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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  libmp3lame: remove outdated comment.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
parent a923b6b8f4 770a5c6d02
commit eadd4264ee
34 changed files with 488 additions and 385 deletions
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15
libavcodec/utils.c
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@@ -993,9 +993,8 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
if (!ret && *got_packet_ptr) {
if (!(avctx->codec->capabilities & CODEC_CAP_DELAY)) {
avpkt->pts = frame->pts;
avpkt->duration = av_rescale_q(frame->nb_samples,
(AVRational){ 1, avctx->sample_rate },
avctx->time_base);
avpkt->duration = ff_samples_to_time_base(avctx,
frame->nb_samples);
}
avpkt->dts = avpkt->pts;
} else {
@@ -1053,9 +1052,8 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
once all encoders supporting CODEC_CAP_SMALL_LAST_FRAME use
encode2() */
if (fs_tmp) {
avpkt->duration = av_rescale_q(avctx->frame_size,
(AVRational){ 1, avctx->sample_rate },
avctx->time_base);
avpkt->duration = ff_samples_to_time_base(avctx,
avctx->frame_size);
}
}
avpkt->size = ret;
@@ -1128,9 +1126,8 @@ int attribute_align_arg avcodec_encode_audio(AVCodecContext *avctx,
this is needed because the avcodec_encode_audio() API does not have
a way for the user to provide pts */
if(avctx->sample_rate && avctx->time_base.num)
frame->pts = av_rescale_q(avctx->internal->sample_count,
(AVRational){ 1, avctx->sample_rate },
avctx->time_base);
frame->pts = ff_samples_to_time_base(avctx,
avctx->internal->sample_count);
else
frame->pts = AV_NOPTS_VALUE;
avctx->internal->sample_count += frame->nb_samples;