ - atmel_pdmic_codec_dai.capture.rate_min = rate_min; - atmel_pdmic_codec_dai.capture.rate_max = rate_max; - ret = devm_snd_soc_register_component(dev, &soc_component_dev_pdmic, - &atmel_pdmic_codec_dai, 1); - if (ret) { - dev_err(dev, "could not register component: %d\n", ret); - return ret; - } - /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) { -- 2.25.1

2 1

[PATCH v3] ASoC: atmel-classd: remove codec component
by Codrin Ciubotariu 10 Jul '20

10 Jul '20

The CPU and the codec both are represented now as components, so for CLASS-D we are registering two componenets with the same name. Since there is no actual codec, we will merge the codec component into the CPU one and use a dummy codec instead, for the DAI link. As a bonus, debugfs will no longer report an error when will try to create entries for both componenets with the same name. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Changes in v3: - fixed some newlines modified by no reason; Changes in v2: - removed no longer used ATMEL_CLASSD_CODEC_DAI_NAME macro; sound/soc/atmel/atmel-classd.c | 132 ++++++++++++--------------------- 1 file changed, 47 insertions(+), 85 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index e98601eccfa3..2d35b08f0565 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -120,39 +120,21 @@ static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + int err; regmap_write(dd->regmap, CLASSD_THR, 0x0); - return clk_prepare_enable(dd->pclk); -} - -static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - clk_disable_unprepare(dd->pclk); + err = clk_prepare_enable(dd->pclk); + if (err) + return err; + err = clk_prepare_enable(dd->gclk); + if (err) { + clk_disable_unprepare(dd->pclk); + return err; + } + return 0; } -static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { - .startup = atmel_classd_cpu_dai_startup, - .shutdown = atmel_classd_cpu_dai_shutdown, -}; - -static struct snd_soc_dai_driver atmel_classd_cpu_dai = { - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = ATMEL_CLASSD_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &atmel_classd_cpu_dai_ops, -}; - -static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { - .name = "atmel-classd", -}; - /* platform */ static int atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream, @@ -306,31 +288,10 @@ static int atmel_classd_component_resume(struct snd_soc_component *component) return regcache_sync(dd->regmap); } -static struct snd_soc_component_driver soc_component_dev_classd = { - .probe = atmel_classd_component_probe, - .resume = atmel_classd_component_resume, - .controls = atmel_classd_snd_controls, - .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -/* codec dai component */ -static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - return clk_prepare_enable(dd->gclk); -} - -static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai, - int mute) +static int atmel_classd_cpu_dai_digital_mute(struct snd_soc_dai *cpu_dai, + int mute) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK; @@ -373,13 +334,13 @@ static struct { }; static int -atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; int fs; int i, best, best_val, cur_val, ret; u32 mask, val; @@ -417,8 +378,8 @@ atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, } static void -atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); @@ -426,10 +387,10 @@ atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(dd->gclk); } -static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; snd_soc_component_update_bits(component, CLASSD_MR, CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK, @@ -439,10 +400,10 @@ static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, return 0; } -static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK; @@ -468,19 +429,16 @@ static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = { - .digital_mute = atmel_classd_codec_dai_digital_mute, - .startup = atmel_classd_codec_dai_startup, - .shutdown = atmel_classd_codec_dai_shutdown, - .hw_params = atmel_classd_codec_dai_hw_params, - .prepare = atmel_classd_codec_dai_prepare, - .trigger = atmel_classd_codec_dai_trigger, +static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { + .startup = atmel_classd_cpu_dai_startup, + .shutdown = atmel_classd_cpu_dai_shutdown, + .digital_mute = atmel_classd_cpu_dai_digital_mute, + .hw_params = atmel_classd_cpu_dai_hw_params, + .prepare = atmel_classd_cpu_dai_prepare, + .trigger = atmel_classd_cpu_dai_trigger, }; -#define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi" - -static struct snd_soc_dai_driver atmel_classd_codec_dai = { - .name = ATMEL_CLASSD_CODEC_DAI_NAME, +static struct snd_soc_dai_driver atmel_classd_cpu_dai = { .playback = { .stream_name = "Playback", .channels_min = 1, @@ -488,7 +446,18 @@ static struct snd_soc_dai_driver atmel_classd_codec_dai = { .rates = ATMEL_CLASSD_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &atmel_classd_codec_dai_ops, + .ops = &atmel_classd_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { + .name = "atmel-classd", + .probe = atmel_classd_component_probe, + .resume = atmel_classd_component_resume, + .controls = atmel_classd_snd_controls, + .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, }; /* ASoC sound card */ @@ -517,9 +486,9 @@ static int atmel_classd_asoc_card_init(struct device *dev, dai_link->name = "CLASSD"; dai_link->stream_name = "CLASSD PCM"; - dai_link->codecs->dai_name = ATMEL_CLASSD_CODEC_DAI_NAME; + dai_link->codecs->dai_name = "snd-soc-dummy-dai"; dai_link->cpus->dai_name = dev_name(dev); - dai_link->codecs->name = dev_name(dev); + dai_link->codecs->name = "snd-soc-dummy"; dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; @@ -620,13 +589,6 @@ static int atmel_classd_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(dev, &soc_component_dev_classd, - &atmel_classd_codec_dai, 1); - if (ret) { - dev_err(dev, "could not register component: %d\n", ret); - return ret; - } - /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) { -- 2.25.1

2 1

[PATCH v3 00/13] ASoC: Clean-up W=1 build warnings - part1
by Pierre-Louis Bossart 10 Jul '20

10 Jul '20

Both Lee Jones and I submitted separate series, this is the first part of the merged result, which includes all previously reviewed patches or suggested changes along with trivial ones for CONFIG_ACPI. Lee Jones (2): ASoC: codecs: jz4770: Remove defined but never used variable 'mic_boost_tlv' ASoC: ti: omap-mcbsp-st: Remove set, but unused variable 'w' Pierre-Louis Bossart (11): ASoC: atmel: fix kernel-doc ASoC: samsung: spdif: fix kernel-doc ASoC: samsung: pcm: fix kernel-doc ASoC: tegra: tegra20_das: remove always-true comparison ASoC: uniphier: aio-core: fix kernel-doc ASoC: codecs: da7210: fix kernel-doc ASoC: codecs: da7219: fix 'defined but not used' warning ASoC: codecs: cros_ec_codec: fix 'defined but not used' warning ASoC: codecs: es8316: fix 'defined but not used' warning ASoC: codecs: max98390: fix 'defined but not used' warning ASoC: codecs: rt*: fix 'defined but not used' warning sound/soc/atmel/atmel-pcm-dma.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 1 + sound/soc/codecs/cros_ec_codec.c | 2 ++ sound/soc/codecs/da7210.c | 12 +++++++----- sound/soc/codecs/da7219.c | 2 ++ sound/soc/codecs/es8316.c | 2 ++ sound/soc/codecs/jz4770.c | 1 - sound/soc/codecs/max98390.c | 8 -------- sound/soc/codecs/rt274.c | 2 ++ sound/soc/codecs/rt286.c | 2 ++ sound/soc/codecs/rt298.c | 2 ++ sound/soc/codecs/rt5660.c | 2 ++ sound/soc/codecs/rt5677-spi.c | 2 ++ sound/soc/samsung/pcm.c | 5 +++++ sound/soc/samsung/spdif.c | 6 +++--- sound/soc/tegra/tegra20_das.c | 3 +-- sound/soc/ti/omap-mcbsp-st.c | 3 --- sound/soc/uniphier/aio-core.c | 7 +++---- 18 files changed, 37 insertions(+), 27 deletions(-) -- 2.25.1

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