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author | Michael Niedermayer <michaelni@gmx.at> | 2012-03-21 00:15:18 +0100 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-03-21 01:33:53 +0100 |
commit | 0ebd83617fe008b7e9766f659cc3d9618b2d80d2 (patch) | |
tree | 23bc388bf6b66cf58d7a90c0d2529e53ed984561 /libavcodec/utils.c | |
parent | 745a33a44318ad6d6f74835a417397cdd9dda9a9 (diff) | |
parent | c9594fe0fb6dd123fa25cb27fe5bc976ff3a9051 (diff) | |
download | ffmpeg-0ebd83617fe008b7e9766f659cc3d9618b2d80d2.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master: (27 commits)
avconv: free packet in write_frame() when discarding due to frame number limit
FATE: use +/- flag option syntax for vp8 emu-edge tests
lavf: make av_interleave_packet_per_dts() private.
lavf: deprecate av_read_packet().
oggdec: output correct timestamps for Vorbis
avconv: pass input stream timestamps to audio encoders
lavc: shrink encoded audio packet size after encoding.
xa: set correct bit rate
xa: do not set bit_rate, block_align, or bits_per_coded_sample
xa: fix end-of-file handling
xa: fix timestamp calculation
bink: fix typo in FFALIGN() argument
bink: align plane width to 8 when calculating bundle sizes
doc: pass -Idoc texi2html and texi2pod
doc: texi2pod: add -I flag
movenc: Add a min_frag_duration option
rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
libavformat: Set the default for the max_delay option to -1
Generate manpages for AV{Format,Codec}Context AVOptions.
doc/avconv: remove entries for AVOptions.
...
Conflicts:
doc/Makefile
doc/ffmpeg.texi
doc/muxers.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/options.c
libavcodec/vp8.c
libavformat/options.c
tests/fate/demux.mak
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/utils.c')
-rw-r--r-- | libavcodec/utils.c | 9 |
1 files changed, 8 insertions, 1 deletions
diff --git a/libavcodec/utils.c b/libavcodec/utils.c index 7e95560f30..e61ace6e75 100644 --- a/libavcodec/utils.c +++ b/libavcodec/utils.c @@ -1069,8 +1069,15 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx, if (fs_tmp) avctx->frame_size = fs_tmp; } - if (!ret) + if (!ret) { + if (!user_packet && avpkt->data) { + uint8_t *new_data = av_realloc(avpkt->data, avpkt->size); + if (new_data) + avpkt->data = new_data; + } + avctx->frame_number++; + } if (ret < 0 || !*got_packet_ptr) av_free_packet(avpkt); |