Commit d70fa4c4 authored by Alex Converse's avatar Alex Converse Committed by Justin Ruggles

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

of hardcoding 200 everywhere.
parent e4744b59
......@@ -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
......
......@@ -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 */
......
......@@ -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;
......
......@@ -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];
}
}
}
......
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