summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
authorAlex Converse <alex.converse@gmail.com>2009-02-12 13:31:43 +0000
committerRobert Swain <robert.swain@gmail.com>2009-02-12 13:31:43 +0000
commit30272450f98473dee6f6d06e42cae37c3787ebf8 (patch)
tree00293e458c67b914e3ce0f96bbb9b82dc1fc8bfd
parentff587009ae60f6cf76d70879986125696490f99c (diff)
Add support for sample rate index 12, 7350 Hz
Patch by Alex Converse ( alex converse gmail com ) Originally committed as revision 17180 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavcodec/aac.c6
-rw-r--r--libavcodec/aacdectab.h10
-rw-r--r--libavcodec/aactab.c6
3 files changed, 12 insertions, 10 deletions
diff --git a/libavcodec/aac.c b/libavcodec/aac.c
index 4273e796be..57bafda0a6 100644
--- a/libavcodec/aac.c
+++ b/libavcodec/aac.c
@@ -173,7 +173,7 @@ static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_E
skip_bits(gb, 2); // object_type
sampling_index = get_bits(gb, 4);
- if(sampling_index > 11) {
+ if(sampling_index > 12) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
return -1;
}
@@ -326,7 +326,7 @@ static int decode_audio_specific_config(AACContext * ac, void *data, int data_si
if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
return -1;
- if(ac->m4ac.sampling_index > 11) {
+ if(ac->m4ac.sampling_index > 12) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
return -1;
}
@@ -1555,7 +1555,7 @@ static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data
av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
return -1;
}
- if (ac->m4ac.sampling_index > 11) {
+ if (ac->m4ac.sampling_index > 12) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
return -1;
}
diff --git a/libavcodec/aacdectab.h b/libavcodec/aacdectab.h
index d7e0099d3c..34b0abe729 100644
--- a/libavcodec/aacdectab.h
+++ b/libavcodec/aacdectab.h
@@ -142,7 +142,8 @@ static const uint16_t *swb_offset_1024[] = {
swb_offset_1024_96, swb_offset_1024_96, swb_offset_1024_64,
swb_offset_1024_48, swb_offset_1024_48, swb_offset_1024_32,
swb_offset_1024_24, swb_offset_1024_24, swb_offset_1024_16,
- swb_offset_1024_16, swb_offset_1024_16, swb_offset_1024_8
+ swb_offset_1024_16, swb_offset_1024_16, swb_offset_1024_8,
+ swb_offset_1024_8
};
static const uint16_t *swb_offset_128[] = {
@@ -151,7 +152,8 @@ static const uint16_t *swb_offset_128[] = {
swb_offset_128_96, swb_offset_128_96, swb_offset_128_96,
swb_offset_128_48, swb_offset_128_48, swb_offset_128_48,
swb_offset_128_24, swb_offset_128_24, swb_offset_128_16,
- swb_offset_128_16, swb_offset_128_16, swb_offset_128_8
+ swb_offset_128_16, swb_offset_128_16, swb_offset_128_8,
+ swb_offset_128_8
};
// @}
@@ -163,11 +165,11 @@ static const uint16_t *swb_offset_128[] = {
* @{
*/
static const uint8_t tns_max_bands_1024[] = {
- 31, 31, 34, 40, 42, 51, 46, 46, 42, 42, 42, 39
+ 31, 31, 34, 40, 42, 51, 46, 46, 42, 42, 42, 39, 39
};
static const uint8_t tns_max_bands_128[] = {
- 9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14
+ 9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14
};
// @}
diff --git a/libavcodec/aactab.c b/libavcodec/aactab.c
index 56c7bf707f..b14b3958b2 100644
--- a/libavcodec/aactab.c
+++ b/libavcodec/aactab.c
@@ -36,15 +36,15 @@ DECLARE_ALIGNED(16, float, ff_aac_kbd_long_1024[1024]);
DECLARE_ALIGNED(16, float, ff_aac_kbd_short_128[128]);
const uint8_t ff_aac_num_swb_1024[] = {
- 41, 41, 47, 49, 49, 51, 47, 47, 43, 43, 43, 40
+ 41, 41, 47, 49, 49, 51, 47, 47, 43, 43, 43, 40, 40
};
const uint8_t ff_aac_num_swb_128[] = {
- 12, 12, 12, 14, 14, 14, 15, 15, 15, 15, 15, 15
+ 12, 12, 12, 14, 14, 14, 15, 15, 15, 15, 15, 15, 15
};
const uint8_t ff_aac_pred_sfb_max[] = {
- 33, 33, 38, 40, 40, 40, 41, 41, 37, 37, 37, 34
+ 33, 33, 38, 40, 40, 40, 41, 41, 37, 37, 37, 34, 34
};
const uint32_t ff_aac_scalefactor_code[121] = {