[alsa-devel] PulseAudio and softvol
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. 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. Cheers, Arun [1] https://bugzilla.redhat.com/show_bug.cgi?id=953352 [2] https://bugs.freedesktop.org/show_bug.cgi?id=64299
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.
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. Jaroslav
Cheers, Arun
[1] https://bugzilla.redhat.com/show_bug.cgi?id=953352 [2] https://bugs.freedesktop.org/show_bug.cgi?id=64299
-- Jaroslav Kysela <perex@perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.
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). thanks, Takashi
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.. Jaroslav -- Jaroslav Kysela <perex@perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.
At Wed, 15 May 2013 12:53:30 +0200, 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..
Oh, yeah, I completely forgot it! Takashi
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? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic
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 Jaroslav -- Jaroslav Kysela <perex@perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.
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. https://launchpad.net/~diwic
At Wed, 15 May 2013 13:33:01 +0200, David Henningsson wrote:
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.
It'd be rather a flag to exclude the user controls, not specific to softvol. But, the problem is that snd_mixer_attach() takes no extra flag argument. So, we may need to add a new function, either defining only the attach mode, or an equivalent function with snd_mixer_attach() but with an extra flag argument. Takashi
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
On 05/15/2013 02:42 PM, Takashi Iwai wrote:
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.
That seems solvable to me, something like this: diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 56e023d..4afa979 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -65,13 +65,14 @@ static int snd_mixer_compare_default(const snd_mixer_elem_t *c1, * \param mode Open mode * \return 0 on success otherwise a negative error code */ -int snd_mixer_open(snd_mixer_t **mixerp, int mode ATTRIBUTE_UNUSED) +int snd_mixer_open(snd_mixer_t **mixerp, int mode) { snd_mixer_t *mixer; assert(mixerp); mixer = calloc(1, sizeof(*mixer)); if (mixer == NULL) return -ENOMEM; + mixer->attach_mode = mode; INIT_LIST_HEAD(&mixer->slaves); INIT_LIST_HEAD(&mixer->classes); INIT_LIST_HEAD(&mixer->elems); @@ -200,7 +201,7 @@ int snd_mixer_attach(snd_mixer_t *mixer, const char *name) snd_hctl_t *hctl; int err; - err = snd_hctl_open(&hctl, name, 0); + err = snd_hctl_open(&hctl, name, mixer->attach_mode); if (err < 0) return err; err = snd_mixer_attach_hctl(mixer, hctl); diff --git a/src/mixer/mixer_local.h b/src/mixer/mixer_local.h index 27b4a3b..2d1866e 100644 --- a/src/mixer/mixer_local.h +++ b/src/mixer/mixer_local.h @@ -71,6 +71,7 @@ struct _snd_mixer { unsigned int count; unsigned int alloc; unsigned int events; + int attach_mode; snd_mixer_callback_t callback; void *callback_private; snd_mixer_compare_t compare; -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic
At Wed, 15 May 2013 14:47:15 +0200, David Henningsson wrote:
On 05/15/2013 02:42 PM, Takashi Iwai wrote:
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.
That seems solvable to me, something like this:
diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 56e023d..4afa979 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -65,13 +65,14 @@ static int snd_mixer_compare_default(const snd_mixer_elem_t *c1, * \param mode Open mode * \return 0 on success otherwise a negative error code */ -int snd_mixer_open(snd_mixer_t **mixerp, int mode ATTRIBUTE_UNUSED) +int snd_mixer_open(snd_mixer_t **mixerp, int mode)
Let's hope that no one sets the mode value ever... :) But yes, other than that, it looks feasible indeed. Takashi
{ snd_mixer_t *mixer; assert(mixerp); mixer = calloc(1, sizeof(*mixer)); if (mixer == NULL) return -ENOMEM; + mixer->attach_mode = mode; INIT_LIST_HEAD(&mixer->slaves); INIT_LIST_HEAD(&mixer->classes); INIT_LIST_HEAD(&mixer->elems); @@ -200,7 +201,7 @@ int snd_mixer_attach(snd_mixer_t *mixer, const char *name) snd_hctl_t *hctl; int err;
- err = snd_hctl_open(&hctl, name, 0); + err = snd_hctl_open(&hctl, name, mixer->attach_mode); if (err < 0) return err; err = snd_mixer_attach_hctl(mixer, hctl); diff --git a/src/mixer/mixer_local.h b/src/mixer/mixer_local.h index 27b4a3b..2d1866e 100644 --- a/src/mixer/mixer_local.h +++ b/src/mixer/mixer_local.h @@ -71,6 +71,7 @@ struct _snd_mixer { unsigned int count; unsigned int alloc; unsigned int events; + int attach_mode; snd_mixer_callback_t callback; void *callback_private; snd_mixer_compare_t compare;
-- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic
Date 15.5.2013 14:47, David Henningsson wrote:
On 05/15/2013 02:42 PM, Takashi Iwai wrote:
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.
That seems solvable to me, something like this:
diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 56e023d..4afa979 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -65,13 +65,14 @@ static int snd_mixer_compare_default(const snd_mixer_elem_t *c1, * \param mode Open mode * \return 0 on success otherwise a negative error code */ -int snd_mixer_open(snd_mixer_t **mixerp, int mode ATTRIBUTE_UNUSED) +int snd_mixer_open(snd_mixer_t **mixerp, int mode)
Yes, it could be implemented in this way. A special TLV entry may be introduced to detect, if the control is created by softvol. I wouldn't ignore all user created controls, because they can be used to reroute the controls to the real hardware (the alsaloop daemon does it in this way and PA can run on top). Jaroslav -- Jaroslav Kysela <perex@perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.
At Wed, 15 May 2013 14:52:53 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 14:47, David Henningsson wrote:
On 05/15/2013 02:42 PM, Takashi Iwai wrote:
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.
That seems solvable to me, something like this:
diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 56e023d..4afa979 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -65,13 +65,14 @@ static int snd_mixer_compare_default(const snd_mixer_elem_t *c1, * \param mode Open mode * \return 0 on success otherwise a negative error code */ -int snd_mixer_open(snd_mixer_t **mixerp, int mode ATTRIBUTE_UNUSED) +int snd_mixer_open(snd_mixer_t **mixerp, int mode)
Yes, it could be implemented in this way. A special TLV entry may be introduced to detect, if the control is created by softvol.
The additional TLV won't work if a control is restored by alsactl, for example, unfortunately.
I wouldn't ignore all user created controls, because they can be used to reroute the controls to the real hardware (the alsaloop daemon does it in this way and PA can run on top).
Yeah, it's a difficult point. Even a PCM control created by softvol might be used by other plugins. We can't exclude such a possibility. In other words, if user wants to run PA in special environment with virtual devices, it needs a special setup that allows indirect accesses, basically without any limitation. Takashi
Date 15.5.2013 15:05, Takashi Iwai wrote:
At Wed, 15 May 2013 14:52:53 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 14:47, David Henningsson wrote:
On 05/15/2013 02:42 PM, Takashi Iwai wrote:
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.
That seems solvable to me, something like this:
diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 56e023d..4afa979 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -65,13 +65,14 @@ static int snd_mixer_compare_default(const snd_mixer_elem_t *c1, * \param mode Open mode * \return 0 on success otherwise a negative error code */ -int snd_mixer_open(snd_mixer_t **mixerp, int mode ATTRIBUTE_UNUSED) +int snd_mixer_open(snd_mixer_t **mixerp, int mode)
Yes, it could be implemented in this way. A special TLV entry may be introduced to detect, if the control is created by softvol.
The additional TLV won't work if a control is restored by alsactl, for example, unfortunately.
This looks like a bug, doesn't? Anyway, I see some TLV restore code in alsactl, but the support for all control types should be added not only for SND_CTL_ELEM_TYPE_INTEGER. Jaroslav -- Jaroslav Kysela <perex@perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.
At Wed, 15 May 2013 15:12:17 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 15:05, Takashi Iwai wrote:
At Wed, 15 May 2013 14:52:53 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 14:47, David Henningsson wrote:
On 05/15/2013 02:42 PM, Takashi Iwai wrote:
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.
That seems solvable to me, something like this:
diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 56e023d..4afa979 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -65,13 +65,14 @@ static int snd_mixer_compare_default(const snd_mixer_elem_t *c1, * \param mode Open mode * \return 0 on success otherwise a negative error code */ -int snd_mixer_open(snd_mixer_t **mixerp, int mode ATTRIBUTE_UNUSED) +int snd_mixer_open(snd_mixer_t **mixerp, int mode)
Yes, it could be implemented in this way. A special TLV entry may be introduced to detect, if the control is created by softvol.
The additional TLV won't work if a control is restored by alsactl, for example, unfortunately.
This looks like a bug, doesn't? Anyway, I see some TLV restore code in alsactl, but the support for all control types should be added not only for SND_CTL_ELEM_TYPE_INTEGER.
Well, alsactl would just restore what's saved. So, if the saved data already contains the softvol ctl element with the old TLV, it's simply restored as is. You may think of adding the code to softvol plugin to automatically rewrite TLV of the existing ctl element if it contains no new TLV type. But, PA shall skip softvol. Thus, it won't be touched. And yet, PA would like to skip the control elements that have been created beforehand. Certainly this is a corner case, but the requirement is incompatible with old data. If it's only about the change of the library code, it would work by a simple update. But if an additional metadata has to be embedded, it's a different question... Takashi
Date 15.5.2013 15:26, Takashi Iwai wrote:
At Wed, 15 May 2013 15:12:17 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 15:05, Takashi Iwai wrote:
At Wed, 15 May 2013 14:52:53 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 14:47, David Henningsson wrote:
On 05/15/2013 02:42 PM, Takashi Iwai wrote:
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.
That seems solvable to me, something like this:
diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 56e023d..4afa979 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -65,13 +65,14 @@ static int snd_mixer_compare_default(const snd_mixer_elem_t *c1, * \param mode Open mode * \return 0 on success otherwise a negative error code */ -int snd_mixer_open(snd_mixer_t **mixerp, int mode ATTRIBUTE_UNUSED) +int snd_mixer_open(snd_mixer_t **mixerp, int mode)
Yes, it could be implemented in this way. A special TLV entry may be introduced to detect, if the control is created by softvol.
The additional TLV won't work if a control is restored by alsactl, for example, unfortunately.
This looks like a bug, doesn't? Anyway, I see some TLV restore code in alsactl, but the support for all control types should be added not only for SND_CTL_ELEM_TYPE_INTEGER.
Well, alsactl would just restore what's saved. So, if the saved data already contains the softvol ctl element with the old TLV, it's simply restored as is.
It's enough.
You may think of adding the code to softvol plugin to automatically rewrite TLV of the existing ctl element if it contains no new TLV type. But, PA shall skip softvol. Thus, it won't be touched. And yet, PA would like to skip the control elements that have been created beforehand.
The alsa-lib code can be modified to create or modify the user space control also in the SND_PCM_NO_SOFTVOL case, so the mixer API will be informed that the PCM controls belongs to softvol. I don't see any other problems. Jaroslav -- Jaroslav Kysela <perex@perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.
At Wed, 15 May 2013 16:55:05 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 15:26, Takashi Iwai wrote:
At Wed, 15 May 2013 15:12:17 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 15:05, Takashi Iwai wrote:
At Wed, 15 May 2013 14:52:53 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 14:47, David Henningsson wrote:
On 05/15/2013 02:42 PM, Takashi Iwai wrote: > 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.
That seems solvable to me, something like this:
diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 56e023d..4afa979 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -65,13 +65,14 @@ static int snd_mixer_compare_default(const snd_mixer_elem_t *c1, * \param mode Open mode * \return 0 on success otherwise a negative error code */ -int snd_mixer_open(snd_mixer_t **mixerp, int mode ATTRIBUTE_UNUSED) +int snd_mixer_open(snd_mixer_t **mixerp, int mode)
Yes, it could be implemented in this way. A special TLV entry may be introduced to detect, if the control is created by softvol.
The additional TLV won't work if a control is restored by alsactl, for example, unfortunately.
This looks like a bug, doesn't? Anyway, I see some TLV restore code in alsactl, but the support for all control types should be added not only for SND_CTL_ELEM_TYPE_INTEGER.
Well, alsactl would just restore what's saved. So, if the saved data already contains the softvol ctl element with the old TLV, it's simply restored as is.
It's enough.
Enough for...? It restores the value without the new TLV, thus it doesn't show it's a softvol element.
You may think of adding the code to softvol plugin to automatically rewrite TLV of the existing ctl element if it contains no new TLV type. But, PA shall skip softvol. Thus, it won't be touched. And yet, PA would like to skip the control elements that have been created beforehand.
The alsa-lib code can be modified to create or modify the user space control also in the SND_PCM_NO_SOFTVOL case, so the mixer API will be informed that the PCM controls belongs to softvol.
But how would you know exactly? Parsing the PCM definition at each time if a user ctl element is found and check whether it *might* belong to softvol plugin defined in some of card's default config? What if a user takes own definition temporarily? There can be endless corner cases. BTW, does the alsaloop device just work as is, i.e. without specifying anything in PA's configuration? I'm asking it because what we're dealing with is the case where PA probes as default via "front", "spdif" or such pre-definitions bound with a real sound card instance. The special filter could be used only for these cases. For the devices specified by user, it doesn't need such filters. Takashi
Date 15.5.2013 17:06, Takashi Iwai wrote:
At Wed, 15 May 2013 16:55:05 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 15:26, Takashi Iwai wrote:
At Wed, 15 May 2013 15:12:17 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 15:05, Takashi Iwai wrote:
At Wed, 15 May 2013 14:52:53 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 14:47, David Henningsson wrote: > On 05/15/2013 02:42 PM, Takashi Iwai wrote: >> 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. > > That seems solvable to me, something like this: > > diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c > index 56e023d..4afa979 100644 > --- a/src/mixer/mixer.c > +++ b/src/mixer/mixer.c > @@ -65,13 +65,14 @@ static int snd_mixer_compare_default(const > snd_mixer_elem_t *c1, > * \param mode Open mode > * \return 0 on success otherwise a negative error code > */ > -int snd_mixer_open(snd_mixer_t **mixerp, int mode ATTRIBUTE_UNUSED) > +int snd_mixer_open(snd_mixer_t **mixerp, int mode)
Yes, it could be implemented in this way. A special TLV entry may be introduced to detect, if the control is created by softvol.
The additional TLV won't work if a control is restored by alsactl, for example, unfortunately.
This looks like a bug, doesn't? Anyway, I see some TLV restore code in alsactl, but the support for all control types should be added not only for SND_CTL_ELEM_TYPE_INTEGER.
Well, alsactl would just restore what's saved. So, if the saved data already contains the softvol ctl element with the old TLV, it's simply restored as is.
It's enough.
Enough for...? It restores the value without the new TLV, thus it doesn't show it's a softvol element.
Yes, for the first call, but then PA will open the pcm device and this call will add the softvol information to TLV which inactivates the softvol control in the mixer.
You may think of adding the code to softvol plugin to automatically rewrite TLV of the existing ctl element if it contains no new TLV type. But, PA shall skip softvol. Thus, it won't be touched. And yet, PA would like to skip the control elements that have been created beforehand.
The alsa-lib code can be modified to create or modify the user space control also in the SND_PCM_NO_SOFTVOL case, so the mixer API will be informed that the PCM controls belongs to softvol.
But how would you know exactly? Parsing the PCM definition at each time if a user ctl element is found and check whether it *might* belong to softvol plugin defined in some of card's default config?
No. The information in TLV would be enough to determine the softvol functionality.
What if a user takes own definition temporarily?
Then user is responsible to remove this control later.
There can be endless corner cases.
I'm not sure what you talk about. Yes, everything can be misused, but we're talking about the standard usage. You can do weird things with the softvol user controls anyway (remove them during runtime, because they're not locked).
BTW, does the alsaloop device just work as is, i.e. without specifying anything in PA's configuration?
The alsaloop can run on top of the snd-aloop, so it behaves like a standard hardware, only the mixer controls can be rerouted using the user controls to the real card. Jaroslav -- Jaroslav Kysela <perex@perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.
At Wed, 15 May 2013 17:25:08 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 17:06, Takashi Iwai wrote:
At Wed, 15 May 2013 16:55:05 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 15:26, Takashi Iwai wrote:
At Wed, 15 May 2013 15:12:17 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 15:05, Takashi Iwai wrote:
At Wed, 15 May 2013 14:52:53 +0200, Jaroslav Kysela wrote: > > Date 15.5.2013 14:47, David Henningsson wrote: >> On 05/15/2013 02:42 PM, Takashi Iwai wrote: >>> 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. >> >> That seems solvable to me, something like this: >> >> diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c >> index 56e023d..4afa979 100644 >> --- a/src/mixer/mixer.c >> +++ b/src/mixer/mixer.c >> @@ -65,13 +65,14 @@ static int snd_mixer_compare_default(const >> snd_mixer_elem_t *c1, >> * \param mode Open mode >> * \return 0 on success otherwise a negative error code >> */ >> -int snd_mixer_open(snd_mixer_t **mixerp, int mode ATTRIBUTE_UNUSED) >> +int snd_mixer_open(snd_mixer_t **mixerp, int mode) > > Yes, it could be implemented in this way. A special TLV entry may be > introduced to detect, if the control is created by softvol.
The additional TLV won't work if a control is restored by alsactl, for example, unfortunately.
This looks like a bug, doesn't? Anyway, I see some TLV restore code in alsactl, but the support for all control types should be added not only for SND_CTL_ELEM_TYPE_INTEGER.
Well, alsactl would just restore what's saved. So, if the saved data already contains the softvol ctl element with the old TLV, it's simply restored as is.
It's enough.
Enough for...? It restores the value without the new TLV, thus it doesn't show it's a softvol element.
Yes, for the first call, but then PA will open the pcm device and this call will add the softvol information to TLV which inactivates the softvol control in the mixer.
Yes, but the invocation of PCM softvol isn't guaranteed to be first before the reference to the already existing user ctl element. snd_mixer_open() can be called before that.
You may think of adding the code to softvol plugin to automatically rewrite TLV of the existing ctl element if it contains no new TLV type. But, PA shall skip softvol. Thus, it won't be touched. And yet, PA would like to skip the control elements that have been created beforehand.
The alsa-lib code can be modified to create or modify the user space control also in the SND_PCM_NO_SOFTVOL case, so the mixer API will be informed that the PCM controls belongs to softvol.
But how would you know exactly? Parsing the PCM definition at each time if a user ctl element is found and check whether it *might* belong to softvol plugin defined in some of card's default config?
No. The information in TLV would be enough to determine the softvol functionality.
What if a user takes own definition temporarily?
Then user is responsible to remove this control later.
That's messy. We provide no such tool.
There can be endless corner cases.
I'm not sure what you talk about. Yes, everything can be misused, but we're talking about the standard usage. You can do weird things with the softvol user controls anyway (remove them during runtime, because they're not locked).
I know I'm picky, but adding a new metadata *onto* the existing data structure has to be always done carefully. Otherwise it'll hit back us later. For example, another corner case I can think of easily is that user downgrades alsa-lib. Then softvol overrides the TLV again without the new tag (the current softvol code assumes the single TLV and overrides if it doesn't fit), and it can be saved so...
BTW, does the alsaloop device just work as is, i.e. without specifying anything in PA's configuration?
The alsaloop can run on top of the snd-aloop, so it behaves like a standard hardware, only the mixer controls can be rerouted using the user controls to the real card.
OK. But, looking at the current implementation, you are allowed to map freely the source ctl element. It can lead to the same problem in PA. The very reason we'd like to filter out the mixer control created by softvol is that this mixer element confuses PA as if it actually changes the volume (e.g. "PCM") although PA ignores the softvol. If user creates PCM volume in alsaloop in a different fashion as PA expected, the similar problem may happen. How can we detect this logically...? In other words, how can PA adjust the mixer elements for alsaloop properly? Takashi
On 05/15/2013 06:28 PM, Takashi Iwai wrote:
At Wed, 15 May 2013 17:25:08 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 17:06, Takashi Iwai wrote:
Enough for...? It restores the value without the new TLV, thus it doesn't show it's a softvol element.
Yes, for the first call, but then PA will open the pcm device and this call will add the softvol information to TLV which inactivates the softvol control in the mixer.
Yes, but the invocation of PCM softvol isn't guaranteed to be first before the reference to the already existing user ctl element. snd_mixer_open() can be called before that.
So this is a transient problem, right? As soon as the first PCM is opened, the TLV would be corrected, and then stay corrected for all times to come. And looking at the current PulseAudio code, it does open the pcm device before it opens the mixer/ctl device. So, if this isn't possible to solve in a better way, maybe we need to be pragmatic about it - PulseAudio is the only application we know that would care, and it opens the pcm device first. So in practice, it looks like the TLV approach would work.
You may think of adding the code to softvol plugin to automatically rewrite TLV of the existing ctl element if it contains no new TLV type. But, PA shall skip softvol. Thus, it won't be touched. And yet, PA would like to skip the control elements that have been created beforehand.
The alsa-lib code can be modified to create or modify the user space control also in the SND_PCM_NO_SOFTVOL case, so the mixer API will be informed that the PCM controls belongs to softvol.
But how would you know exactly? Parsing the PCM definition at each time if a user ctl element is found and check whether it *might* belong to softvol plugin defined in some of card's default config?
No. The information in TLV would be enough to determine the softvol functionality.
What if a user takes own definition temporarily?
Then user is responsible to remove this control later.
That's messy. We provide no such tool.
There can be endless corner cases.
I'm not sure what you talk about. Yes, everything can be misused, but we're talking about the standard usage. You can do weird things with the softvol user controls anyway (remove them during runtime, because they're not locked).
I know I'm picky, but adding a new metadata *onto* the existing data structure has to be always done carefully. Otherwise it'll hit back us later.
I think it's good that you are picky. Missing a use case can cause problems indeed. But is there another solution to this problem which is better?
For example, another corner case I can think of easily is that user downgrades alsa-lib. Then softvol overrides the TLV again without the new tag (the current softvol code assumes the single TLV and overrides if it doesn't fit), and it can be saved so...
BTW, does the alsaloop device just work as is, i.e. without specifying anything in PA's configuration?
The alsaloop can run on top of the snd-aloop, so it behaves like a standard hardware, only the mixer controls can be rerouted using the user controls to the real card.
OK. But, looking at the current implementation, you are allowed to map freely the source ctl element. It can lead to the same problem in PA.
The very reason we'd like to filter out the mixer control created by softvol is that this mixer element confuses PA as if it actually changes the volume (e.g. "PCM") although PA ignores the softvol. If user creates PCM volume in alsaloop in a different fashion as PA expected, the similar problem may happen. How can we detect this logically...? In other words, how can PA adjust the mixer elements for alsaloop properly?
So if alsaloop is run, only once, that could cause a control to be added for all future, due to alsactl saving and restoring it? If so, that looks like a problem with alsaloop. If it adds controls, it should also remove them. If no, I don't think we need to worry. Alsaloop is probably mostly used on non-PA systems (as PA has module-loopback which does the same thing). -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic
Date 16.5.2013 08:31, David Henningsson wrote:
On 05/15/2013 06:28 PM, Takashi Iwai wrote:
At Wed, 15 May 2013 17:25:08 +0200, Jaroslav Kysela wrote:
Date 15.5.2013 17:06, Takashi Iwai wrote:
Enough for...? It restores the value without the new TLV, thus it doesn't show it's a softvol element.
Yes, for the first call, but then PA will open the pcm device and this call will add the softvol information to TLV which inactivates the softvol control in the mixer.
Yes, but the invocation of PCM softvol isn't guaranteed to be first before the reference to the already existing user ctl element. snd_mixer_open() can be called before that.
So this is a transient problem, right? As soon as the first PCM is opened, the TLV would be corrected, and then stay corrected for all times to come.
And looking at the current PulseAudio code, it does open the pcm device before it opens the mixer/ctl device. So, if this isn't possible to solve in a better way, maybe we need to be pragmatic about it - PulseAudio is the only application we know that would care, and it opens the pcm device first. So in practice, it looks like the TLV approach would work.
It would work even if the mixer is opened before PCM, because the mixer will get notified about the TLV change and can make the PCM element inactive after. PA should only handle this situation correctly.
You may think of adding the code to softvol plugin to automatically rewrite TLV of the existing ctl element if it contains no new TLV type. But, PA shall skip softvol. Thus, it won't be touched. And yet, PA would like to skip the control elements that have been created beforehand.
The alsa-lib code can be modified to create or modify the user space control also in the SND_PCM_NO_SOFTVOL case, so the mixer API will be informed that the PCM controls belongs to softvol.
But how would you know exactly? Parsing the PCM definition at each time if a user ctl element is found and check whether it *might* belong to softvol plugin defined in some of card's default config?
No. The information in TLV would be enough to determine the softvol functionality.
What if a user takes own definition temporarily?
Then user is responsible to remove this control later.
That's messy. We provide no such tool.
Perhaps amixer should be extended.
There can be endless corner cases.
I'm not sure what you talk about. Yes, everything can be misused, but we're talking about the standard usage. You can do weird things with the softvol user controls anyway (remove them during runtime, because they're not locked).
I know I'm picky, but adding a new metadata *onto* the existing data structure has to be always done carefully. Otherwise it'll hit back us later.
I think it's good that you are picky. Missing a use case can cause problems indeed. But is there another solution to this problem which is better?
For example, another corner case I can think of easily is that user downgrades alsa-lib. Then softvol overrides the TLV again without the new tag (the current softvol code assumes the single TLV and overrides if it doesn't fit), and it can be saved so...
BTW, does the alsaloop device just work as is, i.e. without specifying anything in PA's configuration?
The alsaloop can run on top of the snd-aloop, so it behaves like a standard hardware, only the mixer controls can be rerouted using the user controls to the real card.
OK. But, looking at the current implementation, you are allowed to map freely the source ctl element. It can lead to the same problem in PA.
The very reason we'd like to filter out the mixer control created by softvol is that this mixer element confuses PA as if it actually
Why it can confuse PA? It behaves like standard soundcard and user can select the forwarded controls (for example Master and PCM or Front only). PA don't have any hint that the audio is rerouted and it's correct.
changes the volume (e.g. "PCM") although PA ignores the softvol. If user creates PCM volume in alsaloop in a different fashion as PA expected, the similar problem may happen. How can we detect this logically...? In other words, how can PA adjust the mixer elements for alsaloop properly?
So if alsaloop is run, only once, that could cause a control to be added for all future, due to alsactl saving and restoring it?
If so, that looks like a problem with alsaloop. If it adds controls, it should also remove them.
These controls are removed when alsaloop exits. Jaroslav -- Jaroslav Kysela <perex@perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.
On Thu, 2013-05-16 at 08:31 +0200, David Henningsson wrote:
On 05/15/2013 06:28 PM, Takashi Iwai wrote: [...]
Yes, but the invocation of PCM softvol isn't guaranteed to be first before the reference to the already existing user ctl element. snd_mixer_open() can be called before that.
So this is a transient problem, right? As soon as the first PCM is opened, the TLV would be corrected, and then stay corrected for all times to come.
And looking at the current PulseAudio code, it does open the pcm device before it opens the mixer/ctl device. So, if this isn't possible to solve in a better way, maybe we need to be pragmatic about it - PulseAudio is the only application we know that would care, and it opens the pcm device first. So in practice, it looks like the TLV approach would work.
[...]
The very reason we'd like to filter out the mixer control created by softvol is that this mixer element confuses PA as if it actually changes the volume (e.g. "PCM") although PA ignores the softvol. If
Actually, we don't currently ignore softvol. I guess we could add the no-softvol flag once we're able to make sure we don't have any softvol controls.
user creates PCM volume in alsaloop in a different fashion as PA expected, the similar problem may happen. How can we detect this logically...? In other words, how can PA adjust the mixer elements for alsaloop properly?
So if alsaloop is run, only once, that could cause a control to be added for all future, due to alsactl saving and restoring it?
If so, that looks like a problem with alsaloop. If it adds controls, it should also remove them.
If no, I don't think we need to worry. Alsaloop is probably mostly used on non-PA systems (as PA has module-loopback which does the same thing).
Seems this thread died out. I didn't quite understand the alsaloop/alsactl-specific concerns, tbh. What do we need to do to take this forwards? Cheers, Arun
On 06/26/2013 08:59 AM, Arun Raghavan wrote:
On Thu, 2013-05-16 at 08:31 +0200, David Henningsson wrote:
On 05/15/2013 06:28 PM, Takashi Iwai wrote: [...]
Yes, but the invocation of PCM softvol isn't guaranteed to be first before the reference to the already existing user ctl element. snd_mixer_open() can be called before that.
So this is a transient problem, right? As soon as the first PCM is opened, the TLV would be corrected, and then stay corrected for all times to come.
And looking at the current PulseAudio code, it does open the pcm device before it opens the mixer/ctl device. So, if this isn't possible to solve in a better way, maybe we need to be pragmatic about it - PulseAudio is the only application we know that would care, and it opens the pcm device first. So in practice, it looks like the TLV approach would work.
[...]
The very reason we'd like to filter out the mixer control created by softvol is that this mixer element confuses PA as if it actually changes the volume (e.g. "PCM") although PA ignores the softvol. If
Actually, we don't currently ignore softvol. I guess we could add the no-softvol flag once we're able to make sure we don't have any softvol controls.
Correct.
user creates PCM volume in alsaloop in a different fashion as PA expected, the similar problem may happen. How can we detect this logically...? In other words, how can PA adjust the mixer elements for alsaloop properly?
So if alsaloop is run, only once, that could cause a control to be added for all future, due to alsactl saving and restoring it?
If so, that looks like a problem with alsaloop. If it adds controls, it should also remove them.
If no, I don't think we need to worry. Alsaloop is probably mostly used on non-PA systems (as PA has module-loopback which does the same thing).
Seems this thread died out. I didn't quite understand the alsaloop/alsactl-specific concerns, tbh. What do we need to do to take this forwards?
It was suggested that softvol controls would store this information in TLV, which leads to a transient problem on upgrading, but nobody suggested anything better. The transient problem is caused by alsactl loading the control from asound.state, which contains the old TLV information. But as soon as a PCM stream is opened, this will be corrected, AFAIU, and then the correct TLV information will be stored in asound.state at next reboot. So I'm guessing that patches are welcome for that solution (in combination with a CTL_OPEN_NO_SOFTVOL flag), and nobody volunteered yet to implement it. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic
On Wed, 2013-05-15 at 12:26 +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.
Unfortunately, after the time I got the backtrace, I haven't been able to reproduce the problem. -- Arun
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.
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?
Because alc660 codec did not has any hardware volume control on the output http://www.alsa-project.org/db/?f=a327b0925a1697d60ea427fe3f16821d7ef7030b
participants (5)
-
Arun Raghavan -
David Henningsson -
Jaroslav Kysela -
Raymond Yau -
Takashi Iwai