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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
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c367d06702
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ffmpeg.c
2
ffmpeg.c
@ -1049,7 +1049,7 @@ static int output_packet(AVInputStream *ist, int ist_index,
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switch(ist->st->codec->codec_type) {
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case CODEC_TYPE_AUDIO:{
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if(pkt)
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samples= av_fast_realloc(samples, &samples_size, FFMAX(pkt->size, AVCODEC_MAX_AUDIO_FRAME_SIZE));
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samples= av_fast_realloc(samples, &samples_size, FFMAX(pkt->size*sizeof(*samples), AVCODEC_MAX_AUDIO_FRAME_SIZE));
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data_size= samples_size;
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/* XXX: could avoid copy if PCM 16 bits with same
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endianness as CPU */
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