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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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11
libavformat/segafilm.c
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@ -197,8 +197,13 @@ static int film_read_header(AVFormatContext *s)
if (!film->sample_table)
return AVERROR(ENOMEM);
for(i=0; i<s->nb_streams; i++)
avpriv_set_pts_info(s->streams[i], 33, 1, film->base_clock);
for (i = 0; i < s->nb_streams; i++) {
st = s->streams[i];
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO)
avpriv_set_pts_info(st, 33, 1, film->base_clock);
else
avpriv_set_pts_info(st, 64, 1, film->audio_samplerate);
}
audio_frame_counter = 0;
for (i = 0; i < film->sample_count; i++) {
@ -213,8 +218,6 @@ static int film_read_header(AVFormatContext *s)
if (AV_RB32(&scratch[8]) == 0xFFFFFFFF) {
film->sample_table[i].stream = film->audio_stream_index;
film->sample_table[i].pts = audio_frame_counter;
film->sample_table[i].pts *= film->base_clock;
film->sample_table[i].pts /= film->audio_samplerate;
if (film->audio_type == CODEC_ID_ADPCM_ADX)
audio_frame_counter += (film->sample_table[i].sample_size * 32 /