Add support for sample rate index 12, 7350 Hz

Patch by Alex Converse ( alex converse gmail com )


git-svn-id: file:///var/local/repositories/ffmpeg/trunk@17180 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

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;
         }

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
 };
 // @}
 

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] = {