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Merge remote-tracking branch 'qatar/master'

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* 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>
This commit is contained in:
Michael Niedermayer
2012-03-04 02:03:25 +01:00
parent f972193a15 b7beabab4b
commit 15c6be8c7d
24 changed files with 473 additions and 83 deletions
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3
libavformat/tiertexseq.c
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@@ -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;
}