On Thu, Feb 26, 2015 at 02:14:33PM +0100, Tobias Rapp wrote: > On 26.02.2015 00:51, Michael Niedermayer wrote: > >On Wed, Feb 25, 2015 at 05:35:33PM +0100, Tobias Rapp wrote: > >> avienc.c | 9 ++++++++- > >> 1 file changed, 8 insertions(+), 1 deletion(-) > >>09f8c8250ce5ec1bdad79a1bf280028c9d3af376 > >>0001-libavformat-avienc-Fix-duration-for-audio-stream-Ope.patch > >> From 3a8e15598fa3e044f3ef65b5063c633cb4b3afed Mon Sep 17 00:00:00 2001 > >>From: Tobias Rapp <t.r...@noa-audio.com> > >>Date: Wed, 25 Feb 2015 17:10:13 +0100 > >>Subject: [PATCH] libavformat/avienc: Fix duration for audio stream OpenDML > >> super index > >> > >>Fixes the duration field of the OpenDML super index "indx" chunk to > >>contain the number of samples instead of the number of packets for > >>(linear/PCM) audio streams. > >> > >>This matches the OpenDML V1.02 standard text which states that the > >>duration field shall contain "time span in stream ticks". > >>--- > >> libavformat/avienc.c | 9 ++++++++- > >> 1 file changed, 8 insertions(+), 1 deletion(-) > >> > >>diff --git a/libavformat/avienc.c b/libavformat/avienc.c > >>index 53c2fe7..b77b295 100644 > >>--- a/libavformat/avienc.c > >>+++ b/libavformat/avienc.c > >>@@ -49,6 +49,7 @@ typedef struct AVIIentry { > >> > >> typedef struct AVIIndex { > >> int64_t indx_start; > >>+ int64_t audio_strm_offset; > >> int entry; > >> int ents_allocated; > >> AVIIentry** cluster; > >>@@ -91,6 +92,7 @@ static int64_t avi_start_new_riff(AVFormatContext *s, > >>AVIOContext *pb, > >> avi->riff_id++; > >> for (i = 0; i < s->nb_streams; i++) { > >> AVIStream *avist = s->streams[i]->priv_data; > >>+ avist->indexes.audio_strm_offset = avist->audio_strm_length; > >> avist->indexes.entry = 0; > >> } > >> > >>@@ -476,6 +478,7 @@ static int avi_write_ix(AVFormatContext *s) > >> for (i = 0; i < s->nb_streams; i++) { > >> AVIStream *avist = s->streams[i]->priv_data; > >> int64_t ix, pos; > >>+ int au_byterate, au_ssize, au_scale; > >> > >> avi_stream2fourcc(tag, i, s->streams[i]->codec->codec_type); > >> ix_tag[3] = '0' + i; > >>@@ -511,7 +514,11 @@ static int avi_write_ix(AVFormatContext *s) > >> avio_skip(pb, 16 * avi->riff_id); > >> avio_wl64(pb, ix); /* qwOffset */ > >> avio_wl32(pb, pos - ix); /* dwSize */ > >>- avio_wl32(pb, avist->indexes.entry); /* dwDuration */ > >>+ ff_parse_specific_params(s->streams[i], &au_byterate, &au_ssize, > >>&au_scale); > >>+ if (au_ssize == 0) > >>+ avio_wl32(pb, avist->indexes.entry); /* dwDuration (packet > >>count) */ > >>+ else > > > >>+ avio_wl32(pb, (avist->audio_strm_length - > >>avist->indexes.audio_strm_offset) / au_ssize); /* dwDuration (sample > >>count) */ > > > >the rounding here is wrong > >the index to be useable needs to have segments duration summable > >as is this would lead to significangt errors if many of these > >durations would be summed up > >something like > >avist->audio_strm_length / au_ssize - avist->indexes.audio_strm_offset / > >au_ssize > >might avoid that but ive not tested it > > My assumption was that a stream packet contains only whole samples > for linear/uncompressed audio (PCM) and that only for linear audio > the expression "au_ssize > 0" evaluates to TRUE.
We try to use au_ssize = 0 for compressed audio but i think it can be non zero in some stream copy cases, iam not 100% sure though > > If that is not the case I can do the rounding as suggested or maybe > introduce some separate stream duration counter based on > AVPacket.duration? i think best to keep it simple and maybe print a request for a sample/ testcase if some combination of parameters is found which we are unsure about if it works correctly [...] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB Many things microsoft did are stupid, but not doing something just because microsoft did it is even more stupid. If everything ms did were stupid they would be bankrupt already.
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