[FFmpeg-cvslog] Define POW_SF2_ZERO in aac.h and use for ff_aac_pow2sf_tabp[] offsets instead

[Alex Converse]

ffmpeg | branch: master | Alex Converse <alex.converse at gmail.com> | Tue Apr 26 14:45:48 2011 -0400| [d70fa4c4238ffa69592fffa7c817532534f303c4] | committer: Justin Ruggles

Define POW_SF2_ZERO in aac.h and use for ff_aac_pow2sf_tabp[] offsets instead
of hardcoding 200 everywhere.

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d70fa4c4238ffa69592fffa7c817532534f303c4
---

 libavcodec/aac.h          |    1 +
 libavcodec/aac_tablegen.h |    3 ++-
 libavcodec/aaccoder.c     |    8 ++++----
 libavcodec/aacdec.c       |    6 +++---
 4 files changed, 10 insertions(+), 8 deletions(-)

diff --git a/libavcodec/aac.h b/libavcodec/aac.h
index bbe7912..63ed251 100644
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -130,6 +130,7 @@ typedef struct {
 #define SCALE_MAX_POS   255    ///< scalefactor index maximum value
 #define SCALE_MAX_DIFF   60    ///< maximum scalefactor difference allowed by standard
 #define SCALE_DIFF_ZERO  60    ///< codebook index corresponding to zero scalefactor indices difference
+#define POW_SF2_ZERO    200    ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);
 
 /**
  * Long Term Prediction
diff --git a/libavcodec/aac_tablegen.h b/libavcodec/aac_tablegen.h
index c7be492..9889569 100644
--- a/libavcodec/aac_tablegen.h
+++ b/libavcodec/aac_tablegen.h
@@ -29,13 +29,14 @@
 #include "libavcodec/aac_tables.h"
 #else
 #include "libavutil/mathematics.h"
+#include "libavcodec/aac.h"
 float ff_aac_pow2sf_tab[428];
 
 void ff_aac_tableinit(void)
 {
     int i;
     for (i = 0; i < 428; i++)
-        ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
+        ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.);
 }
 #endif /* CONFIG_HARDCODED_TABLES */
 
diff --git a/libavcodec/aaccoder.c b/libavcodec/aaccoder.c
index bc59ac5..15fe430 100644
--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -108,8 +108,8 @@ static av_always_inline float quantize_and_encode_band_cost_template(
                                 int *bits, int BT_ZERO, int BT_UNSIGNED,
                                 int BT_PAIR, int BT_ESC)
 {
-    const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
-    const float  Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
+    const float IQ = ff_aac_pow2sf_tab[POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
+    const float  Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
     const float CLIPPED_ESCAPE = 165140.0f*IQ;
     int i, j;
     float cost = 0;
@@ -280,7 +280,7 @@ static float find_max_val(int group_len, int swb_size, const float *scaled) {
 }
 
 static int find_min_book(float maxval, int sf) {
-    float Q = ff_aac_pow2sf_tab[200 - sf + SCALE_ONE_POS - SCALE_DIV_512];
+    float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - sf + SCALE_ONE_POS - SCALE_DIV_512];
     float Q34 = sqrtf(Q * sqrtf(Q));
     int qmaxval, cb;
     qmaxval = maxval * Q34 + 0.4054f;
@@ -955,7 +955,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
                     dist -= b;
                 }
                 dist *= 1.0f / 512.0f / lambda;
-                quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[200 - scf + SCALE_ONE_POS - SCALE_DIV_512]);
+                quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512]);
                 if (quant_max >= 8191) { // too much, return to the previous quantizer
                     sce->sf_idx[w*16+g] = prev_scf;
                     break;
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index e289caa..bcc277c 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -811,7 +811,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                                 "audible artifact, there may be a bug in the "
                                 "decoder. ", offset[2], clipped_offset);
                     }
-                    sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + 200];
+                    sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
                 }
             } else if (band_type[idx] == NOISE_BT) {
                 for (; i < run_end; i++, idx++) {
@@ -826,7 +826,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                                 "artifact, there may be a bug in the decoder. ",
                                 offset[1], clipped_offset);
                     }
-                    sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + sf_offset + 100];
+                    sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + sf_offset - 100 + POW_SF2_ZERO];
                 }
             } else {
                 for (; i < run_end; i++, idx++) {
@@ -836,7 +836,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
                                "%s (%d) out of range.\n", sf_str[0], offset[0]);
                         return -1;
                     }
-                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
+                    sf[idx] = -ff_aac_pow2sf_tab[offset[0] + sf_offset - 200 + POW_SF2_ZERO];
                 }
             }
         }