2010/4/26 Stefan Schoenleitner <dev.c0debabe@xxxxxxxxx> > Raymond Yau wrote: > > Are you sure that you really need pulseaudio since your request latency > is > > quite low ? > > You mean the small period_size of 160 frames or rather the buffer size > of 2*period_size? > > In fact at 8kHz sampling rate a period_size of 160 equals a full 20ms of > sound: > > duration of one frame: 1000ms / 8000Hz = 0.125ms = 125 us > 160 frames * 125us = 20000us = 20ms > > This is by far more than a large period size holds at a higher sampling > rate (e.g. 44.1 kHz). > But the period time of your sound card is 40.634ms which is more than double of your requested 20ms D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3584 D: alsa-util.c: period_size : 1792 D: alsa-util.c: period_time : 40634 > > To the question whether pulseaudio is needed: > On the embedded target PA will not be used, but since development takes > place on a PC, I need a soundcard that supports the mentioned audio > format contraints. > And at the moment this is pulseaudio which is the reason why I really > need it for application development. > > > However, we should try to not drift away from the actual problem which > is that *poll() returns an event even if far less than avail_min frames > are available*. > > > > You have to ask PA developer whether PA support such low latecny ? > > > > PA (tsched=0 ) configure your sound card 1792 frames per period but your > > application request for 160 frames per period > > > > You will need the PA expert to answer how PA server capture 1792 frames > from > > sound card and send it to your application > > > > : module-alsa-source.c: Using 2 fragments of size 7168 bytes, buffer > > time is 81.27ms > > > > D: alsa-util.c: Its setup is: > > D: alsa-util.c: stream : CAPTURE > > D: alsa-util.c: access : MMAP_INTERLEAVED > > D: alsa-util.c: format : S16_LE > > D: alsa-util.c: subformat : STD > > D: alsa-util.c: channels : 2 > > D: alsa-util.c: rate : 44100 > > D: alsa-util.c: exact rate : 44100 (44100/1) > > D: alsa-util.c: msbits : 16 > > D: alsa-util.c: buffer_size : 3584 > > D: alsa-util.c: period_size : 1792 > > D: alsa-util.c: period_time : 40634 > > D: alsa-util.c: tstamp_mode : ENABLE > > D: alsa-util.c: period_step : 1 > > D: alsa-util.c: avail_min : 1792 > > D: alsa-util.c: period_event : 0 > > D: alsa-util.c: start_threshold : -1 > > D: alsa-util.c: stop_threshold : 8070450532247928832 > > D: alsa-util.c: silence_threshold: 0 > > D: alsa-util.c: silence_size : 0 > > D: alsa-util.c: boundary : 8070450532247928832 > > This is how pulseaudio works *internally*. > Hence it opens my hardware sound card "hw" with the above format. > If sound it recorded/played back at a different sampling rate, PA > converts it. > If you play a prerecorded audio file that has been recorded at a > different sampling rate, you can see that behavior. > After the sound driver capture 40.634 ms of audio , PA have to convert the 44100Hz stereo to 8000Hz mono seem just add left + right to mono without halve the sum , you may hear clipping if you are using line in instead of mic or PA clamp the output D: resampler.c: Channel matrix: D: resampler.c: I00 I01 D: resampler.c: +------------ D: resampler.c: O00 | 1.000 1.000 > What happens is: > > [sound application (e.g.. aplay)] ---(audio format of sound file)---> [ > PA plugin] --> [PA daemon] ---(audio format of PA)--> [soundcard] > > cheers, > stefan > _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
