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Merge remote-tracking branch 'qatar/master'

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* qatar/master:
  mpegaudioenc: Move some static tables to MpegAudioContext

Conflicts:
	libavcodec/mpegaudioenc.c
	libavcodec/mpegaudiotab.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2013-11-11 22:21:38 +01:00
commit 43bf4297e4
2 changed files with 32 additions and 35 deletions
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53
libavcodec/mpegaudioenc.c
View File

@ -64,6 +64,16 @@ typedef struct MpegAudioContext {
unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
int sblimit; /* number of used subbands */ int sblimit; /* number of used subbands */
const unsigned char *alloc_table; const unsigned char *alloc_table;
int16_t filter_bank[512];
int scale_factor_table[64];
unsigned char scale_diff_table[128];
#ifdef USE_FLOATS
float scale_factor_inv_table[64];
#else
int8_t scale_factor_shift[64];
unsigned short scale_factor_mult[64];
#endif
unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
} MpegAudioContext; } MpegAudioContext;
static av_cold int MPA_encode_init(AVCodecContext *avctx) static av_cold int MPA_encode_init(AVCodecContext *avctx)
@ -139,24 +149,24 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
#if WFRAC_BITS != 16 #if WFRAC_BITS != 16
v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS); v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
#endif #endif
filter_bank[i] = v; s->filter_bank[i] = v;
if ((i & 63) != 0) if ((i & 63) != 0)
v = -v; v = -v;
if (i != 0) if (i != 0)
filter_bank[512 - i] = v; s->filter_bank[512 - i] = v;
} }
for(i=0;i<64;i++) { for(i=0;i<64;i++) {
v = (int)(exp2((3 - i) / 3.0) * (1 << 20)); v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
if (v <= 0) if (v <= 0)
v = 1; v = 1;
scale_factor_table[i] = v; s->scale_factor_table[i] = v;
#ifdef USE_FLOATS #ifdef USE_FLOATS
scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20); s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
#else #else
#define P 15 #define P 15
scale_factor_shift[i] = 21 - P - (i / 3); s->scale_factor_shift[i] = 21 - P - (i / 3);
scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0); s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
#endif #endif
} }
for(i=0;i<128;i++) { for(i=0;i<128;i++) {
@ -171,7 +181,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
v = 3; v = 3;
else else
v = 4; v = 4;
scale_diff_table[i] = v; s->scale_diff_table[i] = v;
} }
for(i=0;i<17;i++) { for(i=0;i<17;i++) {
@ -180,7 +190,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
v = -v; v = -v;
else else
v = v * 3; v = v * 3;
total_quant_bits[i] = 12 * v; s->total_quant_bits[i] = 12 * v;
} }
return 0; return 0;
@ -327,7 +337,7 @@ static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
/* filter */ /* filter */
p = s->samples_buf[ch] + offset; p = s->samples_buf[ch] + offset;
q = filter_bank; q = s->filter_bank;
/* maxsum = 23169 */ /* maxsum = 23169 */
for(i=0;i<64;i++) { for(i=0;i<64;i++) {
sum = p[0*64] * q[0*64]; sum = p[0*64] * q[0*64];
@ -361,7 +371,8 @@ static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
s->samples_offset[ch] = offset; 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], unsigned char scale_factors[SBLIMIT][3],
int sb_samples[3][12][SBLIMIT], int sb_samples[3][12][SBLIMIT],
int sblimit) int sblimit)
@ -388,7 +399,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
use at most 2 compares to find the index */ use at most 2 compares to find the index */
index = (21 - n) * 3 - 3; index = (21 - n) * 3 - 3;
if (index >= 0) { if (index >= 0) {
while (vmax <= scale_factor_table[index+1]) while (vmax <= s->scale_factor_table[index+1])
index++; index++;
} else { } else {
index = 0; /* very unlikely case of overflow */ index = 0; /* very unlikely case of overflow */
@ -398,7 +409,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
} }
av_dlog(NULL, "%2d:%d in=%x %x %d\n", 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 */ /* store the scale factor */
av_assert2(index >=0 && index <= 63); av_assert2(index >=0 && index <= 63);
sf[i] = index; sf[i] = index;
@ -406,8 +417,8 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
/* compute the transmission factor : look if the scale factors /* compute the transmission factor : look if the scale factors
are close enough to each other */ are close enough to each other */
d1 = scale_diff_table[sf[0] - sf[1] + 64]; d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
d2 = scale_diff_table[sf[1] - sf[2] + 64]; d2 = s->scale_diff_table[sf[1] - sf[2] + 64];
/* handle the 25 cases */ /* handle the 25 cases */
switch(d1 * 5 + d2) { switch(d1 * 5 + d2) {
@ -557,12 +568,12 @@ static void compute_bit_allocation(MpegAudioContext *s,
if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) { if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
/* nothing was coded for this band: add the necessary bits */ /* nothing was coded for this band: add the necessary bits */
incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6; 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 { } else {
/* increments bit allocation */ /* increments bit allocation */
b = bit_alloc[max_ch][max_sb]; b = bit_alloc[max_ch][max_sb];
incr = total_quant_bits[alloc[b + 1]] - incr = s->total_quant_bits[alloc[b + 1]] -
total_quant_bits[alloc[b]]; s->total_quant_bits[alloc[b]];
} }
if (current_frame_size + incr <= max_frame_size) { if (current_frame_size + incr <= max_frame_size) {
@ -676,15 +687,15 @@ static void encode_frame(MpegAudioContext *s,
#ifdef USE_FLOATS #ifdef USE_FLOATS
{ {
float a; float a;
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); q[m] = (int)((a + 1.0) * steps * 0.5);
} }
#else #else
{ {
int q1, e, shift, mult; int q1, e, shift, mult;
e = s->scale_factors[ch][i][k]; e = s->scale_factors[ch][i][k];
shift = scale_factor_shift[e]; shift = s->scale_factor_shift[e];
mult = scale_factor_mult[e]; mult = s->scale_factor_mult[e];
/* normalize to P bits */ /* normalize to P bits */
if (shift < 0) if (shift < 0)
@ -739,7 +750,7 @@ static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
} }
for(i=0;i<s->nb_channels;i++) { 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); s->sb_samples[i], s->sblimit);
} }
for(i=0;i<s->nb_channels;i++) { for(i=0;i<s->nb_channels;i++) {

14
libavcodec/mpegaudiotab.h
View File

@ -79,20 +79,6 @@ static const int bitinv32[32] = {
}; };
static int16_t filter_bank[512];
static int scale_factor_table[64];
#ifdef USE_FLOATS
static float scale_factor_inv_table[64];
#else
static int8_t scale_factor_shift[64];
static unsigned short scale_factor_mult[64];
#endif
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 /* signal to noise ratio of each quantification step (could be
computed from quant_steps[]). The values are dB multiplied by 10 computed from quant_steps[]). The values are dB multiplied by 10
*/ */