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use get_bits1/skip_bits[1] in ac3 and aac parsers

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Originally committed as revision 4969 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Måns Rullgård 2006-02-10 01:19:54 +00:00
parent b96532e970
commit 100a74222b
1 changed files with 21 additions and 21 deletions
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42
libavcodec/parser.c
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@ -738,7 +738,7 @@ typedef struct AC3ParseContext {
} AC3ParseContext; } AC3ParseContext;
#define AC3_HEADER_SIZE 7 #define AC3_HEADER_SIZE 7
#define AAC_HEADER_SIZE 8 #define AAC_HEADER_SIZE 7
static const int ac3_sample_rates[4] = { static const int ac3_sample_rates[4] = {
48000, 44100, 32000, 0 48000, 44100, 32000, 0
@ -815,7 +815,7 @@ static int ac3_sync(const uint8_t *buf, int *channels, int *sample_rate,
if(get_bits(&bits, 16) != 0x0b77) if(get_bits(&bits, 16) != 0x0b77)
return 0; return 0;
get_bits(&bits, 16); /* crc */ skip_bits(&bits, 16); /* crc */
fscod = get_bits(&bits, 2); fscod = get_bits(&bits, 2);
frmsizecod = get_bits(&bits, 6); frmsizecod = get_bits(&bits, 6);
@ -825,15 +825,15 @@ static int ac3_sync(const uint8_t *buf, int *channels, int *sample_rate,
bsid = get_bits(&bits, 5); bsid = get_bits(&bits, 5);
if(bsid > 8) if(bsid > 8)
return 0; return 0;
get_bits(&bits, 3); /* bsmod */ skip_bits(&bits, 3); /* bsmod */
acmod = get_bits(&bits, 3); acmod = get_bits(&bits, 3);
if(acmod & 1 && acmod != 1) if(acmod & 1 && acmod != 1)
get_bits(&bits, 2); /* cmixlev */ skip_bits(&bits, 2); /* cmixlev */
if(acmod & 4) if(acmod & 4)
get_bits(&bits, 2); /* surmixlev */ skip_bits(&bits, 2); /* surmixlev */
if(acmod & 2) if(acmod & 2)
get_bits(&bits, 2); /* dsurmod */ skip_bits(&bits, 2); /* dsurmod */
lfeon = get_bits(&bits, 1); lfeon = get_bits1(&bits);
*sample_rate = ac3_sample_rates[fscod]; *sample_rate = ac3_sample_rates[fscod];
*bit_rate = ac3_bitrates[frmsizecod] * 1000; *bit_rate = ac3_bitrates[frmsizecod] * 1000;
@ -854,26 +854,26 @@ static int aac_sync(const uint8_t *buf, int *channels, int *sample_rate,
if(get_bits(&bits, 12) != 0xfff) if(get_bits(&bits, 12) != 0xfff)
return 0; return 0;
get_bits(&bits, 1); skip_bits1(&bits); /* id */
get_bits(&bits, 2); skip_bits(&bits, 2); /* layer */
get_bits(&bits, 1); /* protection_absent */ skip_bits1(&bits); /* protection_absent */
get_bits(&bits, 2); skip_bits(&bits, 2); /* profile_objecttype */
sr = get_bits(&bits, 4); sr = get_bits(&bits, 4); /* sample_frequency_index */
if(!aac_sample_rates[sr]) if(!aac_sample_rates[sr])
return 0; return 0;
get_bits(&bits, 1); /* private_bit */ skip_bits1(&bits); /* private_bit */
ch = get_bits(&bits, 3); ch = get_bits(&bits, 3); /* channel_configuration */
if(!aac_channels[ch]) if(!aac_channels[ch])
return 0; return 0;
get_bits(&bits, 1); /* original/copy */ skip_bits1(&bits); /* original/copy */
get_bits(&bits, 1); /* home */ skip_bits1(&bits); /* home */
/* adts_variable_header */ /* adts_variable_header */
get_bits(&bits, 1); /* copyright_identification_bit */ skip_bits1(&bits); /* copyright_identification_bit */
get_bits(&bits, 1); /* copyright_identification_start */ skip_bits1(&bits); /* copyright_identification_start */
size = get_bits(&bits, 13); size = get_bits(&bits, 13); /* aac_frame_length */
get_bits(&bits, 11); /* adts_buffer_fullness */ skip_bits(&bits, 11); /* adts_buffer_fullness */
rdb = get_bits(&bits, 2); rdb = get_bits(&bits, 2); /* number_of_raw_data_blocks_in_frame */
*channels = aac_channels[ch]; *channels = aac_channels[ch];
*sample_rate = aac_sample_rates[sr]; *sample_rate = aac_sample_rates[sr];