From mboxrd@z Thu Jan  1 00:00:00 1970
From: Daniel Mack <zonque@gmail.com>
Subject: Re: [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in
 sync with the kernel list
Date: Sat, 06 Apr 2013 19:58:34 +0200
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To: Jussi Laako <jussi@sonarnerd.net>
Cc: alsa-devel@alsa-project.org, tiwai@suse.de, clemens@ladisch.de, demian@auraliti.com, ray@auraliti.com, Andreas Koch <andreas@akdesigninc.com>
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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