[FFmpeg-cvslog] r14695 - trunk/libavcodec/aac.c

[superdump]

Author: superdump
Date: Mon Aug 11 13:22:48 2008
New Revision: 14695

Log:
Synchronise code with AAC decoder in SoC


Modified:
   trunk/libavcodec/aac.c

Modified: trunk/libavcodec/aac.c
==============================================================================
--- trunk/libavcodec/aac.c	(original)
+++ trunk/libavcodec/aac.c	Mon Aug 11 13:22:48 2008
@@ -436,7 +436,6 @@ static int decode_band_types(AACContext 
 /**
  * Decode scalefactors; reference: table 4.47.
  *
- * @param   mix_gain            channel gain (Not used by AAC bitstream.)
  * @param   global_gain         first scalefactor value as scalefactors are differentially coded
  * @param   band_type           array of the used band type
  * @param   band_type_run_end   array of the last scalefactor band of a band type run
@@ -445,7 +444,7 @@ static int decode_band_types(AACContext 
  * @return  Returns error status. 0 - OK, !0 - error
  */
 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb,
-        float mix_gain, unsigned int global_gain, IndividualChannelStream * ics,
+        unsigned int global_gain, IndividualChannelStream * ics,
         enum BandType band_type[120], int band_type_run_end[120]) {
     const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
     int g, i, idx = 0;
@@ -469,7 +468,6 @@ static int decode_scalefactors(AACContex
                         return -1;
                     }
                     sf[idx]  = ff_aac_pow2sf_tab[-offset[2] + 300];
-                    sf[idx] *= mix_gain;
                 }
             }else if(band_type[idx] == NOISE_BT) {
                 for(; i < run_end; i++, idx++) {
@@ -483,7 +481,6 @@ static int decode_scalefactors(AACContex
                         return -1;
                     }
                     sf[idx]  = -ff_aac_pow2sf_tab[ offset[1] + sf_offset];
-                    sf[idx] *= mix_gain;
                 }
             }else {
                 for(; i < run_end; i++, idx++) {
@@ -494,7 +491,6 @@ static int decode_scalefactors(AACContex
                         return -1;
                     }
                     sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
-                    sf[idx] *= mix_gain;
                 }
             }
         }
@@ -649,7 +645,7 @@ static int decode_cpe(AACContext * ac, G
  */
 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
     // TODO : sbr_extension implementation
-    av_log(ac->avccontext, AV_LOG_DEBUG, "aac: SBR not yet supported.\n");
+    av_log_missing_feature(ac->avccontext, "SBR", 0);
     skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type
     return cnt;
 }
@@ -769,11 +765,10 @@ static void apply_dependent_coupling(AAC
     for (g = 0; g < ics->num_window_groups; g++) {
         for (i = 0; i < ics->max_sfb; i++, idx++) {
             if (cc->ch[0].band_type[idx] != ZERO_BT) {
-                float gain = cc->coup.gain[index][idx] * sce->mixing_gain;
                 for (group = 0; group < ics->group_len[g]; group++) {
                     for (k = offsets[i]; k < offsets[i+1]; k++) {
                         // XXX dsputil-ize
-                        dest[group*128+k] += gain * src[group*128+k];
+                        dest[group*128+k] += cc->coup.gain[index][idx] * src[group*128+k];
                     }
                 }
             }
@@ -790,9 +785,8 @@ static void apply_dependent_coupling(AAC
  */
 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
     int i;
-    float gain = cc->coup.gain[index][0] * sce->mixing_gain;
     for (i = 0; i < 1024; i++)
-        sce->ret[i] += gain * (cc->ch[0].ret[i] - ac->add_bias);
+        sce->ret[i] += cc->coup.gain[index][0] * (cc->ch[0].ret[i] - ac->add_bias);
 }
 
     if (!ac->is_saved) {
@@ -817,11 +811,11 @@ static void apply_independent_coupling(A
 
 static av_cold int aac_decode_close(AVCodecContext * avccontext) {
     AACContext * ac = avccontext->priv_data;
-    int i, j;
+    int i, type;
 
     for (i = 0; i < MAX_ELEM_ID; i++) {
-        for(j = 0; j < 4; j++)
-            av_freep(&ac->che[j][i]);
+        for(type = 0; type < 4; type++)
+            av_freep(&ac->che[type][i]);
     }
 
     ff_mdct_end(&ac->mdct);