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author | Michael Niedermayer <michaelni@gmx.at> | 2012-03-04 02:03:25 +0100 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-03-04 04:26:04 +0100 |
commit | 15c6be8c7da7b94b5e131396e9a82ab6fe4a6b64 (patch) | |
tree | 84db7f4851faba26561f846b4f112ef64d01b3ad /libavformat/tiertexseq.c | |
parent | f972193a15026a99eb2b08e7913a03f2123663da (diff) | |
parent | b7beabab4b78cc253d06c0a33f15b8ff79866e85 (diff) | |
download | ffmpeg-15c6be8c7da7b94b5e131396e9a82ab6fe4a6b64.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
tiertexseq: set correct block_align for audio
tiertexseq: set audio stream start time to 0
voc/avs: Do not change the sample rate mid-stream.
segafilm: use the sample rate as the time base for audio streams
ea: fix audio pts
psx-str: fix audio pts
vqf: set packet duration
tta demuxer: set packet duration
mpegaudio_parser: do not ignore information from the first parsed frame
mpegaudio_parser: be less picky about the start position
thp: set audio packet durations
avcodec: add a Vorbis parser to get packet duration
vorbisdec: read the previous window flag for long windows
lavc: free the output packet when encoding failed or produced no output.
lavc: preserve avpkt->destruct in ff_alloc_packet().
lavc: clarify the meaning of AVCodecContext.frame_number.
mpegts: Pad the packet buffer in handle_packet().
mpegts: Do not call read_sl_header() when no bytes remain in the buffer.
Conflicts:
libavcodec/mpegaudio_parser.c
libavcodec/version.h
libavformat/mpegts.c
tests/ref/fate/pva-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/tiertexseq.c')
-rw-r--r-- | libavformat/tiertexseq.c | 3 |
1 files changed, 2 insertions, 1 deletions
diff --git a/libavformat/tiertexseq.c b/libavformat/tiertexseq.c index d2d4e880c2..a0c06deab6 100644 --- a/libavformat/tiertexseq.c +++ b/libavformat/tiertexseq.c @@ -224,6 +224,7 @@ static int seq_read_header(AVFormatContext *s) if (!st) return AVERROR(ENOMEM); + st->start_time = 0; avpriv_set_pts_info(st, 32, 1, SEQ_SAMPLE_RATE); seq->audio_stream_index = st->index; st->codec->codec_type = AVMEDIA_TYPE_AUDIO; @@ -233,7 +234,7 @@ static int seq_read_header(AVFormatContext *s) st->codec->sample_rate = SEQ_SAMPLE_RATE; st->codec->bits_per_coded_sample = 16; st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_coded_sample * st->codec->channels; - st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample; + st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample / 8; return 0; } |