On Sat, Aug 01, 2015 at 11:29:05AM +1000, Peter Ross 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 > > <mich...@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 <itumas...@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 <itumas...@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,
i have nothing against it being in swr, i also have nothing against it being in the codec, that is iam fine with either. if some people expressed a preferrance previously maybe you should ask them directly to ensure you dont spend time updating a patch that someone then objects against [...] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB No human being will ever know the Truth, for even if they happen to say it by chance, they would not even known they had done so. -- Xenophanes
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