generic-library/ffmpeg
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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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6
libavformat/utils.c
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@@ -759,9 +759,6 @@ static int get_audio_frame_size(AVCodecContext *enc, int size)
{
int frame_size;
if(enc->codec_id == CODEC_ID_VORBIS)
return -1;
if (enc->frame_size <= 1) {
int bits_per_sample = av_get_bits_per_sample(enc->codec_id);
@@ -2105,8 +2102,7 @@ static int has_codec_parameters(AVCodecContext *avctx)
case AVMEDIA_TYPE_AUDIO:
val = avctx->sample_rate && avctx->channels && avctx->sample_fmt != AV_SAMPLE_FMT_NONE;
if (!avctx->frame_size &&
(avctx->codec_id == CODEC_ID_VORBIS ||
avctx->codec_id == CODEC_ID_AAC ||
(avctx->codec_id == CODEC_ID_AAC ||
avctx->codec_id == CODEC_ID_MP1 ||
avctx->codec_id == CODEC_ID_MP2 ||
avctx->codec_id == CODEC_ID_MP3 ||