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| author | Marton Balint <cus@passwd.hu> | 2018-08-26 21:23:02 +0200 |
|---|---|---|
| committer | Marton Balint <cus@passwd.hu> | 2018-09-05 22:41:09 +0200 |
| commit | 6aaf1b504c6a0e0e5e2f4c97712b0f83a4115e10 (patch) | |
| tree | e98b234c1ccf7e4b18d07aa212d6484fa99a6b08 | |
| parent | 776909e42e2a9cef37035f070464bbbfa3441c39 (diff) | |
| download | ffmpeg-6aaf1b504c6a0e0e5e2f4c97712b0f83a4115e10.tar.gz | |
avformat/mxfdec: do not use sound essence descriptor quantization bits for bits_per_coded_sample
It refers to the uncompressed quantization, therefore is not correct for AAC.
Also change mxf_set_pts to work based on current edit unit if
bits_per_coded_sample is not available.
Fixes error messages in the sample of ticket #7366.
Signed-off-by: Marton Balint <cus@passwd.hu>
| -rw-r--r-- | libavformat/mxfdec.c | 11 |
1 files changed, 7 insertions, 4 deletions
diff --git a/libavformat/mxfdec.c b/libavformat/mxfdec.c index 8e1089620f..134f27784b 100644 --- a/libavformat/mxfdec.c +++ b/libavformat/mxfdec.c @@ -2390,7 +2390,6 @@ static int mxf_parse_structural_metadata(MXFContext *mxf) if (st->codecpar->codec_id == AV_CODEC_ID_NONE || (st->codecpar->codec_id == AV_CODEC_ID_PCM_ALAW && (enum AVCodecID)container_ul->id != AV_CODEC_ID_NONE)) st->codecpar->codec_id = (enum AVCodecID)container_ul->id; st->codecpar->channels = descriptor->channels; - st->codecpar->bits_per_coded_sample = descriptor->bits_per_sample; if (descriptor->sample_rate.den > 0) { st->codecpar->sample_rate = descriptor->sample_rate.num / descriptor->sample_rate.den; @@ -2423,6 +2422,7 @@ static int mxf_parse_structural_metadata(MXFContext *mxf) } else if (st->codecpar->codec_id == AV_CODEC_ID_MP2) { st->need_parsing = AVSTREAM_PARSE_FULL; } + st->codecpar->bits_per_coded_sample = av_get_bits_per_sample(st->codecpar->codec_id); } else if (st->codecpar->codec_type == AVMEDIA_TYPE_DATA) { enum AVMediaType type; container_ul = mxf_get_codec_ul(mxf_data_essence_container_uls, essence_container_ul); @@ -3269,7 +3269,8 @@ static int64_t mxf_set_current_edit_unit(MXFContext *mxf, AVStream *st, int64_t static int mxf_set_audio_pts(MXFContext *mxf, AVCodecParameters *par, AVPacket *pkt) { - MXFTrack *track = mxf->fc->streams[pkt->stream_index]->priv_data; + AVStream *st = mxf->fc->streams[pkt->stream_index]; + MXFTrack *track = st->priv_data; int64_t bits_per_sample = par->bits_per_coded_sample; if (!bits_per_sample) @@ -3280,8 +3281,10 @@ static int mxf_set_audio_pts(MXFContext *mxf, AVCodecParameters *par, if ( par->channels <= 0 || bits_per_sample <= 0 || par->channels * (int64_t)bits_per_sample < 8) - return AVERROR(EINVAL); - track->sample_count += pkt->size / (par->channels * (int64_t)bits_per_sample / 8); + track->sample_count = mxf_compute_sample_count(mxf, st, av_rescale_q(track->sample_count, st->time_base, av_inv_q(track->edit_rate)) + 1); + else + track->sample_count += pkt->size / (par->channels * (int64_t)bits_per_sample / 8); + return 0; } |