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author | Alex Converse <alex.converse@gmail.com> | 2011-09-27 10:20:23 -0700 |
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committer | Alex Converse <alex.converse@gmail.com> | 2011-09-27 14:10:12 -0700 |
commit | af3c06b4dbe3bdc801d3fdc3392e7f47b414c17c (patch) | |
tree | 1e0ab4f22f993685029dae14c64aa8a068fa55c5 | |
parent | d3c1d37a9019ed32a1710d53c1ad08c44bd072da (diff) | |
download | ffmpeg-af3c06b4dbe3bdc801d3fdc3392e7f47b414c17c.tar.gz |
avconv: Use the size of the decode sample format when allocating the audio samples buffer.
-rw-r--r-- | avconv.c | 4 |
1 files changed, 2 insertions, 2 deletions
@@ -1573,8 +1573,8 @@ static int output_packet(InputStream *ist, int ist_index, if (ist->decoding_needed) { switch(ist->st->codec->codec_type) { case AVMEDIA_TYPE_AUDIO:{ - if(pkt && samples_size < FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE)) { - samples_size = FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE); + if(pkt && samples_size < FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE)) { + samples_size = FFMAX(pkt->size * bps, AVCODEC_MAX_AUDIO_FRAME_SIZE); av_free(samples); samples= av_malloc(samples_size); } |