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| author | Michael Niedermayer <michaelni@gmx.at> | 2012-02-21 02:49:41 +0100 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2012-02-21 05:10:12 +0100 |
| commit | eadd4264ee4319abf9ec2f618ff925d7529f20ed (patch) | |
| tree | 4251d4eb25af927cb290e24d1b6957d584638403 /libavcodec/utils.c | |
| parent | a923b6b8f4f90d09c7c39cc8bfab7ee9d30a2843 (diff) | |
| parent | 770a5c6d025e9c8eb3f5aba9cf1d7d7938fb918a (diff) | |
| download | ffmpeg-eadd4264ee4319abf9ec2f618ff925d7529f20ed.tar.gz | |
Merge remote-tracking branch 'qatar/master'
* 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libmp3lame: remove outdated comment.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>
Diffstat (limited to 'libavcodec/utils.c')
| -rw-r--r-- | libavcodec/utils.c | 15 |
1 files changed, 6 insertions, 9 deletions
diff --git a/libavcodec/utils.c b/libavcodec/utils.c index c559fd7613..4c6823ea76 100644 --- a/libavcodec/utils.c +++ b/libavcodec/utils.c @@ -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; |