On 06.04.2013 19:52, Jussi Laako wrote: >> All audio formats are specified in *bytes* for sample and buffer sizes, >> so we will also keep it that way for DSD. > > Umm, IIRC, all the ALSA PCM API snd_pcm_*_period_size() stuff is in > number of frames? (number of samples per channel) I was talking about the SND_PCM_FORMAT_ definitions that I augmented. What they denote is bytes per sample frame, not bits. > Thus size of the > buffer is period size * sample size * channels (* nperiods). So if we > would use 2.8 MHz as sampling rate for DSD, then buffer size would be > period size * 1/8 * channels, since one sample of DSD is eight' of a byte. No, you would choose 1/8 of the actual sample rate, because the actual transport goes in multiple of 8 bits. > And sampling rate is number of frames per second. So it is quite clear > and non-confusing. ... and one frame is also just 8 bits (or 16, for the second format). I think that matches the logic quite well. Daniel _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
