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Merge remote-tracking branch 'qatar/master'

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* qatar/master: (36 commits)
  adpcmenc: Use correct frame_size for Yamaha ADPCM.
  avcodec: add ff_samples_to_time_base() convenience function to internal.h
  adx parser: set duration
  mlp parser: set duration instead of frame_size
  gsm parser: set duration
  mpegaudio parser: set duration instead of frame_size
  (e)ac3 parser: set duration instead of frame_size
  flac parser: set duration instead of frame_size
  avcodec: add duration field to AVCodecParserContext
  avutil: add av_rescale_q_rnd() to allow different rounding
  pnmdec: remove useless .pix_fmts
  libmp3lame: support float and s32 sample formats
  libmp3lame: renaming, rearrangement, alignment, and comments
  libmp3lame: use the LAME default bit rate
  libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
  libmp3lame: cosmetics: remove some pointless comments
  libmp3lame: convert some debugging code to av_dlog()
  libmp3lame: remove outdated comment.
  libmp3lame: do not set coded_frame->key_frame.
  libmp3lame: improve error handling in MP3lame_encode_init()
  ...

Conflicts:
	doc/APIchanges
	libavcodec/libmp3lame.c
	libavcodec/pcxenc.c
	libavcodec/pnmdec.c
	libavcodec/pnmenc.c
	libavcodec/sgienc.c
	libavcodec/utils.c
	libavformat/hls.c
	libavutil/avutil.h
	libswscale/x86/swscale_mmx.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-02-21 02:49:41 +01:00
commit eadd4264ee
34 changed files with 488 additions and 385 deletions
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19
libavformat/utils.c
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@ -942,11 +942,10 @@ static void compute_pkt_fields(AVFormatContext *s, AVStream *st,
compute_frame_duration(&num, &den, st, pc, pkt);
if (den && num) {
pkt->duration = av_rescale_rnd(1, num * (int64_t)st->time_base.den, den * (int64_t)st->time_base.num, AV_ROUND_DOWN);
if(pkt->duration != 0 && s->packet_buffer)
update_initial_durations(s, st, pkt);
}
}
if(pkt->duration != 0 && s->packet_buffer)
update_initial_durations(s, st, pkt);
/* correct timestamps with byte offset if demuxers only have timestamps
on packet boundaries */
@ -1099,6 +1098,20 @@ static int read_frame_internal(AVFormatContext *s, AVPacket *pkt)
if (pkt->size) {
got_packet:
pkt->duration = 0;
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
if (st->codec->sample_rate > 0) {
pkt->duration = av_rescale_q_rnd(st->parser->duration,
(AVRational){ 1, st->codec->sample_rate },
st->time_base,
AV_ROUND_DOWN);
}
} else if (st->codec->time_base.num != 0 &&
st->codec->time_base.den != 0) {
pkt->duration = av_rescale_q_rnd(st->parser->duration,
st->codec->time_base,
st->time_base,
AV_ROUND_DOWN);
}
pkt->stream_index = st->index;
pkt->pts = st->parser->pts;
pkt->dts = st->parser->dts;