At Fri, 22 May 2009 09:58:29 -0400, Jon Smirl wrote:
On Fri, May 22, 2009 at 3:14 AM, Takashi Iwai tiwai@suse.de wrote:
At Fri, 22 May 2009 09:07:43 +0200 (CEST), Jaroslav Kysela wrote:
On Fri, 22 May 2009, Takashi Iwai wrote:
At Thu, 21 May 2009 14:44:27 -0400, Jon Smirl wrote:
I've been reading Lennart's article on glitch free audio....
On my scatter/gather DMA hardware I generate an interrupt at the end of each period. On the last period this causes ALSA to issue a trigger(STOP). However, there is a latency from when that interrupt is generated and when ALSA does the trigger(STOP). During that latency my hardware starts playing the next buffer (which contains stale data). Playing stale data causes a noise burst. I fixed this by programming my DMA hardware to not overrun appl_ptr.
There's another solution. ALSA already knows the stream ends mid-period and pads out the partial period with silence. If it simply padded out the next period with silence too the overrun problem would become non-audible and there would time to process trigger(STOP).
If padding out two periods at end of stream is possible, it then becomes possible for me to turn off interrupts in my audio driver.
This can be another interesting feature. For example, you could add a soft-mute padding to avoid a click noise at the end instead of a whole silence.
But, this kind of hack really depends on the hardware implementation. If a hardware has no DMA but require explicit writes, adding two periods is too much. Just a few samples would be enough.
I agree to leave such things to lowlevel drivers.
The core problem is that the lowlevel driver does not have sufficient information to know where the end of stream is.
Yes. In other words, it's because of the current code flow in ISR as I pointed. Issuing trigger(STOP) after queuing is simply stupid, if we know beforehand that the stream will be stopped anyway...
Takashi