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author | Baptiste Coudurier <baptiste.coudurier@gmail.com> | 2007-03-28 11:53:01 +0000 |
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committer | Baptiste Coudurier <baptiste.coudurier@gmail.com> | 2007-03-28 11:53:01 +0000 |
commit | 5508d26b7c22ba05cd036bfe58c9658f9afc0ee0 (patch) | |
tree | 780920137d6ab8e76c5e5b42e48b99b5dc8afeb9 /ffmpeg.c | |
parent | c367d06702b3a9cb6b5cf02435f76c5dd5f321e7 (diff) | |
download | ffmpeg-streaming-5508d26b7c22ba05cd036bfe58c9658f9afc0ee0.zip ffmpeg-streaming-5508d26b7c22ba05cd036bfe58c9658f9afc0ee0.tar.gz |
fix decoding of big audio packets (48k 16bit 2 channels), needed size is related to samples which is short * while len passed to decode_audio2 is related to pkt->data which is uint8_t *
Originally committed as revision 8537 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'ffmpeg.c')
-rw-r--r-- | ffmpeg.c | 2 |
1 files changed, 1 insertions, 1 deletions
@@ -1049,7 +1049,7 @@ static int output_packet(AVInputStream *ist, int ist_index, switch(ist->st->codec->codec_type) { case CODEC_TYPE_AUDIO:{ if(pkt) - samples= av_fast_realloc(samples, &samples_size, FFMAX(pkt->size, AVCODEC_MAX_AUDIO_FRAME_SIZE)); + samples= av_fast_realloc(samples, &samples_size, FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE)); data_size= samples_size; /* XXX: could avoid copy if PCM 16 bits with same endianness as CPU */ |