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) 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.
And sampling rate is number of frames per second. So it is quite clear and non-confusing.
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