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authorMichael Niedermayer <michaelni@gmx.at>2012-03-21 00:15:18 +0100
committerMichael Niedermayer <michaelni@gmx.at>2012-03-21 01:33:53 +0100
commit0ebd83617fe008b7e9766f659cc3d9618b2d80d2 (patch)
tree23bc388bf6b66cf58d7a90c0d2529e53ed984561 /libavcodec/utils.c
parent745a33a44318ad6d6f74835a417397cdd9dda9a9 (diff)
parentc9594fe0fb6dd123fa25cb27fe5bc976ff3a9051 (diff)
downloadffmpeg-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.c9
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);