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

[PATCH] ASoC: Intel: cht_bsw_rt5672: Improve dai-set-fmt comment in cht_codec_fixup()
by Hans de Goede 10 Jul '20

10 Jul '20

As Pierre-Louis Bossart pointed out, saying that the default mode for the SSP is TDM 4 slot is not entirely accurate. There really are 2 default modes: The default mode for the SSP configuration is TDM 4 slot for the cpu-dai (hard-coded in DSP firmware), The default mode for the SSP configuration is I2S for the codec-dai (hard-coded in the 'SSP2-Codec" .dai_fmt masks, so far unused). This commit updates the comment in cht_codec_fixup() to properly reflect this. Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_rt5672.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 1470c3de7895..1812b786d33b 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -253,13 +253,17 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); /* - * Default mode for SSP configuration is TDM 4 slot. One board/design, - * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The - * second piggy-backed, output-only codec is inside the keyboard-dock - * (which has extra speakers). Unlike the main rt5672 codec, we cannot - * configure this codec, it is hard coded to use 2 channel 24 bit I2S. - * Since we only support 2 channels anyways, there is no need for TDM - * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere. + * The default mode for the cpu-dai is TDM 4 slot. The default mode + * for the codec-dai is I2S. So we need to either set the cpu-dai to + * I2S mode to match the codec-dai, or set the codec-dai to TDM 4 slot + * (or program both to yet another mode). + * One board, the Lenovo Miix 2 10, uses not 1 but 2 codecs connected + * to SSP2. The second piggy-backed, output-only codec is inside the + * keyboard-dock (which has extra speakers). Unlike the main rt5672 + * codec, we cannot configure this codec, it is hard coded to use + * 2 channel 24 bit I2S. For this to work we must use I2S mode on this + * board. Since we only support 2 channels anyways, there is no need + * for TDM on any cht-bsw-rt5672 designs. So we use I2S 2ch everywhere. */ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | -- 2.26.2

2 1

[PATCH v4 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. Changes since v3: Improved commit subjects from 'fix kernel-doc' as suggested by Lee Jones. In a couple of cases I just reverted to Lee's patches when the code was identical. Added a couple of CC: tags from Lee's patches. Added Arnaud Pouliquen's Acked-by tag in first patch. Lee Jones (6): ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' in kernel-doc 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: tlv320aic26: Demote seemingly unintentional kerneldoc header ASoC: codecs: ak4458: Remove set but never checked variable 'ret' Pierre-Louis Bossart (4): ASoC: sti: uniperif: fix 'defined by not used' warning ASoC: qcom: qdsp6: q6asm: Provide documentation for 'codec_profile' ASoC: sunxi: sun4i-i2s: add missing clock and format arguments in kernel-doc ASoC: codecs: rt5631: fix spurious kernel-doc start and missing arguments 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(-) base-commit: 6940701c715e193282cf8b31f970b2ea6eb26341 -- 2.25.1

2 11

[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

5 22

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

4 21

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