stack overflow

Originally committed as revision 3389 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Michael Niedermayer 2004-08-14 14:51:10 +00:00
parent b6c50eb17c
commit df84ac2e7d

View File

@ -1078,7 +1078,6 @@ static int output_packet(AVInputStream *ist, int ist_index,
uint8_t *data_buf;
int data_size, got_picture;
AVFrame picture;
short samples[pkt && pkt->size > AVCODEC_MAX_AUDIO_FRAME_SIZE/2 ? pkt->size : AVCODEC_MAX_AUDIO_FRAME_SIZE/2];
void *buffer_to_free;
if (pkt && pkt->dts != AV_NOPTS_VALUE) { //FIXME seems redundant, as libavformat does this too
@ -1103,9 +1102,10 @@ static int output_packet(AVInputStream *ist, int ist_index,
data_size = 0;
if (ist->decoding_needed) {
switch(ist->st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
case CODEC_TYPE_AUDIO:{
/* XXX: could avoid copy if PCM 16 bits with same
endianness as CPU */
short samples[pkt && pkt->size > AVCODEC_MAX_AUDIO_FRAME_SIZE/2 ? pkt->size : AVCODEC_MAX_AUDIO_FRAME_SIZE/2];
ret = avcodec_decode_audio(&ist->st->codec, samples, &data_size,
ptr, len);
if (ret < 0)
@ -1121,7 +1121,7 @@ static int output_packet(AVInputStream *ist, int ist_index,
data_buf = (uint8_t *)samples;
ist->next_pts += ((int64_t)AV_TIME_BASE/2 * data_size) /
(ist->st->codec.sample_rate * ist->st->codec.channels);
break;
break;}
case CODEC_TYPE_VIDEO:
data_size = (ist->st->codec.width * ist->st->codec.height * 3) / 2;
/* XXX: allocate picture correctly */