[alsa-devel] safe support for rewind in ALSA

Kai Vehmanen kvehmanen at eca.cx
Tue Feb 9 21:54:31 CET 2010


On Tue, 9 Feb 2010, Raymond Yau wrote:

>>>  - at T3, application calls snd_pcm_delay() to query how many samples
>>>   of delay there is currently (e.g. if it write a sample to ALSA
>>>   PCM device now, how long before it hits the speaker)
> No, your assumption (how long before it hits the speaker ) is wrong , refer
> to
> http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html

I have trouble following your logic here. I mean:

> For playback the delay is defined as the time that a frame that is written
> to the PCM stream shortly after this call will take to be actually audible.
> It is as such the overall latency from the write call to the final DAC.

Isn't that _exactly_ what I wrote above?

I.e. this is the purpose of snd_pcm_delay() and why application developers 
use it for e.g. audio/video sync. So what's the difference? Do you mean 
that speaker!=DAC, or...?

> why do PA insist to use one period per buffer when only those ISA drivers
> and intel8x0 have periods_min =1 , the most common HDA driver and most sound
> cards have periods_min =2 ?

That is discussed at length here:

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