| Message ID | 20211127214551.22949-2-jpstewart@personalprojects.net |
|---|---|
| State | Accepted |
| Commit | 6c76b6392348460472f0b6deac4d0a161109d498 |
| Headers | show |
| Series | [FFmpeg-devel,v2,1/2] avformat/mvdec: fix reading number of audio channels | expand |
| Context | Check | Description |
|---|---|---|
| andriy/make_x86 | success | Make finished |
| andriy/make_fate_x86 | success | Make fate finished |
| andriy/make_ppc | success | Make finished |
| andriy/make_fate_ppc | success | Make fate finished |
On Sat, Nov 27, 2021 at 04:45:51PM -0500, John-Paul Stewart wrote: > Adds support for reading audio sample size from the data instead of > assuming all audio is 16 bits per sample. > --- > libavformat/mvdec.c | 18 +++++++++++++++--- > 1 file changed, 15 insertions(+), 3 deletions(-) > > diff --git a/libavformat/mvdec.c b/libavformat/mvdec.c > index 8492928820..8b54a9ab04 100644 > --- a/libavformat/mvdec.c > +++ b/libavformat/mvdec.c > @@ -299,6 +299,8 @@ static int mv_read_header(AVFormatContext *avctx) > if (version == 2) { > uint64_t timestamp; > int v; > + uint32_t bytes_per_sample; > + > avio_skip(pb, 22); > > /* allocate audio track first to prevent unnecessary seeking > @@ -341,11 +343,21 @@ static int mv_read_header(AVFormatContext *avctx) > } > avpriv_set_pts_info(ast, 33, 1, ast->codecpar->sample_rate); > > - avio_skip(pb, 4); > + bytes_per_sample = avio_rb32(pb); > > v = avio_rb32(pb); > if (v == AUDIO_FORMAT_SIGNED) { > - ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE; > + switch (bytes_per_sample) { > + case 1: > + ast->codecpar->codec_id = AV_CODEC_ID_PCM_S8; > + break; > + case 2: > + ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE; > + break; > + default: > + avpriv_request_sample(avctx, "Audio sample size %i bytes", bytes_per_sample); > + break; > + } > } else { > avpriv_request_sample(avctx, "Audio compression (format %i)", v); > } > @@ -369,7 +381,7 @@ static int mv_read_header(AVFormatContext *avctx) > avio_skip(pb, 8); > av_add_index_entry(ast, pos, timestamp, asize, 0, AVINDEX_KEYFRAME); > av_add_index_entry(vst, pos + asize, i, vsize, 0, AVINDEX_KEYFRAME); > - timestamp += asize / (ast->codecpar->channels * 2LL); > + timestamp += asize / (ast->codecpar->channels * bytes_per_sample); > } > } else if (!version && avio_rb16(pb) == 3) { > avio_skip(pb, 4); > -- > 2.33.0 thanks for doing this. both patches look good. i will apply in a few days. -- Peter (A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
On Wed, Dec 01, 2021 at 04:18:42PM +1100, Peter Ross wrote: > On Sat, Nov 27, 2021 at 04:45:51PM -0500, John-Paul Stewart wrote: > > Adds support for reading audio sample size from the data instead of > > assuming all audio is 16 bits per sample. > > --- > > libavformat/mvdec.c | 18 +++++++++++++++--- > > 1 file changed, 15 insertions(+), 3 deletions(-) > > > > diff --git a/libavformat/mvdec.c b/libavformat/mvdec.c > > index 8492928820..8b54a9ab04 100644 > > --- a/libavformat/mvdec.c > > +++ b/libavformat/mvdec.c > > @@ -299,6 +299,8 @@ static int mv_read_header(AVFormatContext *avctx) > > if (version == 2) { > > uint64_t timestamp; > > int v; > > + uint32_t bytes_per_sample; > > + > > avio_skip(pb, 22); > > > > /* allocate audio track first to prevent unnecessary seeking > > @@ -341,11 +343,21 @@ static int mv_read_header(AVFormatContext *avctx) > > } > > avpriv_set_pts_info(ast, 33, 1, ast->codecpar->sample_rate); > > > > - avio_skip(pb, 4); > > + bytes_per_sample = avio_rb32(pb); > > > > v = avio_rb32(pb); > > if (v == AUDIO_FORMAT_SIGNED) { > > - ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE; > > + switch (bytes_per_sample) { > > + case 1: > > + ast->codecpar->codec_id = AV_CODEC_ID_PCM_S8; > > + break; > > + case 2: > > + ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE; > > + break; > > + default: > > + avpriv_request_sample(avctx, "Audio sample size %i bytes", bytes_per_sample); > > + break; > > + } > > } else { > > avpriv_request_sample(avctx, "Audio compression (format %i)", v); > > } > > @@ -369,7 +381,7 @@ static int mv_read_header(AVFormatContext *avctx) > > avio_skip(pb, 8); > > av_add_index_entry(ast, pos, timestamp, asize, 0, AVINDEX_KEYFRAME); > > av_add_index_entry(vst, pos + asize, i, vsize, 0, AVINDEX_KEYFRAME); > > - timestamp += asize / (ast->codecpar->channels * 2LL); > > + timestamp += asize / (ast->codecpar->channels * bytes_per_sample); > > } > > } else if (!version && avio_rb16(pb) == 3) { > > avio_skip(pb, 4); > > -- > > 2.33.0 > > thanks for doing this. > both patches look good. i will apply in a few days. applied. -- Peter (A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B)
diff --git a/libavformat/mvdec.c b/libavformat/mvdec.c index 8492928820..8b54a9ab04 100644 --- a/libavformat/mvdec.c +++ b/libavformat/mvdec.c @@ -299,6 +299,8 @@ static int mv_read_header(AVFormatContext *avctx) if (version == 2) { uint64_t timestamp; int v; + uint32_t bytes_per_sample; + avio_skip(pb, 22); /* allocate audio track first to prevent unnecessary seeking @@ -341,11 +343,21 @@ static int mv_read_header(AVFormatContext *avctx) } avpriv_set_pts_info(ast, 33, 1, ast->codecpar->sample_rate); - avio_skip(pb, 4); + bytes_per_sample = avio_rb32(pb); v = avio_rb32(pb); if (v == AUDIO_FORMAT_SIGNED) { - ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE; + switch (bytes_per_sample) { + case 1: + ast->codecpar->codec_id = AV_CODEC_ID_PCM_S8; + break; + case 2: + ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE; + break; + default: + avpriv_request_sample(avctx, "Audio sample size %i bytes", bytes_per_sample); + break; + } } else { avpriv_request_sample(avctx, "Audio compression (format %i)", v); } @@ -369,7 +381,7 @@ static int mv_read_header(AVFormatContext *avctx) avio_skip(pb, 8); av_add_index_entry(ast, pos, timestamp, asize, 0, AVINDEX_KEYFRAME); av_add_index_entry(vst, pos + asize, i, vsize, 0, AVINDEX_KEYFRAME); - timestamp += asize / (ast->codecpar->channels * 2LL); + timestamp += asize / (ast->codecpar->channels * bytes_per_sample); } } else if (!version && avio_rb16(pb) == 3) { avio_skip(pb, 4);