Re: [GIT PULL] sound updates for 5.16-rc1

[Takashi Iwai <tiwai@suse.de>]

On Wed, 17 Nov 2021 18:48:26 +0100,
Kirill A. Shutemov wrote:
> 
> On Wed, Nov 17, 2021 at 05:24:28PM +0100, Takashi Iwai wrote:
> > On Wed, 17 Nov 2021 17:18:55 +0100,
> > Kirill A. Shutemov wrote:
> > > 
> > > On Wed, Nov 03, 2021 at 11:24:10AM +0100, Takashi Iwai wrote:
> > > > Linus,
> > > > 
> > > > please pull sound updates for v5.16-rc1 from:
> > > > 
> > > >   git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-5.16-rc1
> > > > 
> > > > The topmost commit is df0380b9539b04c1ae8854a984098da06d5f1e67
> > > > 
> > > > ----------------------------------------------------------------
> > > > 
> > > > sound updates for 5.16-rc1
> > > > 
> > > > Lots of code development have been see in ASoC side as usual, while
> > > > the continued development on memalloc helper and USB-audio low-
> > > > latency support are found in the rest.  Note that a few changes in the
> > > > unusual places like arch/sh are included, which are a part of ASoC DAI
> > > > format cleanups.
> > > > 
> > > > ALSA core:
> > > > - Continued memallloc helper updates and cleanups, now supporting
> > > >   non-coherent and non-contiguous pages
> > > > - Fixes for races in mixer OSS layer
> > > > 
> > > > ASoC:
> > > > - A new version of the audio graph card which supports a wider range
> > > >   of systems
> > > > - Several conversions to YAML DT bindings
> > > > - Continuing cleanups to the SOF and Intel code
> > > > - Move of the Cirrus DSP framework into drivers/firmware to allow for
> > > >   future use by non-audio DSPs
> > > > - An overhaul of the cs42l42 driver, correcting many problems
> > > > - DAI format terminology conversions over many drivers for cleanups
> > > > - Support for AMD Vangogh and Yelow Cap, Cirrus CS35L41, Maxim
> > > >   MAX98520 and MAX98360A, Mediatek MT8195, Nuvoton NAU8821, nVidia
> > > >   Tegra210, NXP i.MX8ULP, Qualcomm AudioReach, Realtek ALC5682I-VS,
> > > >   RT5682S, and RT9120 and Rockchip RV1126 and RK3568
> > > > 
> > > > USB-audio:
> > > > - Continued improvements on low-latency playback
> > > > - Quirks for Pioneer devices, Line6 HX-Stomp XL, Audient iD14
> > > > 
> > > > HD-audio:
> > > > - Reduce excessive udelay() calls on Intel platforms; this should
> > > >   reduce the CPU load with PulseAudio
> > > > - Quirks for HP and Clevo laptops
> > > > 
> > > > FireWire:
> > > > - Support for meter information on MOTU
> > > > 
> > > 
> > > ...
> > > 
> > > > Takashi Iwai (29):
> > > >       ALSA: hda: Reduce udelay() at SKL+ position reporting
> > > >       ALSA: hda: Use position buffer for SKL+ again
> > > >       ALSA: usb-audio: Restrict rates for the shared clocks
> > > >       ALSA: usb-audio: Fix possible race at sync of urb completions
> > > >       ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback
> > > >       ALSA: usb-audio: Disable low-latency playback for free-wheel mode
> > > >       ALSA: usb-audio: Disable low-latency mode for implicit feedback sync
> > > >       ALSA: usb-audio: Check available frames for the next packet size
> > > >       ALSA: usb-audio: Add spinlock to stop_urbs()
> > > >       ALSA: usb-audio: Improved lowlatency playback support
> > > 
> > > This commit breaks audio playback for me. The sample plays for fraction of
> > > a second than stops.
> > > 
> > > I use SMSL SA300. Let me know what ifo is needed to track it down.
> > > 
> > > Also I can test potential fixups if needed.
> > 
> > Could you give alsa-info.sh output with the device?  Run the script
> > with --no-upload option, and attach the output.
> 
> Attached. I run it on kernel without the commit. I hope it is fine.
> 
> One note: I use monolitic kernel without CONFIG_MODULES. The script
> complained about missing /proc/asound/modules and other modules related
> stuff.

It's a known problem I forgot, but this doesn't matter much.

> > Also /proc/asound/card*/stream* file for that usb device, too,
> > please.
> 
> Attached.

Thanks!

> > And, which environment are you testing, more exactly?  Which platform,
> > and with PulseAudio, pipewire, JACK, or direct ALSA device use?
> 
> Nothing fancy. Direct ALSA. I tested with 'speaker-test -c2 -t wav' during
> bisect.

Did you notice the problem with other applications, too?

When you wrote "The sample plays for fraction of a second than
stops.", it means that the speaker-test program stalls?  Or it's
shorter playbacks?

Also, please test booting with snd_usb_audio.lowlatency=0 boot
option.  Does it behave like before again?

> > For checking the behavior, we'd need to check hw_params in
> > /proc/asound/card*/pcm*/sub*/* for the corresponding device during
> > playback/capture operations.
> 
> You need this to be collected when the problem triggers, right?
> I will follow up with this.

I'll need to test my device at first.  Maybe tomorrow.


thanks,

Takashi