[FFmpeg-devel] DSD and supporting codec-requested sample rate change

[Peter Ross]

On Mon, Apr 28, 2014 at 10:21:48PM +0200, Michael Niedermayer wrote:
> On Sat, Apr 26, 2014 at 04:42:27AM +0200, Peter Ross wrote:
> > Quoting Michael Niedermayer <michaelni at gmx.at>:
> > 
> > >On Mon, Apr 21, 2014 at 04:15:09PM +1000, Peter Ross wrote:
> > >>Hi,
> > >>
> > >>Direct Stream Digital (DSD) is the name given to the 1-bit delta
> > >>sigma encoding
> > >>system found in audiophile equipment. It is an alternative to PCM.
> > >>
> > [..]

> there is some similar mess with AAC-SBR where a incorrect sample
> rate is stored

thanks for that example.

> about the 24bit 176.4kHz, can this be played by normal PCM decoders?
> that is, do the 8 or less high bits represent a downsampled version?

no. the marker alternates between 0x05 and 0xFA.
"This has been chosen to minimize the click that might be experienced when
the receiving hardware misinterpretes the data as PCM when it really is DSD.
If this should happen it would create a tone around 88kHz and roughly -34db,
nothing harmful and something that most D/A converters would suppress to some
degree before it even reaches the loudspeaker." -- specification

> also what about simply considering that 24bit encoding to be a
> opaque encoding like any other encoder produces ? 
> there would be no sample rate on the output just a packet/frame rate

surely libavdevice will want to know the playback sample_rate.

-- Peter
(A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
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