Skip to content

F
ffmpeg.wasm-core
Commit 45ef9639 authored by Diego Biurrun's avatar Diego Biurrun
Browse files
Options

mpegaudioenc: Move some static tables to MpegAudioContext 

This reduces global state and the amount of globally visible tables.
parent 5d6a990c
Hide whitespace changes
Inline Side-by-side
Showing with 23 additions and 26 deletions
libavcodec/mpegaudioenc.c
View file @ 45ef9639
......@@ -61,6 +61,11 @@ typedef struct MpegAudioContext {
unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
int sblimit; /* number of used subbands */
const unsigned char *alloc_table;
int16_t filter_bank[512];
int scale_factor_table[64];
unsigned char scale_diff_table[128];
float scale_factor_inv_table[64];
unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
} MpegAudioContext;
static av_cold int MPA_encode_init(AVCodecContext *avctx)
......@@ -136,19 +141,19 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
#if WFRAC_BITS != 16
v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
#endif
filter_bank[i] = v;
s->filter_bank[i] = v;
if ((i & 63) != 0)
v = -v;
if (i != 0)
filter_bank[512 - i] = v;
s->filter_bank[512 - i] = v;
}
for(i=0;i<64;i++) {
v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
if (v <= 0)
v = 1;
scale_factor_table[i] = v;
scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
s->scale_factor_table[i] = v;
s->scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
}
for(i=0;i<128;i++) {
v = i - 64;
......@@ -162,7 +167,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
v = 3;
else
v = 4;
scale_diff_table[i] = v;
s->scale_diff_table[i] = v;
}
for(i=0;i<17;i++) {
......@@ -171,7 +176,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
v = -v;
else
v = v * 3;
total_quant_bits[i] = 12 * v;
s->total_quant_bits[i] = 12 * v;
}
return 0;
......@@ -318,7 +323,7 @@ static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
/* filter */
p = s->samples_buf[ch] + offset;
q = filter_bank;
q = s->filter_bank;
/* maxsum = 23169 */
for(i=0;i<64;i++) {
sum = p[0*64] * q[0*64];
......@@ -352,7 +357,8 @@ static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
s->samples_offset[ch] = offset;
}
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
static void compute_scale_factors(MpegAudioContext *s,
unsigned char scale_code[SBLIMIT],
unsigned char scale_factors[SBLIMIT][3],
int sb_samples[3][12][SBLIMIT],
int sblimit)
......@@ -379,7 +385,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
use at most 2 compares to find the index */
index = (21 - n) * 3 - 3;
if (index >= 0) {
while (vmax <= scale_factor_table[index+1])
while (vmax <= s->scale_factor_table[index+1])
index++;
} else {
index = 0; /* very unlikely case of overflow */
......@@ -389,7 +395,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
}
av_dlog(NULL, "%2d:%d in=%x %x %d\n",
j, i, vmax, scale_factor_table[index], index);
j, i, vmax, s->scale_factor_table[index], index);
/* store the scale factor */
assert(index >=0 && index <= 63);
sf[i] = index;
......@@ -397,8 +403,8 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
/* compute the transmission factor : look if the scale factors
are close enough to each other */
d1 = scale_diff_table[sf[0] - sf[1] + 64];
d2 = scale_diff_table[sf[1] - sf[2] + 64];
d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
d2 = s->scale_diff_table[sf[1] - sf[2] + 64];
/* handle the 25 cases */
switch(d1 * 5 + d2) {
......@@ -548,12 +554,12 @@ static void compute_bit_allocation(MpegAudioContext *s,
if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
/* nothing was coded for this band: add the necessary bits */
incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
incr += total_quant_bits[alloc[1]];
incr += s->total_quant_bits[alloc[1]];
} else {
/* increments bit allocation */
b = bit_alloc[max_ch][max_sb];
incr = total_quant_bits[alloc[b + 1]] -
total_quant_bits[alloc[b]];
incr = s->total_quant_bits[alloc[b + 1]] -
s->total_quant_bits[alloc[b]];
}
if (current_frame_size + incr <= max_frame_size) {
......@@ -665,7 +671,7 @@ static void encode_frame(MpegAudioContext *s,
float a;
sample = s->sb_samples[ch][k][l + m][i];
/* divide by scale factor */
a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]];
q[m] = (int)((a + 1.0) * steps * 0.5);
if (q[m] >= steps)
q[m] = steps - 1;
......@@ -711,7 +717,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
for(i=0;i<s->nb_channels;i++) {
compute_scale_factors(s->scale_code[i], s->scale_factors[i],
compute_scale_factors(s, s->scale_code[i], s->scale_factors[i],
s->sb_samples[i], s->sblimit);
}
for(i=0;i<s->nb_channels;i++) {
......
libavcodec/mpegaudiotab.h
View file @ 45ef9639
......@@ -79,15 +79,6 @@ static const int bitinv32[32] = {
};
static int16_t filter_bank[512];
static int scale_factor_table[64];
static float scale_factor_inv_table[64];
static unsigned char scale_diff_table[128];
/* total number of bits per allocation group */
static unsigned short total_quant_bits[17];
/* signal to noise ratio of each quantification step (could be
computed from quant_steps[]). The values are dB multiplied by 10
*/
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment