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

[Michael Niedermayer]

On Sat, May 03, 2014 at 11:15:41PM +1000, Peter Ross wrote:
> 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.

yes, some downsample to sample rate, this could also be usefull for
other formats in theory, 96khz aac decoded into 48khz pcm by skiping
the higher frequency components or by skiping SBR

[...]

-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

No great genius has ever existed without some touch of madness. -- Aristotle
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