>> The correct way is to use snd_pcm_get_params() to get buffer_size and > period_size > > *snd_pcm_set_params() has limitation > > The most common sound card (e.g. HDA ) has the following constraints > on period bytes and buffer bytes > > snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, > 128); > snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, > 128); I am probably not good enough to understand these changes =( Pretty hard core stuff =) > you cannot get exactly 0.5 second period time when rate is multiple of 3 ( > e.g. 48000 Hz ) Does this mean, that I shouldn't give snd_pcm_writei() the number of all the frames in the buffer, but only sample_rate * latency = frames ? So if I e.g. have: sample_rate = 44100 latency = 0.5 [s] all_frames = 100000 The number of frames that I should give to snd_pcm_writei() would be sample_rate * latency = frames 44100*0.5 = 22050 and the number of iterations the for-loop should be?: (int) 100000/22050 = 4; with frames=22050 and one extra, but only with 100000 mod 22050 = 11800 frames? _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
