On Mon, 07.01.08 19:33, Jaroslav Kysela (perex@perex.cz) wrote:
I assume that I can enable a mode like that with one of the SW params. But quite frankly the docs for it are not enlighening at all.
Set the stop_threshould sw_params to the boundary size.
snd_pcm_sw_params_get_boundary(sw_params, &boundary); snd_pcm_sw_params_set_stop_threshold(pcm, sw_params, boundary);
then the driver behaves in the "freewheel" mode. The dmix plugin uses this technique.
That's not what I was looking for. This will only disable automatic stopping on buffer underrun. I am using that already in PA (however I pass -1 as stop threshold, which should work, too, shouldn't it?)
What I am really looking for is a way to disable that ALSA reports via poll() the buffer fill level, but instead only reports whether an interrupt happened.
Note that you can control fill level using snd_pcm_forward/rewind without any R/W calls (of course if supported in whole chain). ALSA supports only "controlled" I/O not dumb I/O as OSS driver for mmap.
If you look for timing source, use timer API - you may try alsa-lib/test/timer.c:
./timer class=3 card=0 device=0 subdevice=0
How does this timer depend on the PCM clock? Is its wakeup granularity dependant on the period parameters of the matching PCM device? Or am I supposed to first initialize PCM, and chose some period parameters the hw likes and than pass that on to the timer subsystem?
I assume I don't have any guarantee that all alsa devices have such a timer attached? So I'd need some major non-trivial fallback code if I make use of these timers?
The background why I want this is this: As mentioned I am now scheduling audio in PA mostly based on system timers. To be able to do that I need to be able to translate timespans from the sound card clock to the system clock. Which requires me to get the sample time from the sound card from time to time and filter it through some code that estimates how the sound card clock and the system clock deviates. I'd prefer to do that only once or maybe twice everytime the playback buffer is fully played, and only shortly after an IRQ happened, under the assumption that this is the best time to get the most accurate timing information from the sound card.
It's not really necessary. You can use only one timing source (system timer) and use position timestamps to do corrections.
Position timestamps? You mean status->tstamp, right? I'd like to use that. But this still has two problems:
1) As mentioned, CLOCK_MONOTONIC support is still missing in ALSA(-lib)
2) I'd like to correct my estimations as quickly as possible, i.e. as soon as a new update is available, and not only when I ask for it. So basically, I want to be able to sleep in a poll() for timing updates.
But your example does not explain, why you don't move r/w pointer in the ring buffer (use mmap_commit), thus why you don't fullfill the avail_min requirement for poll wakeup. It seems to me that you're trying to do some crazy things with the ring buffer which are not allowed.
As mentioned, when PA starts up it configures the audio hw buffer to 2s or so with the minimal number of periods (2 on my sound cards). Then, clients come and go. Depending on the what the minimal latency constraints of the clients are, I however will only fill up part of the buffer.
Scenario #1:
Only one simple MP3 playing music application is connected. It doesn't have any real latency constraints. We always fill up the whole 2s buffer, then sleep for 1990 ms, and then fill it up again, and so on. If the MP3 player pauses or seeks, we rewrite the audio buffer with _rewind(). Thus alsthough we buffer two full seconds the user interfaces still reacts snappy.
Now, because the user starts and stops applications all the time, we dynamically change into scenario #2:
The MP3 playing application is still running. However, now a VoIP application is running too. It wants a worst case latency of let's say 20ms. When this applications starts up we don't want to interrupt playback of the MP3 application. So from now on we only use 20ms of the previously configured 2s hw buffer. And as soon as we wrote 20ms, we sleep for 10ms, and then fill it up again, and so on.
Now, after a while the VoIP call is over, we enter scenario #3:
This is identical to #1, we again use the full 2s hw buffer, and sleep for 1990ms.
So, depending on what clients are connected, we dynamically change the wakeups. Now, on ALSA (and with a lot of sound hw, as I understood Takashi) you cannot reconfigure the period sizes dynamically without interruptions of audio output. That's why I want to disable the whole period/buffer fill level management of ALSA, and do all myself with system timers, which I thankfully now can due to the advent of hrtimers (at least on Linux/x86). System timers nowadays are a lot more flexibe than the PCM timer, because they can be reconfigured all the time without any drawbacks. They are not dependant on period sizes or other stuff which may only be reconfigured by resetting the audio devices. The only drawback is that we need to determinine how the sound card clock and the system clock deviate.
Does that make sense to you?
Lennart