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ffmpeg.wasm-core
Commit 06d01188 authored by Vitor Sessak's avatar Vitor Sessak
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More mp{1,2,3} 32-point DCT transform to our common DCT framework. 

Should allow for future SIMD optimizations.

Originally committed as revision 23912 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent cae70f99
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Showing with 29 additions and 240 deletions
libavcodec/dct.c
View file @ 06d01188
......@@ -31,6 +31,9 @@
#include "libavutil/mathematics.h"
#include "fft.h"
#define DCT32_FLOAT
#include "dct32.c"
/* sin((M_PI * x / (2*n)) */
#define SIN(s,n,x) (s->costab[(n) - (x)])
......@@ -167,6 +170,11 @@ static void ff_dct_calc_II_c(DCTContext *ctx, FFTSample *data)
}
}
static void dct32_func(DCTContext *ctx, FFTSample *data)
{
ctx->dct32(data, data);
}
void ff_dct_calc(DCTContext *s, FFTSample *data)
{
s->dct_calc(s, data);
......@@ -200,6 +208,12 @@ av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
case DCT_III: s->dct_calc = ff_dct_calc_III_c; break;
case DST_I : s->dct_calc = ff_dst_calc_I_c; break;
}
if (inverse == DCT_II && nbits == 5)
s->dct_calc = dct32_func;
s->dct32 = dct32;
return 0;
}
......
libavcodec/fft.h
View file @ 06d01188
......@@ -228,6 +228,7 @@ struct DCTContext {
const float *costab;
FFTSample *csc2;
void (*dct_calc)(struct DCTContext *s, FFTSample *data);
void (*dct32)(FFTSample *out, const FFTSample *in);
};
/**
......
libavcodec/mpegaudio.h
View file @ 06d01188
......@@ -33,6 +33,7 @@
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
#define CONFIG_AUDIO_NONSHORT 0
......@@ -156,6 +157,9 @@ typedef struct MPADecodeContext {
int dither_state;
int error_recognition;
AVCodecContext* avctx;
#if CONFIG_FLOAT
DCTContext dct;
#endif
void (*apply_window_mp3)(MPA_INT *synth_buf, MPA_INT *window,
int *dither_state, OUT_INT *samples, int incr);
} MPADecodeContext;
......
libavcodec/mpegaudiodec.c
View file @ 06d01188
......@@ -67,6 +67,12 @@
#include "mpegaudiodata.h"
#include "mpegaudiodectab.h"
#if CONFIG_FLOAT
# include "fft.h"
#else
# include "dct32.c"
#endif
static void compute_antialias_integer(MPADecodeContext *s, GranuleDef *g);
static void compute_antialias_float(MPADecodeContext *s, GranuleDef *g);
static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
......@@ -349,6 +355,9 @@ static av_cold int decode_init(AVCodecContext * avctx)
scale_factor_mult[i][2]);
}
#if CONFIG_FLOAT
ff_dct_init(&s->dct, 5, DCT_II);
#endif
RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
/* huffman decode tables */
......@@ -512,245 +521,6 @@ static av_cold int decode_init(AVCodecContext * avctx)
return 0;
}
/* tab[i][j] = 1.0 / (2.0 * cos(pi*(2*k+1) / 2^(6 - j))) */
/* cos(i*pi/64) */
#define COS0_0 FIXHR(0.50060299823519630134/2)
#define COS0_1 FIXHR(0.50547095989754365998/2)
#define COS0_2 FIXHR(0.51544730992262454697/2)
#define COS0_3 FIXHR(0.53104259108978417447/2)
#define COS0_4 FIXHR(0.55310389603444452782/2)
#define COS0_5 FIXHR(0.58293496820613387367/2)
#define COS0_6 FIXHR(0.62250412303566481615/2)
#define COS0_7 FIXHR(0.67480834145500574602/2)
#define COS0_8 FIXHR(0.74453627100229844977/2)
#define COS0_9 FIXHR(0.83934964541552703873/2)
#define COS0_10 FIXHR(0.97256823786196069369/2)
#define COS0_11 FIXHR(1.16943993343288495515/4)
#define COS0_12 FIXHR(1.48416461631416627724/4)
#define COS0_13 FIXHR(2.05778100995341155085/8)
#define COS0_14 FIXHR(3.40760841846871878570/8)
#define COS0_15 FIXHR(10.19000812354805681150/32)
#define COS1_0 FIXHR(0.50241928618815570551/2)
#define COS1_1 FIXHR(0.52249861493968888062/2)
#define COS1_2 FIXHR(0.56694403481635770368/2)
#define COS1_3 FIXHR(0.64682178335999012954/2)
#define COS1_4 FIXHR(0.78815462345125022473/2)
#define COS1_5 FIXHR(1.06067768599034747134/4)
#define COS1_6 FIXHR(1.72244709823833392782/4)
#define COS1_7 FIXHR(5.10114861868916385802/16)
#define COS2_0 FIXHR(0.50979557910415916894/2)
#define COS2_1 FIXHR(0.60134488693504528054/2)
#define COS2_2 FIXHR(0.89997622313641570463/2)
#define COS2_3 FIXHR(2.56291544774150617881/8)
#define COS3_0 FIXHR(0.54119610014619698439/2)
#define COS3_1 FIXHR(1.30656296487637652785/4)
#define COS4_0 FIXHR(0.70710678118654752439/2)
/* butterfly operator */
#define BF(a, b, c, s)\
{\
tmp0 = val##a + val##b;\
tmp1 = val##a - val##b;\
val##a = tmp0;\
val##b = MULH3(tmp1, c, 1<<(s));\
}
#define BF0(a, b, c, s)\
{\
tmp0 = tab[a] + tab[b];\
tmp1 = tab[a] - tab[b];\
val##a = tmp0;\
val##b = MULH3(tmp1, c, 1<<(s));\
}
#define BF1(a, b, c, d)\
{\
BF(a, b, COS4_0, 1);\
BF(c, d,-COS4_0, 1);\
val##c += val##d;\
}
#define BF2(a, b, c, d)\
{\
BF(a, b, COS4_0, 1);\
BF(c, d,-COS4_0, 1);\
val##c += val##d;\
val##a += val##c;\
val##c += val##b;\
val##b += val##d;\
}
#define ADD(a, b) val##a += val##b
/* DCT32 without 1/sqrt(2) coef zero scaling. */
static void dct32(INTFLOAT *out, const INTFLOAT *tab)
{
INTFLOAT tmp0, tmp1;
INTFLOAT val0 , val1 , val2 , val3 , val4 , val5 , val6 , val7 ,
val8 , val9 , val10, val11, val12, val13, val14, val15,
val16, val17, val18, val19, val20, val21, val22, val23,
val24, val25, val26, val27, val28, val29, val30, val31;
/* pass 1 */
BF0( 0, 31, COS0_0 , 1);
BF0(15, 16, COS0_15, 5);
/* pass 2 */
BF( 0, 15, COS1_0 , 1);
BF(16, 31,-COS1_0 , 1);
/* pass 1 */
BF0( 7, 24, COS0_7 , 1);
BF0( 8, 23, COS0_8 , 1);
/* pass 2 */
BF( 7, 8, COS1_7 , 4);
BF(23, 24,-COS1_7 , 4);
/* pass 3 */
BF( 0, 7, COS2_0 , 1);
BF( 8, 15,-COS2_0 , 1);
BF(16, 23, COS2_0 , 1);
BF(24, 31,-COS2_0 , 1);
/* pass 1 */
BF0( 3, 28, COS0_3 , 1);
BF0(12, 19, COS0_12, 2);
/* pass 2 */
BF( 3, 12, COS1_3 , 1);
BF(19, 28,-COS1_3 , 1);
/* pass 1 */
BF0( 4, 27, COS0_4 , 1);
BF0(11, 20, COS0_11, 2);
/* pass 2 */
BF( 4, 11, COS1_4 , 1);
BF(20, 27,-COS1_4 , 1);
/* pass 3 */
BF( 3, 4, COS2_3 , 3);
BF(11, 12,-COS2_3 , 3);
BF(19, 20, COS2_3 , 3);
BF(27, 28,-COS2_3 , 3);
/* pass 4 */
BF( 0, 3, COS3_0 , 1);
BF( 4, 7,-COS3_0 , 1);
BF( 8, 11, COS3_0 , 1);
BF(12, 15,-COS3_0 , 1);
BF(16, 19, COS3_0 , 1);
BF(20, 23,-COS3_0 , 1);
BF(24, 27, COS3_0 , 1);
BF(28, 31,-COS3_0 , 1);
/* pass 1 */
BF0( 1, 30, COS0_1 , 1);
BF0(14, 17, COS0_14, 3);
/* pass 2 */
BF( 1, 14, COS1_1 , 1);
BF(17, 30,-COS1_1 , 1);
/* pass 1 */
BF0( 6, 25, COS0_6 , 1);
BF0( 9, 22, COS0_9 , 1);
/* pass 2 */
BF( 6, 9, COS1_6 , 2);
BF(22, 25,-COS1_6 , 2);
/* pass 3 */
BF( 1, 6, COS2_1 , 1);
BF( 9, 14,-COS2_1 , 1);
BF(17, 22, COS2_1 , 1);
BF(25, 30,-COS2_1 , 1);
/* pass 1 */
BF0( 2, 29, COS0_2 , 1);
BF0(13, 18, COS0_13, 3);
/* pass 2 */
BF( 2, 13, COS1_2 , 1);
BF(18, 29,-COS1_2 , 1);
/* pass 1 */
BF0( 5, 26, COS0_5 , 1);
BF0(10, 21, COS0_10, 1);
/* pass 2 */
BF( 5, 10, COS1_5 , 2);
BF(21, 26,-COS1_5 , 2);
/* pass 3 */
BF( 2, 5, COS2_2 , 1);
BF(10, 13,-COS2_2 , 1);
BF(18, 21, COS2_2 , 1);
BF(26, 29,-COS2_2 , 1);
/* pass 4 */
BF( 1, 2, COS3_1 , 2);
BF( 5, 6,-COS3_1 , 2);
BF( 9, 10, COS3_1 , 2);
BF(13, 14,-COS3_1 , 2);
BF(17, 18, COS3_1 , 2);
BF(21, 22,-COS3_1 , 2);
BF(25, 26, COS3_1 , 2);
BF(29, 30,-COS3_1 , 2);
/* pass 5 */
BF1( 0, 1, 2, 3);
BF2( 4, 5, 6, 7);
BF1( 8, 9, 10, 11);
BF2(12, 13, 14, 15);
BF1(16, 17, 18, 19);
BF2(20, 21, 22, 23);
BF1(24, 25, 26, 27);
BF2(28, 29, 30, 31);
/* pass 6 */
ADD( 8, 12);
ADD(12, 10);
ADD(10, 14);
ADD(14, 9);
ADD( 9, 13);
ADD(13, 11);
ADD(11, 15);
out[ 0] = val0;
out[16] = val1;
out[ 8] = val2;
out[24] = val3;
out[ 4] = val4;
out[20] = val5;
out[12] = val6;
out[28] = val7;
out[ 2] = val8;
out[18] = val9;
out[10] = val10;
out[26] = val11;
out[ 6] = val12;
out[22] = val13;
out[14] = val14;
out[30] = val15;
ADD(24, 28);
ADD(28, 26);
ADD(26, 30);
ADD(30, 25);
ADD(25, 29);
ADD(29, 27);
ADD(27, 31);
out[ 1] = val16 + val24;
out[17] = val17 + val25;
out[ 9] = val18 + val26;
out[25] = val19 + val27;
out[ 5] = val20 + val28;
out[21] = val21 + val29;
out[13] = val22 + val30;
out[29] = val23 + val31;
out[ 3] = val24 + val20;
out[19] = val25 + val21;
out[11] = val26 + val22;
out[27] = val27 + val23;
out[ 7] = val28 + val18;
out[23] = val29 + val19;
out[15] = val30 + val17;
out[31] = val31;
}
#if CONFIG_FLOAT
static inline float round_sample(float *sum)
......
libavcodec/mpegaudiodec_float.c
View file @ 06d01188
......@@ -34,7 +34,7 @@ void ff_mpa_synth_filter_float(MPADecodeContext *s, float *synth_buf_ptr,
offset = *synth_buf_offset;
synth_buf = synth_buf_ptr + offset;
dct32(synth_buf, sb_samples);
s->dct.dct32(synth_buf, sb_samples);
s->apply_window_mp3(synth_buf, window, dither_state, samples, incr);
offset = (offset - 32) & 511;
......
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