[alsa-devel] PulseAudio and softvol

David Henningsson david.henningsson at canonical.com
Wed May 15 13:33:01 CEST 2013

On 05/15/2013 01:22 PM, 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

Or perhaps add a SND_CTL_NO_SOFTVOL flag that can be used in the call to 
snd_mixer_open / snd_ctl_open? That would make it somewhat consistent 
with the approach recommended for snd_pcm_open.

David Henningsson, Canonical Ltd.

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