[FFmpeg-devel] [PATCH 2/2] libavformat/dsfdec: Error dsf_read_header fixed.

[Ganesh Ajjanagadde]

On Fri, Jul 31, 2015 at 9:29 PM, Peter Ross <pross at xvid.org> wrote:
> On Fri, Jul 31, 2015 at 08:56:49PM -0400, Ganesh Ajjanagadde wrote:
>> On Fri, Jul 31, 2015 at 8:08 PM, Michael Niedermayer
>> <michael at niedermayer.cc> wrote:
>> > On Fri, Jul 31, 2015 at 07:33:01PM -0400, Ganesh Ajjanagadde wrote:
>> >> On Fri, Jul 31, 2015 at 7:01 PM, Ihar A. Tumashyk <itumashyk at gmail.com> wrote:
>> >> > Sample rate is written "as is" in header . Is *should not* be devined by
>> >> > 8. Refer spec.:
>> >> > http://dsd-guide.com/sites/default/files/white-
>> >> > papers/DSFFileFormatSpec_E.pdf
>> >> >
>> >> > After this patch ffprobe will corretly show sample rate for DSF files.
>> >> >
>> >> > Signed-off-by: Ihar A. Tumashyk <itumashyk at gmail.com>
>> >> > ---
>> >> >  libavformat/dsfdec.c | 2 +-
>> >> >  1 file changed, 1 insertion(+), 1 deletion(-)
>> >> >
>> >> > diff --git a/libavformat/dsfdec.c b/libavformat/dsfdec.c
>> >> > index ae198b2..3e162ae 100644
>> >> > --- a/libavformat/dsfdec.c
>> >> > +++ b/libavformat/dsfdec.c
>> >> > @@ -105,7 +105,7 @@ static int dsf_read_header(AVFormatContext *s)
>> >> >
>> >> >      st->codec->codec_type   = AVMEDIA_TYPE_AUDIO;
>> >> >      st->codec->channels     = avio_rl32(pb);
>> >> > -    st->codec->sample_rate  = avio_rl32(pb) / 8;
>> >> > +    st->codec->sample_rate  = avio_rl32(pb);
>> >> >
>> >> >      switch(avio_rl32(pb)) {
>> >> >      case 1: st->codec->codec_id = AV_CODEC_ID_DSD_LSBF_PLANAR; break;
>> >>
>> >> LGTM, thanks for clarifying intent of previous patch.
>> >> I am a little puzzled as to why this was never caught till now,
>> >
>> > Changing the sample rate like in the patch breaks playback (it plays
>> > much too quick)
>> > you can find samples to test at:
>> > http://www.2l.no/hires/
>>
>> I really need to stop trusting these spec files that much,
>> they are often incomplete as in this case. Ironically,
>> wikipedia here gives a much better description of the format:
>> https://en.wikipedia.org/wiki/Direct_Stream_Digital,
>> where they mention that essentially this signal is oversampled by a factor of 8.
>> As the wiki entry indicates, max frequency of PCM is 352.8 kHz.
>>
>> In fact, I think previous patch should be reverted:
>> the wikipedia article says that newer format is called "DSD-wide" and
>> uses 8-bit word length.
>> ffprobe yields the downsampled frequency of 352.8 kHz which is what
>> actually gets used when one demuxes/
>> operates on the data per wikipedia.
>> Thus, I believe previous behavior was correct.
>
> converting between DSD->PCM reduces the sample rate by 8.
> at present, the DSD->PCM conversion happens libavcodec/dsddec.c.
> however, a 'codec' cannot change sample_rate, so the divide by 8 is performed earlier in libavformat.
>
> yes, this is a hack.
>
> i have a patch set that moves DSD->PCM conversion into swresample, where it belongs,
> along with adding a DST decoder and support for additional file formats.
> If anyone is interested, I can repost.

Was there a reason why previous patch set was ignored or not applied?
(Relevant mailing list link would help to get a sense for it).
At the minimum though, this hack should have relevant comments for it
in the source code.
This would have removed some of the confusion.

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