[alsa-devel] PulseAudio and softvol

[Takashi Iwai]

At Wed, 15 May 2013 13:22:03 +0200,
Jaroslav Kysela wrote:
> 
> Date 15.5.2013 13:03, David Henningsson wrote:
> > On 05/15/2013 12:53 PM, Jaroslav Kysela wrote:
> >> Date 15.5.2013 12:48, Takashi Iwai wrote:
> >>> At Wed, 15 May 2013 12:26:51 +0200,
> >>> Jaroslav Kysela wrote:
> >>>>
> >>>> Date 15.5.2013 11:55, Arun Raghavan wrote:
> >>>>> Hello,
> >>>>> A number of users have intermittently(?) been hitting a crash in
> >>>>> alsa-lib 1.0.27 [1, 2] related to the softvol plugin. I'm not able to
> >>>>> reproduce this reliably, so can't find an easy way to debug/fix.
> >>>>
> >>>> The problem is that the offsets are not in sync in this case [1]:
> >>>>
> >>>> src_offset = 38560
> >>>> dst_offset = 38568
> >>>> frames = 16374
> >>>>
> >>>> Could you reproduce this bug in any way? At least snd_pcm_dump() before
> >>>> the failing snd_pcm_mmap_commit() call might help to determine what was
> >>>> the status before the assert() was entered.
> >>>
> >>> Yep.  And this path is actually with volume 0dB, that is, a simply
> >>> passthrough in softvol.  Thus the bug may hit essentially any
> >>> plugins, not specifically softvol.
> >>>
> >>>
> >>>>> However, this raises a tangential question - why do we need softvol to
> >>>>> be plugged for 'front' at all? David explained to me that this is to
> >>>>> guarantee the existence of a PCM control. Perhaps I don't fully
> >>>>> understand this, because I'm unconvinced by the reason. Could someone
> >>>>> explain/refute?
> >>>>>
> >>>>> This is especially bad for us, from PulseAudio's perspective, because we
> >>>>> aren't getting a zero-copy path.
> >>>>
> >>>> If the softvol is set to 0dB (no attenuation or gain), then the ring
> >>>> buffer pointers are moved without any sample processing, so the
> >>>> zero-copy functionality is kept.
> >>>
> >>> Yeah, a sort of.  The mmap is cleared in the slave PCM, so actually
> >>> there will be copy operations in underlying layers even though softvol
> >>> itself does zero copy.
> >>>
> >>> Actually it makes no sense to keep softvol for PA, but the problem is
> >>> always the regression.  There are certainly users without PA, which
> >>> might still rely on the softvol for such hardware without the amp
> >>> control.
> >>>
> >>> Maybe We can add some flag to indicate whether to handle softvol or
> >>> not, e.g. defaults.pcm.skip_softvol, and let PA set this in its config
> >>> space.  Setting a config item itself would break anything, so it'll
> >>> still work with old alsa-lib (but with softvol).
> >>
> >> We have already SND_PCM_NO_SOFTVOL open mode for this purpose, so I
> >> wonder, why PA does not use it..
> > 
> > The problem is knowing whether PCM is a softvol or not. In some cases, 
> > we need to set PCM to control hardware volume.
> > 
> > Maybe, if we could figure this out somehow, we could ignore the PCM 
> > mixer control (or possibly set it to zero) in case PCM is a softvol,
> > and actually use it if PCM is not a softvol.
> > 
> > It does not look like this is currently possible from the simple mixer 
> > interface, but I might be missing something?
> 
> It is not possible. Perhaps, we may create a new dummy mixer control (in
> an inactive state) which will identify the presence of the softvol
> plugin, like:
> 
> "Softvol PCM Playback Volume" - full name for the raw control API
> "Softvol PCM" - simple mixer name

Well, if changing in such a way, I'd rather drop softvol from
HDA-Intel.conf.

If we could give some flag in mixer API, we could add a code to filter
out the user controls from the mixer's hctl.  But snd_mixer_attach()
takes only the string, and the string modifier may lead to the
incompatibility when used with an older version.  Hmm.


Takashi