[alsa-devel] proposal: snd_pcm_start_at()
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
On 10/2/14, 9:34 AM, Tim Cussins wrote:
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.
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
Hi Peirre,
On 02/10/14 18:41, Pierre-Louis Bossart wrote:
On 10/2/14, 9:34 AM, Tim Cussins wrote:
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.
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
On 10/3/14, 7:00 AM, Tim Cussins wrote:
Hi Peirre,
On 02/10/14 18:41, Pierre-Louis Bossart wrote:
On 10/2/14, 9:34 AM, Tim Cussins wrote:
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.
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.
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.
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.
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 _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
At Fri, 03 Oct 2014 17:24:22 -0500, Pierre-Louis Bossart wrote:
On 10/3/14, 7:00 AM, Tim Cussins wrote:
Hi Peirre,
On 02/10/14 18:41, Pierre-Louis Bossart wrote:
On 10/2/14, 9:34 AM, Tim Cussins wrote:
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.
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
Hi Takashi,
On 06/10/14 10:45, Takashi Iwai wrote:
At Fri, 03 Oct 2014 17:24:22 -0500, Pierre-Louis Bossart wrote:
On 10/3/14, 7:00 AM, Tim Cussins wrote:
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...
Takashi
On Mon, Oct 06, 2014 at 11:45:57AM +0200, Takashi Iwai wrote:
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.
At Wed, 8 Oct 2014 15:18:32 +0100, Mark Brown wrote:
On Mon, Oct 06, 2014 at 11:45:57AM +0200, Takashi Iwai wrote:
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
On 08/10/14 16:29, Takashi Iwai wrote:
At Wed, 8 Oct 2014 15:18:32 +0100, Mark Brown wrote:
On Mon, Oct 06, 2014 at 11:45:57AM +0200, Takashi Iwai wrote:
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
On Wed, Oct 08, 2014 at 05:09:04PM +0100, Tim Cussins wrote:
On 08/10/14 16:29, Takashi Iwai wrote:
OK, I added it to the topic list of audio mini-summit. Let's see whether we can talk something about this.
- 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.
On 08/10/14 21:16, Mark Brown wrote:
On Wed, Oct 08, 2014 at 05:09:04PM +0100, Tim Cussins wrote:
On 08/10/14 16:29, Takashi Iwai wrote:
OK, I added it to the topic list of audio mini-summit. Let's see whether we can talk something about this.
- 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 :)
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* +1 (510) 388 1413
On Thu, Oct 9, 2014 at 2:20 AM, Tim Cussins timcussins@eml.cc wrote:
On 08/10/14 21:16, Mark Brown wrote:
On Wed, Oct 08, 2014 at 05:09:04PM +0100, Tim Cussins wrote:
On 08/10/14 16:29, Takashi Iwai wrote:
OK, I added it to the topic list of audio mini-summit.
Let's see whether we can talk something about this.
- 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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participants (5)
-
Mark Brown -
Nick Stoughton -
Pierre-Louis Bossart -
Takashi Iwai -
Tim Cussins