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>
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@@ -1069,8 +1069,15 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
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if (fs_tmp)
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avctx->frame_size = fs_tmp;
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}
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if (!ret)
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if (!ret) {
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if (!user_packet && avpkt->data) {
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uint8_t *new_data = av_realloc(avpkt->data, avpkt->size);
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if (new_data)
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avpkt->data = new_data;
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}
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avctx->frame_number++;
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}
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if (ret < 0 || !*got_packet_ptr)
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av_free_packet(avpkt);
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