Hi, 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: http://0pointer.de/blog/projects/pulse-glitch-free.html _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
