[FFmpeg-devel] DSD and supporting codec-requested sample rate change
michaelni at gmx.at
Mon Apr 21 23:34:35 CEST 2014
On Mon, Apr 21, 2014 at 04:15:09PM +1000, Peter Ross wrote:
> 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.
> A DSD decoder was added recently that transforms an input DSD bitstream into
> floating point PCM. While this is sufficient to listen to content on a computer,
> it doesn't permit direct manipulation of the DSD samples. Nor does it let FFmpeg
> feed DSD samples into an external DAC.
> I have started modifying FFmpeg to better support DSD, but have hit problems with
> sample rate changes. Potentially some API changes to libswresample is required.
> Ideas or comments welcome.
> EFFORT SO FAR:
> 1. add AV_SAMPLE_FMT_DSD (and the planar variant).
> 2. incorporate DSD->PCM conversion into libswresample.
> 3. rework the dsd_msbf, dsd_lsbf, etc decoders to output AV_SAMPLE_FMT_DSD.
> (very similar in function to the libavcodec/pcm.c decoders)
> 4. DSD-over-PCM (DoP) encoder
> DoP is a method of storing DSD bits into a 24-bit PCM stream. The high 8-bits
> of each PCM sample is marked with a pattern, and the lower 16-bits are DSD samples.
> External DACs are able to detect the pattern and process the raw DSD samples.
> 5. DST (Direct Stream Transfer) decoder
> Lossless compression for DSD samples. Described in MPEG-4 Part 3, Subpart 10.
> 1. DSD->PCM conversion results in a change to the sample rate. For example,
> 2.8224 MHz 1-bit DSD samples => 352.8 kHz PCM samples; and
> 5.6448 MHz 1-bit DSD samples => 705.6 kHz PCM samples.
> a) Should libswresample return the new sample rate?
id tend to suggest that the sample rates should represent the true
(in a mathematical sense) samplerates of the signals.
And i realize that this definition is a bit fuzzy without a definition
of what a sample is.
> b) Should the user of libswresample be expected to know the change between
> DSD->PCM and populate the 'in_sample_rate' and 'out_sample_rate' accordingly?
id suggest that it should be handled like any other sample format
but iam not working on this, you do, so there may be better options
that iam not recognizing ...
> 2. DoP encoding takes DSD 1-bit samples @ 2.8224 MHz and smushes them into 24-bit
> PCM @ 176.4 kHz. How can the DoP codec inform avcodec that the output packets have
> a different sample rate? Editing 'avctx->sample_rate' doesnt work because other
> modules expect this information to remain static.
iam not sure i understand the problem
where does mhz DSD -> khz PCM differ from mhz PCM -> khz PCM ?
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
If you think the mosad wants you dead since a long time then you are either
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