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fix parsing of RealAudio AC-3/DolbyNet

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Originally committed as revision 7888 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Justin Ruggles 2007-02-09 02:02:09 +00:00
parent 7156aeb98a
commit f7a49547de
1 changed files with 6 additions and 5 deletions

11
libavcodec/parser.c
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@ -657,7 +657,7 @@ static const int aac_channels[8] = {
static int ac3_sync(const uint8_t *buf, int *channels, int *sample_rate,
int *bit_rate, int *samples)
{
unsigned int fscod, frmsizecod, acmod, bsid, lfeon;
unsigned int fscod, frmsizecod, acmod, bsid, lfeon, halfratecod;
unsigned int strmtyp, substreamid, frmsiz, fscod2, numblkscod;
GetBitContext bits;
@ -667,7 +667,7 @@ static int ac3_sync(const uint8_t *buf, int *channels, int *sample_rate,
return 0;
bsid = show_bits_long(&bits, 29) & 0x1f;
if(bsid <= 8) { /* Normal AC-3 */
if(bsid <= 10) { /* Normal AC-3 */
skip_bits(&bits, 16); /* crc */
fscod = get_bits(&bits, 2);
frmsizecod = get_bits(&bits, 6);
@ -686,13 +686,14 @@ static int ac3_sync(const uint8_t *buf, int *channels, int *sample_rate,
skip_bits(&bits, 2); /* dsurmod */
lfeon = get_bits1(&bits);
*sample_rate = ac3_sample_rates[fscod];
*bit_rate = ac3_bitrates[frmsizecod] * 1000;
halfratecod = FFMAX(bsid, 8) - 8;
*sample_rate = ac3_sample_rates[fscod] >> halfratecod;
*bit_rate = (ac3_bitrates[frmsizecod] * 1000) >> halfratecod;
*channels = ac3_channels[acmod] + lfeon;
*samples = 6 * 256;
return ac3_frame_sizes[frmsizecod][fscod] * 2;
} else if (bsid >= 10 && bsid <= 16) { /* Enhanced AC-3 */
} else if (bsid > 10 && bsid <= 16) { /* Enhanced AC-3 */
strmtyp = get_bits(&bits, 2);
substreamid = get_bits(&bits, 3);