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aacdec: move spectrum decode and dequantization to a new file

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This commit is contained in:
Lynne 2024-03-16 02:43:33 +01:00
parent 5c026e6637
commit 41ae2b03a5
Signed by: Lynne
GPG key ID: A2FEA5F03F034464
10 changed files with 626 additions and 531 deletions
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4
libavcodec/aac/aacdec.c
View file

@ -48,6 +48,9 @@
extern const AACDecDSP aac_dsp;
extern const AACDecDSP aac_dsp_fixed;
extern const AACDecProc aac_proc;
extern const AACDecProc aac_proc_fixed;
av_cold int ff_aac_decode_close(AVCodecContext *avctx)
{
AACDecContext *ac = avctx->priv_data;
@ -119,6 +122,7 @@ av_cold int ff_aac_decode_init_common(AVCodecContext *avctx)
return ret;
ac->dsp = is_fixed ? aac_dsp_fixed : aac_dsp;
ac->proc = is_fixed ? aac_proc_fixed : aac_proc;
ac->dsp.init_tables();

2
libavcodec/aac/aacdec_dsp_template.c
View file

@ -151,7 +151,7 @@ static void AAC_RENAME(apply_intensity_stereo)(AACDecContext *ac,
scale = c * sce1->AAC_RENAME(sf)[idx];
for (group = 0; group < ics->group_len[g]; group++)
#if USE_FIXED
ac->subband_scale(coef1 + group * 128 + offsets[i],
subband_scale(coef1 + group * 128 + offsets[i],
coef0 + group * 128 + offsets[i],
scale,
23,

5
libavcodec/aac/aacdec_fixed.c
View file

@ -38,6 +38,7 @@
#include "libavcodec/aactab.h"
#include "libavcodec/sinewin_fixed_tablegen.h"
#include "libavcodec/kbdwin.h"
#include "libavcodec/cbrt_data.h"
DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME2(aac_kbd_long_1024))[1024];
DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME2(aac_kbd_short_128))[128];
@ -61,4 +62,8 @@ static void init_tables_fixed(void)
ff_thread_once(&init_fixed_once, init_tables_fixed_fn);
}
/** Dequantization-related */
#include "aacdec_fixed_dequant.h"
#include "aacdec_dsp_template.c"
#include "aacdec_proc_template.c"

174
libavcodec/aac/aacdec_fixed_dequant.h Normal file
View file

@ -0,0 +1,174 @@
/*
* AAC decoder
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
* Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
*
* AAC LATM decoder
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
*
* AAC decoder fixed-point implementation
* Copyright (c) 2013
* MIPS Technologies, Inc., California.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H
#define AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H
#include "aacdec_tab.h"
static void inline vector_pow43(int *coefs, int len)
{
int i, coef;
for (i=0; i<len; i++) {
coef = coefs[i];
if (coef < 0)
coef = -(int)ff_cbrt_tab_fixed[(-coef) & 8191];
else
coef = (int)ff_cbrt_tab_fixed[ coef & 8191];
coefs[i] = coef;
}
}
/* 2^0, 2^0.25, 2^0.5, 2^0.75 */
static const int exp2tab[4] = {
Q31(1.0000000000/2), Q31(1.1892071150/2),
Q31(1.4142135624/2), Q31(1.6817928305/2)
};
static void inline subband_scale(int *dst, int *src, int scale,
int offset, int len, void *log_context)
{
int ssign = scale < 0 ? -1 : 1;
int s = FFABS(scale);
unsigned int round;
int i, out, c = exp2tab[s & 3];
s = offset - (s >> 2);
if (s > 31) {
for (i=0; i<len; i++) {
dst[i] = 0;
}
} else if (s > 0) {
round = 1 << (s-1);
for (i=0; i<len; i++) {
out = (int)(((int64_t)src[i] * c) >> 32);
dst[i] = ((int)(out+round) >> s) * ssign;
}
} else if (s > -32) {
s = s + 32;
round = 1U << (s-1);
for (i=0; i<len; i++) {
out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
dst[i] = out * (unsigned)ssign;
}
} else {
av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
}
}
static void noise_scale(int *coefs, int scale, int band_energy, int len)
{
int s = -scale;
unsigned int round;
int i, out, c = exp2tab[s & 3];
int nlz = 0;
av_assert0(s >= 0);
while (band_energy > 0x7fff) {
band_energy >>= 1;
nlz++;
}
c /= band_energy;
s = 21 + nlz - (s >> 2);
if (s > 31) {
for (i=0; i<len; i++) {
coefs[i] = 0;
}
} else if (s >= 0) {
round = s ? 1 << (s-1) : 0;
for (i=0; i<len; i++) {
out = (int)(((int64_t)coefs[i] * c) >> 32);
coefs[i] = -((int)(out+round) >> s);
}
}
else {
s = s + 32;
if (s > 0) {
round = 1 << (s-1);
for (i=0; i<len; i++) {
out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
coefs[i] = -out;
}
} else {
for (i=0; i<len; i++)
coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
}
}
}
static inline int *DEC_SPAIR(int *dst, unsigned idx)
{
dst[0] = (idx & 15) - 4;
dst[1] = (idx >> 4 & 15) - 4;
return dst + 2;
}
static inline int *DEC_SQUAD(int *dst, unsigned idx)
{
dst[0] = (idx & 3) - 1;
dst[1] = (idx >> 2 & 3) - 1;
dst[2] = (idx >> 4 & 3) - 1;
dst[3] = (idx >> 6 & 3) - 1;
return dst + 4;
}
static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
{
dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
return dst + 2;
}
static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
{
unsigned nz = idx >> 12;
dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
return dst + 4;
}
#endif /* AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H */

74
libavcodec/aac/aacdec_float.c
View file

@ -38,6 +38,7 @@
#include "libavcodec/aactab.h"
#include "libavcodec/sinewin.h"
#include "libavcodec/kbdwin.h"
#include "libavcodec/cbrt_data.h"
DECLARE_ALIGNED(32, static float, sine_120)[120];
DECLARE_ALIGNED(32, static float, sine_960)[960];
@ -63,4 +64,77 @@ static void init_tables(void)
ff_thread_once(&init_float_once, init_tables_float_fn);
}
/** Dequantization-related **/
#include "aacdec_tab.h"
#include "libavutil/intfloat.h"
#ifndef VMUL2
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
const float *scale)
{
float s = *scale;
*dst++ = v[idx & 15] * s;
*dst++ = v[idx>>4 & 15] * s;
return dst;
}
#endif
#ifndef VMUL4
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
const float *scale)
{
float s = *scale;
*dst++ = v[idx & 3] * s;
*dst++ = v[idx>>2 & 3] * s;
*dst++ = v[idx>>4 & 3] * s;
*dst++ = v[idx>>6 & 3] * s;
return dst;
}
#endif
#ifndef VMUL2S
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
unsigned sign, const float *scale)
{
union av_intfloat32 s0, s1;
s0.f = s1.f = *scale;
s0.i ^= sign >> 1 << 31;
s1.i ^= sign << 31;
*dst++ = v[idx & 15] * s0.f;
*dst++ = v[idx>>4 & 15] * s1.f;
return dst;
}
#endif
#ifndef VMUL4S
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
unsigned sign, const float *scale)
{
unsigned nz = idx >> 12;
union av_intfloat32 s = { .f = *scale };
union av_intfloat32 t;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>2 & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>4 & 3] * t.f;
sign <<= nz & 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>6 & 3] * t.f;
return dst;
}
#endif
#include "aacdec_dsp_template.c"
#include "aacdec_proc_template.c"

354
libavcodec/aac/aacdec_proc_template.c Normal file
View file

@ -0,0 +1,354 @@
/*
* AAC decoder
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
* Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
*
* AAC LATM decoder
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
*
* AAC decoder fixed-point implementation
* Copyright (c) 2013
* MIPS Technologies, Inc., California.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* linear congruential pseudorandom number generator
*
* @param previous_val pointer to the current state of the generator
*
* @return Returns a 32-bit pseudorandom integer
*/
static av_always_inline int lcg_random(unsigned previous_val)
{
union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
return v.s;
}
/**
* Decode spectral data; reference: table 4.50.
* Dequantize and scale spectral data; reference: 4.6.3.3.
*
* @param coef array of dequantized, scaled spectral data
* @param sf array of scalefactors or intensity stereo positions
* @param pulse_present set if pulses are present
* @param pulse pointer to pulse data struct
* @param band_type array of the used band type
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int AAC_RENAME(decode_spectrum_and_dequant)(AACDecContext *ac,
GetBitContext *gb,
const Pulse *pulse,
SingleChannelElement *sce)
{
int i, k, g, idx = 0;
INTFLOAT *coef = sce->AAC_RENAME(coeffs);
IndividualChannelStream *ics = &sce->ics;
const int c = 1024 / ics->num_windows;
const uint16_t *offsets = ics->swb_offset;
const INTFLOAT *sf = sce->AAC_RENAME(sf);
const enum BandType *band_type = sce->band_type;
INTFLOAT *coef_base = coef;
for (g = 0; g < ics->num_windows; g++)
memset(coef + g * 128 + offsets[ics->max_sfb], 0,
sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
for (g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (i = 0; i < ics->max_sfb; i++, idx++) {
const unsigned cbt_m1 = band_type[idx] - 1;
INTFLOAT *cfo = coef + offsets[i];
int off_len = offsets[i + 1] - offsets[i];
int group;
if (cbt_m1 >= INTENSITY_BT2 - 1) {
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
memset(cfo, 0, off_len * sizeof(*cfo));
}
} else if (cbt_m1 == NOISE_BT - 1) {
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT band_energy;
#if USE_FIXED
for (k = 0; k < off_len; k++) {
ac->random_state = lcg_random(ac->random_state);
cfo[k] = ac->random_state >> 3;
}
band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
band_energy = fixed_sqrt(band_energy, 31);
noise_scale(cfo, sf[idx], band_energy, off_len);
#else
float scale;
for (k = 0; k < off_len; k++) {
ac->random_state = lcg_random(ac->random_state);
cfo[k] = ac->random_state;
}
band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
scale = sf[idx] / sqrtf(band_energy);
ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
#endif /* USE_FIXED */
}
} else {
#if !USE_FIXED
const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
#endif /* !USE_FIXED */
const VLCElem *vlc_tab = ff_vlc_spectral[cbt_m1];
OPEN_READER(re, gb);
switch (cbt_m1 >> 1) {
case 0:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned cb_idx;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = code;
#if USE_FIXED
cf = DEC_SQUAD(cf, cb_idx);
#else
cf = VMUL4(cf, vq, cb_idx, sf + idx);
#endif /* USE_FIXED */
} while (len -= 4);
}
break;
case 1:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned nnz;
unsigned cb_idx;
uint32_t bits;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = code;
nnz = cb_idx >> 8 & 15;
bits = nnz ? GET_CACHE(re, gb) : 0;
LAST_SKIP_BITS(re, gb, nnz);
#if USE_FIXED
cf = DEC_UQUAD(cf, cb_idx, bits);
#else
cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
#endif /* USE_FIXED */
} while (len -= 4);
}
break;
case 2:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned cb_idx;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = code;
#if USE_FIXED
cf = DEC_SPAIR(cf, cb_idx);
#else
cf = VMUL2(cf, vq, cb_idx, sf + idx);
#endif /* USE_FIXED */
} while (len -= 2);
}
break;
case 3:
case 4:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned nnz;
unsigned cb_idx;
unsigned sign;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = code;
nnz = cb_idx >> 8 & 15;
sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
LAST_SKIP_BITS(re, gb, nnz);
#if USE_FIXED
cf = DEC_UPAIR(cf, cb_idx, sign);
#else
cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
#endif /* USE_FIXED */
} while (len -= 2);
}
break;
default:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
#if USE_FIXED
int *icf = cfo;
int v;
#else
float *cf = cfo;
uint32_t *icf = (uint32_t *) cf;
#endif /* USE_FIXED */
int len = off_len;
do {
int code;
unsigned nzt, nnz;
unsigned cb_idx;
uint32_t bits;
int j;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = code;
if (cb_idx == 0x0000) {
*icf++ = 0;
*icf++ = 0;
continue;
}
nnz = cb_idx >> 12;
nzt = cb_idx >> 8;
bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
LAST_SKIP_BITS(re, gb, nnz);
for (j = 0; j < 2; j++) {
if (nzt & 1<<j) {
uint32_t b;
int n;
/* The total length of escape_sequence must be < 22 bits according
to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
UPDATE_CACHE(re, gb);
b = GET_CACHE(re, gb);
b = 31 - av_log2(~b);
if (b > 8) {
av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
return AVERROR_INVALIDDATA;
}
SKIP_BITS(re, gb, b + 1);
b += 4;
n = (1 << b) + SHOW_UBITS(re, gb, b);
LAST_SKIP_BITS(re, gb, b);
#if USE_FIXED
v = n;
if (bits & 1U<<31)
v = -v;
*icf++ = v;
#else
*icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
#endif /* USE_FIXED */
bits <<= 1;
} else {
#if USE_FIXED
v = cb_idx & 15;
if (bits & 1U<<31)
v = -v;
*icf++ = v;
#else
unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
*icf++ = (bits & 1U<<31) | v;
#endif /* USE_FIXED */
bits <<= !!v;
}
cb_idx >>= 4;
}
} while (len -= 2);
#if !USE_FIXED
ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
#endif /* !USE_FIXED */
}
}
CLOSE_READER(re, gb);
}
}
coef += g_len << 7;
}
if (pulse) {
idx = 0;
for (i = 0; i < pulse->num_pulse; i++) {
INTFLOAT co = coef_base[ pulse->pos[i] ];
while (offsets[idx + 1] <= pulse->pos[i])
idx++;
if (band_type[idx] != NOISE_BT && sf[idx]) {
INTFLOAT ico = -pulse->amp[i];
#if USE_FIXED
if (co) {
ico = co + (co > 0 ? -ico : ico);
}
coef_base[ pulse->pos[i] ] = ico;
#else
if (co) {
co /= sf[idx];
ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
}
coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
#endif /* USE_FIXED */
}
}
}
#if USE_FIXED
coef = coef_base;
idx = 0;
for (g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (i = 0; i < ics->max_sfb; i++, idx++) {
const unsigned cbt_m1 = band_type[idx] - 1;
int *cfo = coef + offsets[i];
int off_len = offsets[i + 1] - offsets[i];
int group;
if (cbt_m1 < NOISE_BT - 1) {
for (group = 0; group < (int)g_len; group++, cfo+=128) {
vector_pow43(cfo, off_len);
subband_scale(cfo, cfo, sf[idx], 34, off_len, ac->avctx);
}
}
}
coef += g_len << 7;
}
#endif /* USE_FIXED */
return 0;
}
const AACDecProc AAC_RENAME(aac_proc) = {
.decode_spectrum_and_dequant = AAC_RENAME(decode_spectrum_and_dequant),
};

68
libavcodec/aacdec.c
View file

@ -78,74 +78,6 @@ static av_always_inline void reset_predict_state(PredictorState *ps)
ps->var1 = 1.0f;
}
#ifndef VMUL2
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
const float *scale)
{
float s = *scale;
*dst++ = v[idx & 15] * s;
*dst++ = v[idx>>4 & 15] * s;
return dst;
}
#endif
#ifndef VMUL4
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
const float *scale)
{
float s = *scale;
*dst++ = v[idx & 3] * s;
*dst++ = v[idx>>2 & 3] * s;
*dst++ = v[idx>>4 & 3] * s;
*dst++ = v[idx>>6 & 3] * s;
return dst;
}
#endif
#ifndef VMUL2S
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
unsigned sign, const float *scale)
{
union av_intfloat32 s0, s1;
s0.f = s1.f = *scale;
s0.i ^= sign >> 1 << 31;
s1.i ^= sign << 31;
*dst++ = v[idx & 15] * s0.f;
*dst++ = v[idx>>4 & 15] * s1.f;
return dst;
}
#endif
#ifndef VMUL4S
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
unsigned sign, const float *scale)
{
unsigned nz = idx >> 12;
union av_intfloat32 s = { .f = *scale };
union av_intfloat32 t;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>2 & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>4 & 3] * t.f;
sign <<= nz & 1;
t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>6 & 3] * t.f;
return dst;
}
#endif
static av_always_inline float flt16_round(float pf)
{
union av_intfloat32 tmp;

15
libavcodec/aacdec.h
View file

@ -200,6 +200,16 @@ typedef struct DynamicRangeControl {
*/
} DynamicRangeControl;
/**
* Decode-specific primitives
*/
typedef struct AACDecProc {
int (*decode_spectrum_and_dequant)(AACDecContext *ac,
GetBitContext *gb,
const Pulse *pulse,
SingleChannelElement *sce);
} AACDecProc;
/**
* DSP-specific primitives
*/
@ -232,6 +242,7 @@ struct AACDecContext {
struct AVCodecContext *avctx;
AACDecDSP dsp;
AACDecProc proc;
struct AVFrame *frame;
@ -309,10 +320,6 @@ struct AACDecContext {
int warned_he_aac_mono;
int is_fixed;
/* aacdec functions pointers */
void (*vector_pow43)(int *coefs, int len);
void (*subband_scale)(int *dst, int *src, int scale, int offset, int len, void *log_context);
};
#if defined(USE_FIXED) && USE_FIXED

130
libavcodec/aacdec_fixed.c
View file

@ -147,136 +147,6 @@ static av_always_inline void reset_predict_state(PredictorState *ps)
static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) }; // 2^0, 2^0.25, 2^0.5, 2^0.75
static inline int *DEC_SPAIR(int *dst, unsigned idx)
{
dst[0] = (idx & 15) - 4;
dst[1] = (idx >> 4 & 15) - 4;
return dst + 2;
}
static inline int *DEC_SQUAD(int *dst, unsigned idx)
{
dst[0] = (idx & 3) - 1;
dst[1] = (idx >> 2 & 3) - 1;
dst[2] = (idx >> 4 & 3) - 1;
dst[3] = (idx >> 6 & 3) - 1;
return dst + 4;
}
static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
{
dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
return dst + 2;
}
static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
{
unsigned nz = idx >> 12;
dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
return dst + 4;
}
static void vector_pow43(int *coefs, int len)
{
int i, coef;
for (i=0; i<len; i++) {
coef = coefs[i];
if (coef < 0)
coef = -(int)ff_cbrt_tab_fixed[(-coef) & 8191];
else
coef = (int)ff_cbrt_tab_fixed[ coef & 8191];
coefs[i] = coef;
}
}
static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
{
int ssign = scale < 0 ? -1 : 1;
int s = FFABS(scale);
unsigned int round;
int i, out, c = exp2tab[s & 3];
s = offset - (s >> 2);
if (s > 31) {
for (i=0; i<len; i++) {
dst[i] = 0;
}
} else if (s > 0) {
round = 1 << (s-1);
for (i=0; i<len; i++) {
out = (int)(((int64_t)src[i] * c) >> 32);
dst[i] = ((int)(out+round) >> s) * ssign;
}
} else if (s > -32) {
s = s + 32;
round = 1U << (s-1);
for (i=0; i<len; i++) {
out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
dst[i] = out * (unsigned)ssign;
}
} else {
av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
}
}
static void noise_scale(int *coefs, int scale, int band_energy, int len)
{
int s = -scale;
unsigned int round;
int i, out, c = exp2tab[s & 3];
int nlz = 0;
av_assert0(s >= 0);
while (band_energy > 0x7fff) {
band_energy >>= 1;
nlz++;
}
c /= band_energy;
s = 21 + nlz - (s >> 2);
if (s > 31) {
for (i=0; i<len; i++) {
coefs[i] = 0;
}
} else if (s >= 0) {
round = s ? 1 << (s-1) : 0;
for (i=0; i<len; i++) {
out = (int)(((int64_t)coefs[i] * c) >> 32);
coefs[i] = -((int)(out+round) >> s);
}
}
else {
s = s + 32;
if (s > 0) {
round = 1 << (s-1);
for (i=0; i<len; i++) {
out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
coefs[i] = -out;
}
} else {
for (i=0; i<len; i++)
coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
}
}
}
static av_always_inline SoftFloat flt16_round(SoftFloat pf)
{
SoftFloat tmp;

331
libavcodec/aacdec_template.c
View file

@ -1072,19 +1072,6 @@ static int decode_audio_specific_config(AACDecContext *ac,
sync_extension);
}
/**
* linear congruential pseudorandom number generator
*
* @param previous_val pointer to the current state of the generator
*
* @return Returns a 32-bit pseudorandom integer
*/
static av_always_inline int lcg_random(unsigned previous_val)
{
union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
return v.s;
}
static void reset_all_predictors(PredictorState *ps)
{
int i;
@ -1598,313 +1585,6 @@ static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
}
}
/**
* Decode spectral data; reference: table 4.50.
* Dequantize and scale spectral data; reference: 4.6.3.3.
*
* @param coef array of dequantized, scaled spectral data
* @param sf array of scalefactors or intensity stereo positions
* @param pulse_present set if pulses are present
* @param pulse pointer to pulse data struct
* @param band_type array of the used band type
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_spectrum_and_dequant(AACDecContext *ac,
GetBitContext *gb,
const Pulse *pulse,
SingleChannelElement *sce)
{
int i, k, g, idx = 0;
INTFLOAT *coef = sce->AAC_RENAME(coeffs);
IndividualChannelStream *ics = &sce->ics;
const int c = 1024 / ics->num_windows;
const uint16_t *offsets = ics->swb_offset;
const INTFLOAT *sf = sce->AAC_RENAME(sf);
const enum BandType *band_type = sce->band_type;
INTFLOAT *coef_base = coef;
for (g = 0; g < ics->num_windows; g++)
memset(coef + g * 128 + offsets[ics->max_sfb], 0,
sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
for (g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (i = 0; i < ics->max_sfb; i++, idx++) {
const unsigned cbt_m1 = band_type[idx] - 1;
INTFLOAT *cfo = coef + offsets[i];
int off_len = offsets[i + 1] - offsets[i];
int group;
if (cbt_m1 >= INTENSITY_BT2 - 1) {
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
memset(cfo, 0, off_len * sizeof(*cfo));
}
} else if (cbt_m1 == NOISE_BT - 1) {
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT band_energy;
#if USE_FIXED
for (k = 0; k < off_len; k++) {
ac->random_state = lcg_random(ac->random_state);
cfo[k] = ac->random_state >> 3;
}
band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
band_energy = fixed_sqrt(band_energy, 31);
noise_scale(cfo, sf[idx], band_energy, off_len);
#else
float scale;
for (k = 0; k < off_len; k++) {
ac->random_state = lcg_random(ac->random_state);
cfo[k] = ac->random_state;
}
band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
scale = sf[idx] / sqrtf(band_energy);
ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
#endif /* USE_FIXED */
}
} else {
#if !USE_FIXED
const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
#endif /* !USE_FIXED */
const VLCElem *vlc_tab = ff_vlc_spectral[cbt_m1];
OPEN_READER(re, gb);
switch (cbt_m1 >> 1) {
case 0:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned cb_idx;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = code;
#if USE_FIXED
cf = DEC_SQUAD(cf, cb_idx);
#else
cf = VMUL4(cf, vq, cb_idx, sf + idx);
#endif /* USE_FIXED */
} while (len -= 4);
}
break;
case 1:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned nnz;
unsigned cb_idx;
uint32_t bits;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = code;
nnz = cb_idx >> 8 & 15;
bits = nnz ? GET_CACHE(re, gb) : 0;
LAST_SKIP_BITS(re, gb, nnz);
#if USE_FIXED
cf = DEC_UQUAD(cf, cb_idx, bits);
#else
cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
#endif /* USE_FIXED */
} while (len -= 4);
}
break;
case 2:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned cb_idx;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = code;
#if USE_FIXED
cf = DEC_SPAIR(cf, cb_idx);
#else
cf = VMUL2(cf, vq, cb_idx, sf + idx);
#endif /* USE_FIXED */
} while (len -= 2);
}
break;
case 3:
case 4:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
int code;
unsigned nnz;
unsigned cb_idx;
unsigned sign;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = code;
nnz = cb_idx >> 8 & 15;
sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
LAST_SKIP_BITS(re, gb, nnz);
#if USE_FIXED
cf = DEC_UPAIR(cf, cb_idx, sign);
#else
cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
#endif /* USE_FIXED */
} while (len -= 2);
}
break;
default:
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
#if USE_FIXED
int *icf = cfo;
int v;
#else
float *cf = cfo;
uint32_t *icf = (uint32_t *) cf;
#endif /* USE_FIXED */
int len = off_len;
do {
int code;
unsigned nzt, nnz;
unsigned cb_idx;
uint32_t bits;
int j;
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = code;
if (cb_idx == 0x0000) {
*icf++ = 0;
*icf++ = 0;
continue;
}
nnz = cb_idx >> 12;
nzt = cb_idx >> 8;
bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
LAST_SKIP_BITS(re, gb, nnz);
for (j = 0; j < 2; j++) {
if (nzt & 1<<j) {
uint32_t b;
int n;
/* The total length of escape_sequence must be < 22 bits according
to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
UPDATE_CACHE(re, gb);
b = GET_CACHE(re, gb);
b = 31 - av_log2(~b);
if (b > 8) {
av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
return AVERROR_INVALIDDATA;
}
SKIP_BITS(re, gb, b + 1);
b += 4;
n = (1 << b) + SHOW_UBITS(re, gb, b);
LAST_SKIP_BITS(re, gb, b);
#if USE_FIXED
v = n;
if (bits & 1U<<31)
v = -v;
*icf++ = v;
#else
*icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
#endif /* USE_FIXED */
bits <<= 1;
} else {
#if USE_FIXED
v = cb_idx & 15;
if (bits & 1U<<31)
v = -v;
*icf++ = v;
#else
unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
*icf++ = (bits & 1U<<31) | v;
#endif /* USE_FIXED */
bits <<= !!v;
}
cb_idx >>= 4;
}
} while (len -= 2);
#if !USE_FIXED
ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
#endif /* !USE_FIXED */
}
}
CLOSE_READER(re, gb);
}
}
coef += g_len << 7;
}
if (pulse) {
idx = 0;
for (i = 0; i < pulse->num_pulse; i++) {
INTFLOAT co = coef_base[ pulse->pos[i] ];
while (offsets[idx + 1] <= pulse->pos[i])
idx++;
if (band_type[idx] != NOISE_BT && sf[idx]) {
INTFLOAT ico = -pulse->amp[i];
#if USE_FIXED
if (co) {
ico = co + (co > 0 ? -ico : ico);
}
coef_base[ pulse->pos[i] ] = ico;
#else
if (co) {
co /= sf[idx];
ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
}
coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
#endif /* USE_FIXED */
}
}
}
#if USE_FIXED
coef = coef_base;
idx = 0;
for (g = 0; g < ics->num_window_groups; g++) {
unsigned g_len = ics->group_len[g];
for (i = 0; i < ics->max_sfb; i++, idx++) {
const unsigned cbt_m1 = band_type[idx] - 1;
int *cfo = coef + offsets[i];
int off_len = offsets[i + 1] - offsets[i];
int group;
if (cbt_m1 < NOISE_BT - 1) {
for (group = 0; group < (int)g_len; group++, cfo+=128) {
ac->vector_pow43(cfo, off_len);
ac->subband_scale(cfo, cfo, sf[idx], 34, off_len, ac->avctx);
}
}
}
coef += g_len << 7;
}
#endif /* USE_FIXED */
return 0;
}
/**
* Apply AAC-Main style frequency domain prediction.
*/
@ -2047,9 +1727,9 @@ static int decode_ics(AACDecContext *ac, SingleChannelElement *sce,
}
}
ret = decode_spectrum_and_dequant(ac, gb,
pulse_present ? &pulse : NULL,
sce);
ret = ac->proc.decode_spectrum_and_dequant(ac, gb,
pulse_present ? &pulse : NULL,
sce);
if (ret < 0)
goto fail;
@ -2892,11 +2572,6 @@ static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame,
static void aacdec_init(AACDecContext *c)
{
#if USE_FIXED
c->vector_pow43 = vector_pow43;
c->subband_scale = subband_scale;
#endif
#if !USE_FIXED
#if ARCH_MIPS
ff_aacdec_init_mips(c);