On Wed, 13 May 2009, Jon Smirl wrote:
There's a long thread over on the pulse list about glitch free playback. The glitches they are encountering are caused by CPU scheduling latency. They are trying to fix this by setting HZ up to 1000 and constantly polling the audio DMA queue to keep it 99% full.
This doesn't seem like the right solution to me. It is fixing the symptom not the cause. The cause is 200-300ms scheduling latency. The source of that needs to be tracked down and fixed in the kernel. But we have to live with the latencies until they are fixed.
The strategy of checking the queue at 1000Hz works but it is very inefficient. The underlying problem is that the buffer ALSA is using is too small on systems with 300ms latency. The buffer is just big enough to cover 300ms so they rapidly check and fill it at 1000Hz to ensure that it is full if the 300ms latency strikes.
??? The ring buffer size is not limited if hw allows that.
On my hardware with period interrupts ALSA is only checking the buffer at 8Hz. Since I'm checking appl_ptr I know when DMA over runs the buffers. This allows me to insert silence and I could indicated this
Inserting silence might be wrong, if you broke stream timing. The elapsed() callback should be called at exact timing (and position should be updated, too).
condition to ALSA if there was a mechanism for doing so. ALSA could use this over run knowledge to measure scheduling latency and adjust the buffering.
But the DMA interface between ALSA and the driver has been fixed at stream creation time. There's no way to dynamically alter it (like window size changes in TCP/IP). With networking you get a list of buffers to send. As you send these buffers you mark them sent. The core is free to hand you buffers straight from user space or do copies and use internal ring buffers. The network driver just gets a list of physical addresses to send. This buffer bookkeeping could occur in snd_pcm_period_elapsed().
A dynamic chaining mechanism allows you to alter the buffering mid-stream. If the driver indication a DMA over run error this tells ALSA that it needs to insert another buffer. After a while these errors will stop and ALSA will have measured worst case CPU scheduling latency. From then on it will know the exact size of buffering needed for the kernel it it running on and it can use this knowledge at stream creation time. Now filling the buffer at 8Hz or lower will work and you don't have to spend the power associated with 1000Hz timer interrupts.
Nothing prevents to application to allocate a big ring buffer and write samples only as necessary. Application is a producer and controller in this case. The midlevel layer can hardly do something if samples are not available. The situation will be more or less bad.
The whole problem is that standard Linux kernel is not realtime, but audio is realtime task.
The result is to set optimal ring buffer parameters in the application.
Jaroslav
----- Jaroslav Kysela perex@perex.cz Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.