proposal: snd_pcm_start_at()

[Tim Cussins]

Hi all,

I'm Tim: I work at Linn Products Ltd - we make Network Music Players,
amongst other things.

As you might imagine, synchronised-start is important when multiple
devices on the network are rendering the same audio. We'd be interested
in contributing a small expansion of the alsa-lib API to support
synchronised start.

Assuming we can synchronise the audio clocks (I'm aware this is not
trivial - It's not the topic of this post), we'd propose something like:

int snd_pcm_start_at(snd_pcm_t* pcm, snd_htimestamp_t* tstamp);

and playback would begin as close to tstamp as possible. If tstamp is in
the past, it would should return an error.

Recent work by Takashi Iwai enables client code to set the clock type of
timestamps using snd_pcm_sw_params_set_tstamp_type(). This context could
quite naturally extend to tstamp argument of snd_pcm_start_at().

Before I get stuck into working up the details under the hood, it'd be
good to get some feedback/objections regarding this approach.

Thanks in advance!
Tim

[Pierre-Louis Bossart]

Post by Tim Cussins
Hi all,
I'm Tim: I work at Linn Products Ltd - we make Network Music Players,
amongst other things.
As you might imagine, synchronised-start is important when multiple
devices on the network are rendering the same audio. We'd be interested
in contributing a small expansion of the alsa-lib API to support
synchronised start.
Assuming we can synchronise the audio clocks (I'm aware this is not
int snd_pcm_start_at(snd_pcm_t* pcm, snd_htimestamp_t* tstamp);
and playback would begin as close to tstamp as possible. If tstamp is in
the past, it would should return an error.
Recent work by Takashi Iwai enables client code to set the clock type of
timestamps using snd_pcm_sw_params_set_tstamp_type(). This context could
quite naturally extend to tstamp argument of snd_pcm_start_at().
Before I get stuck into working up the details under the hood, it'd be
good to get some feedback/objections regarding this approach.

It's probably better idea to start PCM playback with a bunch of zeroes
and then rely on existing timestamping to insert samples at the right
location in the ring buffer - which you have to do anyway to compensate
for drifts between your network clock and audio clock. This is a more
predictable solution that abstracts away all the time needed to arm DMA,
FIFOs, etc. The only hardware-dependent variable that would remain is
the precision/granularity of the timestamping.
-Pierre

[Tim Cussins]

Hi Peirre,

Post by Pierre-Louis Bossart

Post by Tim Cussins
Hi all,
I'm Tim: I work at Linn Products Ltd - we make Network Music Players,
amongst other things.
As you might imagine, synchronised-start is important when multiple
devices on the network are rendering the same audio. We'd be interested
in contributing a small expansion of the alsa-lib API to support
synchronised start.
Assuming we can synchronise the audio clocks (I'm aware this is not
int snd_pcm_start_at(snd_pcm_t* pcm, snd_htimestamp_t* tstamp);
and playback would begin as close to tstamp as possible. If tstamp is in
the past, it would should return an error.
Recent work by Takashi Iwai enables client code to set the clock type of
timestamps using snd_pcm_sw_params_set_tstamp_type(). This context could
quite naturally extend to tstamp argument of snd_pcm_start_at().
Before I get stuck into working up the details under the hood, it'd be
good to get some feedback/objections regarding this approach.

It's probably better idea to start PCM playback with a bunch of zeroes
and then rely on existing timestamping to insert samples at the right
location in the ring buffer - which you have to do anyway to compensate
for drifts between your network clock and audio clock. This is a more
predictable solution that abstracts away all the time needed to arm DMA,
FIFOs, etc. The only hardware-dependent variable that would remain is
the precision/granularity of the timestamping.
-Pierre

Thanks heaps for the feedback :)

Agreed - streaming zeros 'as required' is pretty much the obvious
solution when confined to the existing API.

So I guess I'll have to show a few more cards from my hand :D

The next iteration of our hardware platform will be accompanied by a
move to linux, which explains our interest in ALSA for sound delivery.

The pcm hardware for the new platform can start rendering when a compare
register matches a hw counter (driven by the audio clock). This allows
for starting with frame-accurate timing.

By adding another tstamp_type - let's call it
SND_PCM_TSTAMP_TYPE_HARDWARE for now - snd_pcm_start_at() could
internally delegate to the driver (if it supports it, otherwise error)

I take your point about drift, but we're approaching that as a related,
but orthogonal, problem. As a business we've decided that dropouts are
unacceptable. Our products have several audio clock sources available,
one of which is a high-quality VXCO. With an appropriate servo, it will
track a clock recovered from the incoming stream.

I'm still working through a couple of options for exposing the VXCO
audio clock to a userspace servo [The linux kernel has the PCH
framework, for example, which is almost enough, but doesn't dovetail
with ALSA in an obvious way].

In summary, we'd like to leverage our pcm hardware's ability to start on
time (hence the chat about snd_pcm_start_at), and we'd also like to
expose control of the pcm's audio clock in some way (WIP, suggestions
welcome!).

The clock control stuff is harder, but I wanted to get on the ml with
something simple, say hi, propose a thing

Thanks for reading - I'll have questions about clock control soon
enough, see ya then.

Cheers,
Tim

[Pierre-Louis Bossart]

Post by Tim Cussins
Hi Peirre,

Post by Pierre-Louis Bossart

Post by Tim Cussins
Hi all,
I'm Tim: I work at Linn Products Ltd - we make Network Music Players,
amongst other things.
As you might imagine, synchronised-start is important when multiple
devices on the network are rendering the same audio. We'd be interested
in contributing a small expansion of the alsa-lib API to support
synchronised start.
Assuming we can synchronise the audio clocks (I'm aware this is not
int snd_pcm_start_at(snd_pcm_t* pcm, snd_htimestamp_t* tstamp);
and playback would begin as close to tstamp as possible. If tstamp is in
the past, it would should return an error.
Recent work by Takashi Iwai enables client code to set the clock type of
timestamps using snd_pcm_sw_params_set_tstamp_type(). This context could
quite naturally extend to tstamp argument of snd_pcm_start_at().
Before I get stuck into working up the details under the hood, it'd be
good to get some feedback/objections regarding this approach.

It's probably better idea to start PCM playback with a bunch of zeroes
and then rely on existing timestamping to insert samples at the right
location in the ring buffer - which you have to do anyway to compensate
for drifts between your network clock and audio clock. This is a more
predictable solution that abstracts away all the time needed to arm DMA,
FIFOs, etc. The only hardware-dependent variable that would remain is
the precision/granularity of the timestamping.
-Pierre

Thanks heaps for the feedback :)
Agreed - streaming zeros 'as required' is pretty much the obvious
solution when confined to the existing API.
So I guess I'll have to show a few more cards from my hand :D
The next iteration of our hardware platform will be accompanied by a
move to linux, which explains our interest in ALSA for sound delivery.
The pcm hardware for the new platform can start rendering when a compare
register matches a hw counter (driven by the audio clock). This allows
for starting with frame-accurate timing.

Interesting. I wonder if you actually need a new extension for this, you
could write the timestamp in an ALSA control and implement your .trigger
function by using the contents of the control, i.e. delay the actual
start. it wouldn't be generic but your hardware isn't either.

Post by Tim Cussins
By adding another tstamp_type - let's call it
SND_PCM_TSTAMP_TYPE_HARDWARE for now - snd_pcm_start_at() could
internally delegate to the driver (if it supports it, otherwise error)
I take your point about drift, but we're approaching that as a related,
but orthogonal, problem. As a business we've decided that dropouts are
unacceptable. Our products have several audio clock sources available,
one of which is a high-quality VXCO. With an appropriate servo, it will
track a clock recovered from the incoming stream.

The drift control can be done with ASRC or tweaking the clock, either
way you need a servo loop based on timestamping info reported by the
hardware.

Post by Tim Cussins
I'm still working through a couple of options for exposing the VXCO
audio clock to a userspace servo [The linux kernel has the PCH
framework, for example, which is almost enough, but doesn't dovetail
with ALSA in an obvious way].

That part would benefit other implementations - whether they have a VCO
or can do ASRC with a fixed clock. Looking forward to your ideas.

Post by Tim Cussins
In summary, we'd like to leverage our pcm hardware's ability to start on
time (hence the chat about snd_pcm_start_at), and we'd also like to
expose control of the pcm's audio clock in some way (WIP, suggestions
welcome!).
The clock control stuff is harder, but I wanted to get on the ml with
something simple, say hi, propose a thing
Thanks for reading - I'll have questions about clock control soon
enough, see ya then.
Cheers,
Tim
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http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

[Takashi Iwai]

At Fri, 03 Oct 2014 17:24:22 -0500,

Post by Pierre-Louis Bossart

Post by Tim Cussins
Hi Peirre,

Post by Pierre-Louis Bossart

Post by Tim Cussins
Hi all,
I'm Tim: I work at Linn Products Ltd - we make Network Music Players,
amongst other things.
As you might imagine, synchronised-start is important when multiple
devices on the network are rendering the same audio. We'd be interested
in contributing a small expansion of the alsa-lib API to support
synchronised start.
Assuming we can synchronise the audio clocks (I'm aware this is not
int snd_pcm_start_at(snd_pcm_t* pcm, snd_htimestamp_t* tstamp);
and playback would begin as close to tstamp as possible. If tstamp is in
the past, it would should return an error.
Recent work by Takashi Iwai enables client code to set the clock type of
timestamps using snd_pcm_sw_params_set_tstamp_type(). This context could
quite naturally extend to tstamp argument of snd_pcm_start_at().
Before I get stuck into working up the details under the hood, it'd be
good to get some feedback/objections regarding this approach.

It's probably better idea to start PCM playback with a bunch of zeroes
and then rely on existing timestamping to insert samples at the right
location in the ring buffer - which you have to do anyway to compensate
for drifts between your network clock and audio clock. This is a more
predictable solution that abstracts away all the time needed to arm DMA,
FIFOs, etc. The only hardware-dependent variable that would remain is
the precision/granularity of the timestamping.
-Pierre

Thanks heaps for the feedback :)
Agreed - streaming zeros 'as required' is pretty much the obvious
solution when confined to the existing API.
So I guess I'll have to show a few more cards from my hand :D
The next iteration of our hardware platform will be accompanied by a
move to linux, which explains our interest in ALSA for sound delivery.
The pcm hardware for the new platform can start rendering when a compare
register matches a hw counter (driven by the audio clock). This allows
for starting with frame-accurate timing.

Interesting. I wonder if you actually need a new extension for this, you
could write the timestamp in an ALSA control and implement your .trigger
function by using the contents of the control, i.e. delay the actual
start. it wouldn't be generic but your hardware isn't either.

Well, your suggestion sounds really tricky. The trigger is supposed
to trigger the stream immediately, and the delay isn't considered
there in principle. The system can work with delays, but it's not in
a form of the initial design.

I think some synchronized triggering mechanism is missing in API, too.
There has been a similar request from others in the past (Digigram
wanted to have such a feature), so maybe it's not so uncommon
scenario.

This would be a good topic to be discussed in the upcoming audio
mini-summit, but both of you won't be there, right?


Takashi

[Tim Cussins]

Hi Takashi,

Post by Takashi Iwai
At Fri, 03 Oct 2014 17:24:22 -0500,

Post by Pierre-Louis Bossart

Post by Tim Cussins
Hi Peirre,
The pcm hardware for the new platform can start rendering when a compare
register matches a hw counter (driven by the audio clock). This allows
for starting with frame-accurate timing.

Interesting. I wonder if you actually need a new extension for this, you
could write the timestamp in an ALSA control and implement your .trigger
function by using the contents of the control, i.e. delay the actual
start. it wouldn't be generic but your hardware isn't either.

Well, your suggestion sounds really tricky. The trigger is supposed
to trigger the stream immediately, and the delay isn't considered
there in principle. The system can work with delays, but it's not in
a form of the initial design.
I think some synchronized triggering mechanism is missing in API, too.
There has been a similar request from others in the past (Digigram
wanted to have such a feature), so maybe it's not so uncommon
scenario.
This would be a good topic to be discussed in the upcoming audio
mini-summit, but both of you won't be there, right?

I didn't have any plans to be at the mini-summit, but Glasgow isn't too
far away...

Post by Takashi Iwai
Takashi

[Mark Brown]

Post by Takashi Iwai
I think some synchronized triggering mechanism is missing in API, too.
There has been a similar request from others in the past (Digigram
wanted to have such a feature), so maybe it's not so uncommon
scenario.

Yes, I've had people ask me about this too, mainly around getting
multiple DSP streams synchronized. It was a while ago and a bit edge
casey though.

[Takashi Iwai]

At Wed, 8 Oct 2014 15:18:32 +0100,

Post by Mark Brown

Post by Takashi Iwai
I think some synchronized triggering mechanism is missing in API, too.
There has been a similar request from others in the past (Digigram
wanted to have such a feature), so maybe it's not so uncommon
scenario.

Yes, I've had people ask me about this too, mainly around getting
multiple DSP streams synchronized. It was a while ago and a bit edge
casey though.

OK, I added it to the topic list of audio mini-summit.
Let's see whether we can talk something about this.


Takashi

[Tim Cussins]

Post by Takashi Iwai
At Wed, 8 Oct 2014 15:18:32 +0100,

Post by Mark Brown

Post by Takashi Iwai
I think some synchronized triggering mechanism is missing in API, too.
There has been a similar request from others in the past (Digigram
wanted to have such a feature), so maybe it's not so uncommon
scenario.

Yes, I've had people ask me about this too, mainly around getting
multiple DSP streams synchronized. It was a while ago and a bit edge
casey though.

OK, I added it to the topic list of audio mini-summit.
Let's see whether we can talk something about this.

Awesome. There's a couple of things I can do here:

1) Post my ideas for selecting/controlling VCO associated with PCM _soon_
2) Come over on Tuesday and join the mini summit
3) Both!

I'm pretty new to ALSA, so I can only realistically contribute to the
scheduled stream trigger discussion, although the Configuration/ASoC
stuff will be relevant to my crowd too.

If no-one has a major objection to a rep from the ALSA-newbie community,
I'll see if I can come over :)

Either way, I'll post my VCO ideas to the list ASAP.

Cheers,
Tim

[Mark Brown]

Post by Tim Cussins

Post by Takashi Iwai
OK, I added it to the topic list of audio mini-summit.
Let's see whether we can talk something about this.

2) Come over on Tuesday and join the mini summit

This would be great, but please note that since the Linux Foundation are
doing the logistics (room and so on) for us you'd need to register for
one of the conferences that week.

[Tim Cussins]

Post by Mark Brown

Post by Tim Cussins

Post by Takashi Iwai
OK, I added it to the topic list of audio mini-summit.
Let's see whether we can talk something about this.

2) Come over on Tuesday and join the mini summit

This would be great, but please note that since the Linux Foundation are
doing the logistics (room and so on) for us you'd need to register for
one of the conferences that week.

Righto. Registration has closed (all full I guess!) - I'll see if I can
find a (legitimate) way in :)

[Nick Stoughton]

Interestingly, I implemented this exact function (snd_pcm_start_at) about a
week ago before I saw this thread! We are still experimenting with it, but
I'd be happy to discuss it next week at the summit.

*______________________________*
*Nick Stoughton*
*Aether Things Inc *
*San Francisco*
[phone]

Post by Tim Cussins

Post by Takashi Iwai
OK, I added it to the topic list of audio mini-summit.

Post by Takashi Iwai
Let's see whether we can talk something about this.

2) Come over on Tuesday and join the mini summit
This would be great, but please note that since the Linux Foundation are
doing the logistics (room and so on) for us you'd need to register for
one of the conferences that week.

Righto. Registration has closed (all full I guess!) - I'll see if I can
find a (legitimate) way in :)
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