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avcodec: Implementation of AAC_fixed_decoder (SBR-module) 

Add fixed poind code.

Signed-off-by: Nedeljko Babic <nedeljko.babic@imgtec.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
tags/n2.8
Djordje Pesut Michael Niedermayer 10 years ago
parent
b0414da90d
commit
f85bc147fb
15 changed files with 1237 additions and 190 deletions
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  1. +3
    -2
      libavcodec/Makefile
  2. +1
    -51
      libavcodec/aac.h
  3. +78
    -0
      libavcodec/aac_defines.h
  4. +7
    -7
      libavcodec/aacdec_template.c
  5. +1
    -0
      libavcodec/aacsbr.c
  6. +6
    -6
      libavcodec/aacsbr.h
  7. +586
    -0
      libavcodec/aacsbr_fixed.c
  8. +168
    -43
      libavcodec/aacsbr_template.c
  9. +10
    -5
      libavcodec/lpc.h
  10. +39
    -39
      libavcodec/sbr.h
  11. +3
    -0
      libavcodec/sbrdsp.c
  12. +19
    -17
      libavcodec/sbrdsp.h
  13. +286
    -0
      libavcodec/sbrdsp_fixed.c
  14. +22
    -20
      libavcodec/sbrdsp_template.c
  15. +8
    -0
      libavutil/softfloat.h

+ 3
- 2
libavcodec/Makefile View File

@@ -126,8 +126,9 @@ OBJS-$(CONFIG_A64MULTI5_ENCODER) += a64multienc.o elbg.o
OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps.o \
aacadtsdec.o mpeg4audio.o kbdwin.o \
sbrdsp.o aacpsdsp.o
OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o \
aacadtsdec.o mpeg4audio.o kbdwin.o
OBJS-$(CONFIG_AAC_FIXED_DECODER) += aacdec_fixed.o aactab.o aacsbr_fixed.o \
aacadtsdec.o mpeg4audio.o kbdwin.o \
sbrdsp_fixed.o
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
aacpsy.o aactab.o \
psymodel.o mpeg4audio.o kbdwin.o


+ 1
- 51
libavcodec/aac.h View File

@@ -30,58 +30,8 @@
#ifndef AVCODEC_AAC_H
#define AVCODEC_AAC_H

#ifndef USE_FIXED
#define USE_FIXED 0
#endif

#if USE_FIXED

#include "libavutil/softfloat.h"

#define FFT_FLOAT 0
#define FFT_FIXED_32 1

#define AAC_RENAME(x) x ## _fixed
#define AAC_RENAME_32(x) x ## _fixed_32
#define AAC_FLOAT SoftFloat
#define INTFLOAT int
#define SHORTFLOAT int16_t
#define AAC_SIGNE int
#define FIXR(a) ((int)((a) * 1 + 0.5))
#define FIXR10(a) ((int)((a) * 1024.0 + 0.5))
#define Q23(a) (int)((a) * 8388608.0 + 0.5)
#define Q30(x) (int)((x)*1073741824.0 + 0.5)
#define Q31(x) (int)((x)*2147483648.0 + 0.5)
#define RANGE15(x) x
#define GET_GAIN(x, y) (-(y) << (x)) + 1024
#define AAC_MUL26(x, y) (int)(((int64_t)(x) * (y) + 0x2000000) >> 26)
#define AAC_MUL30(x, y) (int)(((int64_t)(x) * (y) + 0x20000000) >> 30)
#define AAC_MUL31(x, y) (int)(((int64_t)(x) * (y) + 0x40000000) >> 31)

#else

#define FFT_FLOAT 1
#define FFT_FIXED_32 0

#define AAC_RENAME(x) x
#define AAC_RENAME_32(x) x
#define AAC_FLOAT float
#define INTFLOAT float
#define SHORTFLOAT float
#define AAC_SIGNE unsigned
#define FIXR(x) ((float)(x))
#define FIXR10(x) ((float)(x))
#define Q23(x) x
#define Q30(x) x
#define Q31(x) x
#define RANGE15(x) (32768.0 * (x))
#define GET_GAIN(x, y) powf((x), -(y))
#define AAC_MUL26(x, y) ((x) * (y))
#define AAC_MUL30(x, y) ((x) * (y))
#define AAC_MUL31(x, y) ((x) * (y))

#endif /* USE_FIXED */

#include "aac_defines.h"
#include "libavutil/float_dsp.h"
#include "libavutil/fixed_dsp.h"
#include "avcodec.h"


+ 78
- 0
libavcodec/aac_defines.h View File

@@ -0,0 +1,78 @@
/*
* AAC defines
*
* 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_DEFINES_H
#define AVCODEC_AAC_DEFINES_H

#ifndef USE_FIXED
#define USE_FIXED 0
#endif

#if USE_FIXED

#include "libavutil/softfloat.h"

#define FFT_FLOAT 0
#define FFT_FIXED_32 1

#define AAC_RENAME(x) x ## _fixed
#define AAC_RENAME_32(x) x ## _fixed_32
#define INTFLOAT int
#define SHORTFLOAT int16_t
#define AAC_FLOAT SoftFloat
#define AAC_SIGNE int
#define FIXR(a) ((int)((a) * 1 + 0.5))
#define FIXR10(a) ((int)((a) * 1024.0 + 0.5))
#define Q23(a) (int)((a) * 8388608.0 + 0.5)
#define Q30(x) (int)((x)*1073741824.0 + 0.5)
#define Q31(x) (int)((x)*2147483648.0 + 0.5)
#define RANGE15(x) x
#define GET_GAIN(x, y) (-(y) << (x)) + 1024
#define AAC_MUL26(x, y) (int)(((int64_t)(x) * (y) + 0x2000000) >> 26)
#define AAC_MUL30(x, y) (int)(((int64_t)(x) * (y) + 0x20000000) >> 30)
#define AAC_MUL31(x, y) (int)(((int64_t)(x) * (y) + 0x40000000) >> 31)
#define AAC_SRA_R(x, y) (int)(((x) + (1 << ((y) - 1))) >> (y))

#else

#define FFT_FLOAT 1
#define FFT_FIXED_32 0

#define AAC_RENAME(x) x
#define AAC_RENAME_32(x) x
#define INTFLOAT float
#define SHORTFLOAT float
#define AAC_FLOAT float
#define AAC_SIGNE unsigned
#define FIXR(x) ((float)(x))
#define FIXR10(x) ((float)(x))
#define Q23(x) x
#define Q30(x) x
#define Q31(x) x
#define RANGE15(x) (32768.0 * (x))
#define GET_GAIN(x, y) powf((x), -(y))
#define AAC_MUL26(x, y) ((x) * (y))
#define AAC_MUL30(x, y) ((x) * (y))
#define AAC_MUL31(x, y) ((x) * (y))
#define AAC_SRA_R(x, y) (x)

#endif /* USE_FIXED */

#endif /* AVCODEC_AAC_DEFINES_H */

+ 7
- 7
libavcodec/aacdec_template.c View File

@@ -132,7 +132,7 @@ static av_cold int che_configure(AACContext *ac,
if (!ac->che[type][id]) {
if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
return AVERROR(ENOMEM);
ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
AAC_RENAME(ff_aac_sbr_ctx_init)(ac, &ac->che[type][id]->sbr);
}
if (type != TYPE_CCE) {
if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
@@ -147,7 +147,7 @@ static av_cold int che_configure(AACContext *ac,
}
} else {
if (ac->che[type][id])
ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][id]->sbr);
av_freep(&ac->che[type][id]);
}
return 0;
@@ -1126,7 +1126,7 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
AAC_INIT_VLC_STATIC( 9, 366);
AAC_INIT_VLC_STATIC(10, 462);

ff_aac_sbr_init();
AAC_RENAME(ff_aac_sbr_init)();

#if USE_FIXED
ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & CODEC_FLAG_BITEXACT);
@@ -2315,7 +2315,7 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
ac->oc[1].m4ac.sbr = 1;
ac->avctx->profile = FF_PROFILE_AAC_HE;
}
res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
break;
case EXT_DYNAMIC_RANGE:
res = decode_dynamic_range(&ac->che_drc, gb);
@@ -2357,7 +2357,7 @@ static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
continue;

// tns_decode_coef
compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);

start = ics->swb_offset[FFMIN(bottom, mmm)];
end = ics->swb_offset[FFMIN( top, mmm)];
@@ -2738,7 +2738,7 @@ static void spectral_to_sample(AACContext *ac)
ac->update_ltp(ac, &che->ch[1]);
}
if (ac->oc[1].m4ac.sbr > 0) {
ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
}
}
if (type <= TYPE_CCE)
@@ -3153,7 +3153,7 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
for (i = 0; i < MAX_ELEM_ID; i++) {
for (type = 0; type < 4; type++) {
if (ac->che[type][i])
ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][i]->sbr);
av_freep(&ac->che[type][i]);
}
}


+ 1
- 0
libavcodec/aacsbr.c View File

@@ -25,6 +25,7 @@
* AAC Spectral Band Replication decoding functions
* @author Robert Swain ( rob opendot cl )
*/
#define USE_FIXED 0

#include "aac.h"
#include "sbr.h"


+ 6
- 6
libavcodec/aacsbr.h View File

@@ -79,17 +79,17 @@ static const int8_t vlc_sbr_lav[10] =
{ name ## _codes, name ## _bits, sizeof(name ## _codes), sizeof(name ## _codes[0]) }

/** Initialize SBR. */
void ff_aac_sbr_init(void);
void AAC_RENAME(ff_aac_sbr_init)(void);
/** Initialize one SBR context. */
void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr);
void AAC_RENAME(ff_aac_sbr_ctx_init)(AACContext *ac, SpectralBandReplication *sbr);
/** Close one SBR context. */
void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr);
void AAC_RENAME(ff_aac_sbr_ctx_close)(SpectralBandReplication *sbr);
/** Decode one SBR element. */
int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
int AAC_RENAME(ff_decode_sbr_extension)(AACContext *ac, SpectralBandReplication *sbr,
GetBitContext *gb, int crc, int cnt, int id_aac);
/** Apply one SBR element to one AAC element. */
void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
float* L, float *R);
void AAC_RENAME(ff_sbr_apply)(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
INTFLOAT* L, INTFLOAT *R);

void ff_aacsbr_func_ptr_init_mips(AACSBRContext *c);



+ 586
- 0
libavcodec/aacsbr_fixed.c View File

@@ -0,0 +1,586 @@
/*
* Copyright (c) 2013
* MIPS Technologies, Inc., California.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*
* AAC Spectral Band Replication decoding functions (fixed-point)
* Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
* Copyright (c) 2009-2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*/

/**
* @file
* AAC Spectral Band Replication decoding functions (fixed-point)
* Note: Rounding-to-nearest used unless otherwise stated
* @author Robert Swain ( rob opendot cl )
* @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
*/
#define USE_FIXED 1

#include "aac.h"
#include "sbr.h"
#include "aacsbr.h"
#include "aacsbrdata.h"
#include "aacsbr_fixed_tablegen.h"
#include "fft.h"
#include "aacps.h"
#include "sbrdsp.h"
#include "libavutil/internal.h"
#include "libavutil/libm.h"
#include "libavutil/avassert.h"

#include <stdint.h>
#include <float.h>
#include <math.h>

static VLC vlc_sbr[10];
static void aacsbr_func_ptr_init(AACSBRContext *c);
static const int CONST_LN2 = Q31(0.6931471806/256); // ln(2)/256
static const int CONST_RECIP_LN2 = Q31(0.7213475204); // 0.5/ln(2)
static const int CONST_SQRT2 = Q30(0.7071067812); // sqrt(2)/2
static const int CONST_076923 = Q31(0.76923076923076923077f);

int fixed_log_table[10] =
{
Q31(1.0/2), Q31(1.0/3), Q31(1.0/4), Q31(1.0/5), Q31(1.0/6),
Q31(1.0/7), Q31(1.0/8), Q31(1.0/9), Q31(1.0/10), Q31(1.0/11)
};

static int fixed_log(int x)
{
int i, ret, xpow, tmp;

ret = x;
xpow = x;
for (i=0; i<10; i+=2){
xpow = (int)(((int64_t)xpow * x + 0x40000000) >> 31);
tmp = (int)(((int64_t)xpow * fixed_log_table[i] + 0x40000000) >> 31);
ret -= tmp;

xpow = (int)(((int64_t)xpow * x + 0x40000000) >> 31);
tmp = (int)(((int64_t)xpow * fixed_log_table[i+1] + 0x40000000) >> 31);
ret += tmp;
}

return ret;
}

int fixed_exp_table[7] =
{
Q31(1.0/2), Q31(1.0/6), Q31(1.0/24), Q31(1.0/120),
Q31(1.0/720), Q31(1.0/5040), Q31(1.0/40320)
};

static int fixed_exp(int x)
{
int i, ret, xpow, tmp;

ret = 0x800000 + x;
xpow = x;
for (i=0; i<7; i++){
xpow = (int)(((int64_t)xpow * x + 0x400000) >> 23);
tmp = (int)(((int64_t)xpow * fixed_exp_table[i] + 0x40000000) >> 31);
ret += tmp;
}

return ret;
}

static void make_bands(int16_t* bands, int start, int stop, int num_bands)
{
int k, previous, present;
int base, prod, nz = 0;

base = (stop << 23) / start;
while (base < 0x40000000){
base <<= 1;
nz++;
}
base = fixed_log(base - 0x80000000);
base = (((base + 0x80) >> 8) + (8-nz)*CONST_LN2) / num_bands;
base = fixed_exp(base);

previous = start;
prod = start << 23;

for (k = 0; k < num_bands-1; k++) {
prod = (int)(((int64_t)prod * base + 0x400000) >> 23);
present = (prod + 0x400000) >> 23;
bands[k] = present - previous;
previous = present;
}
bands[num_bands-1] = stop - previous;
}

/// Dequantization and stereo decoding (14496-3 sp04 p203)
static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
{
int k, e;
int ch;

if (id_aac == TYPE_CPE && sbr->bs_coupling) {
int alpha = sbr->data[0].bs_amp_res ? 2 : 1;
int pan_offset = sbr->data[0].bs_amp_res ? 12 : 24;
for (e = 1; e <= sbr->data[0].bs_num_env; e++) {
for (k = 0; k < sbr->n[sbr->data[0].bs_freq_res[e]]; k++) {
SoftFloat temp1, temp2, fac;

temp1.exp = sbr->data[0].env_facs[e][k].mant * alpha + 14;
if (temp1.exp & 1)
temp1.mant = 759250125;
else
temp1.mant = 0x20000000;
temp1.exp = (temp1.exp >> 1) + 1;

temp2.exp = (pan_offset - sbr->data[1].env_facs[e][k].mant) * alpha;
if (temp2.exp & 1)
temp2.mant = 759250125;
else
temp2.mant = 0x20000000;
temp2.exp = (temp2.exp >> 1) + 1;
fac = av_div_sf(temp1, av_add_sf(FLOAT_1, temp2));
sbr->data[0].env_facs[e][k] = fac;
sbr->data[1].env_facs[e][k] = av_mul_sf(fac, temp2);
}
}
for (e = 1; e <= sbr->data[0].bs_num_noise; e++) {
for (k = 0; k < sbr->n_q; k++) {
SoftFloat temp1, temp2, fac;

temp1.exp = NOISE_FLOOR_OFFSET - \
sbr->data[0].noise_facs[e][k].mant + 2;
temp1.mant = 0x20000000;
temp2.exp = 12 - sbr->data[1].noise_facs[e][k].mant + 1;
temp2.mant = 0x20000000;
fac = av_div_sf(temp1, av_add_sf(FLOAT_1, temp2));
sbr->data[0].noise_facs[e][k] = fac;
sbr->data[1].noise_facs[e][k] = av_mul_sf(fac, temp2);
}
}
} else { // SCE or one non-coupled CPE
for (ch = 0; ch < (id_aac == TYPE_CPE) + 1; ch++) {
int alpha = sbr->data[ch].bs_amp_res ? 2 : 1;
for (e = 1; e <= sbr->data[ch].bs_num_env; e++)
for (k = 0; k < sbr->n[sbr->data[ch].bs_freq_res[e]]; k++){
SoftFloat temp1;

temp1.exp = alpha * sbr->data[ch].env_facs[e][k].mant + 12;
if (temp1.exp & 1)
temp1.mant = 759250125;
else
temp1.mant = 0x20000000;
temp1.exp = (temp1.exp >> 1) + 1;

sbr->data[ch].env_facs[e][k] = temp1;
}
for (e = 1; e <= sbr->data[ch].bs_num_noise; e++)
for (k = 0; k < sbr->n_q; k++){
sbr->data[ch].noise_facs[e][k].exp = NOISE_FLOOR_OFFSET - \
sbr->data[ch].noise_facs[e][k].mant + 1;
sbr->data[ch].noise_facs[e][k].mant = 0x20000000;
}
}
}
}

/** High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering
* (14496-3 sp04 p214)
* Warning: This routine does not seem numerically stable.
*/
static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
int (*alpha0)[2], int (*alpha1)[2],
const int X_low[32][40][2], int k0)
{
int k;
int shift, round;

for (k = 0; k < k0; k++) {
SoftFloat phi[3][2][2];
SoftFloat a00, a01, a10, a11;
SoftFloat dk;

dsp->autocorrelate(X_low[k], phi);

dk = av_sub_sf(av_mul_sf(phi[2][1][0], phi[1][0][0]),
av_mul_sf(av_add_sf(av_mul_sf(phi[1][1][0], phi[1][1][0]),
av_mul_sf(phi[1][1][1], phi[1][1][1])), FLOAT_0999999));

if (!dk.mant) {
a10 = FLOAT_0;
a11 = FLOAT_0;
} else {
SoftFloat temp_real, temp_im;
temp_real = av_sub_sf(av_sub_sf(av_mul_sf(phi[0][0][0], phi[1][1][0]),
av_mul_sf(phi[0][0][1], phi[1][1][1])),
av_mul_sf(phi[0][1][0], phi[1][0][0]));
temp_im = av_sub_sf(av_add_sf(av_mul_sf(phi[0][0][0], phi[1][1][1]),
av_mul_sf(phi[0][0][1], phi[1][1][0])),
av_mul_sf(phi[0][1][1], phi[1][0][0]));

a10 = av_div_sf(temp_real, dk);
a11 = av_div_sf(temp_im, dk);
}

if (!phi[1][0][0].mant) {
a00 = FLOAT_0;
a01 = FLOAT_0;
} else {
SoftFloat temp_real, temp_im;
temp_real = av_add_sf(phi[0][0][0],
av_add_sf(av_mul_sf(a10, phi[1][1][0]),
av_mul_sf(a11, phi[1][1][1])));
temp_im = av_add_sf(phi[0][0][1],
av_sub_sf(av_mul_sf(a11, phi[1][1][0]),
av_mul_sf(a10, phi[1][1][1])));

temp_real.mant = -temp_real.mant;
temp_im.mant = -temp_im.mant;
a00 = av_div_sf(temp_real, phi[1][0][0]);
a01 = av_div_sf(temp_im, phi[1][0][0]);
}

shift = a00.exp;
if (shift >= 3)
alpha0[k][0] = 0x7fffffff;
else {
a00.mant <<= 1;
shift = 2-shift;
if (shift == 0)
alpha0[k][0] = a00.mant;
else {
round = 1 << (shift-1);
alpha0[k][0] = (a00.mant + round) >> shift;
}
}

shift = a01.exp;
if (shift >= 3)
alpha0[k][1] = 0x7fffffff;
else {
a01.mant <<= 1;
shift = 2-shift;
if (shift == 0)
alpha0[k][1] = a01.mant;
else {
round = 1 << (shift-1);
alpha0[k][1] = (a01.mant + round) >> shift;
}
}
shift = a10.exp;
if (shift >= 3)
alpha1[k][0] = 0x7fffffff;
else {
a10.mant <<= 1;
shift = 2-shift;
if (shift == 0)
alpha1[k][0] = a10.mant;
else {
round = 1 << (shift-1);
alpha1[k][0] = (a10.mant + round) >> shift;
}
}

shift = a11.exp;
if (shift >= 3)
alpha1[k][1] = 0x7fffffff;
else {
a11.mant <<= 1;
shift = 2-shift;
if (shift == 0)
alpha1[k][1] = a11.mant;
else {
round = 1 << (shift-1);
alpha1[k][1] = (a11.mant + round) >> shift;
}
}

shift = (int)(((int64_t)(alpha1[k][0]>>1) * (alpha1[k][0]>>1) + \
(int64_t)(alpha1[k][1]>>1) * (alpha1[k][1]>>1) + \
0x40000000) >> 31);
if (shift >= 0x20000000){
alpha1[k][0] = 0;
alpha1[k][1] = 0;
alpha0[k][0] = 0;
alpha0[k][1] = 0;
}

shift = (int)(((int64_t)(alpha0[k][0]>>1) * (alpha0[k][0]>>1) + \
(int64_t)(alpha0[k][1]>>1) * (alpha0[k][1]>>1) + \
0x40000000) >> 31);
if (shift >= 0x20000000){
alpha1[k][0] = 0;
alpha1[k][1] = 0;
alpha0[k][0] = 0;
alpha0[k][1] = 0;
}
}
}

/// Chirp Factors (14496-3 sp04 p214)
static void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
{
int i;
int new_bw;
static const int bw_tab[] = { 0, 1610612736, 1932735283, 2104533975 };
int64_t accu;

for (i = 0; i < sbr->n_q; i++) {
if (ch_data->bs_invf_mode[0][i] + ch_data->bs_invf_mode[1][i] == 1)
new_bw = 1288490189;
else
new_bw = bw_tab[ch_data->bs_invf_mode[0][i]];

if (new_bw < ch_data->bw_array[i]){
accu = (int64_t)new_bw * 1610612736;
accu += (int64_t)ch_data->bw_array[i] * 0x20000000;
new_bw = (int)((accu + 0x40000000) >> 31);
} else {
accu = (int64_t)new_bw * 1946157056;
accu += (int64_t)ch_data->bw_array[i] * 201326592;
new_bw = (int)((accu + 0x40000000) >> 31);
}
ch_data->bw_array[i] = new_bw < 0x2000000 ? 0 : new_bw;
}
}

/**
* Calculation of levels of additional HF signal components (14496-3 sp04 p219)
* and Calculation of gain (14496-3 sp04 p219)
*/
static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr,
SBRData *ch_data, const int e_a[2])
{
int e, k, m;
// max gain limits : -3dB, 0dB, 3dB, inf dB (limiter off)
static const SoftFloat limgain[4] = { { 760155524, 0 }, { 0x20000000, 1 },
{ 758351638, 1 }, { 625000000, 34 } };

for (e = 0; e < ch_data->bs_num_env; e++) {
int delta = !((e == e_a[1]) || (e == e_a[0]));
for (k = 0; k < sbr->n_lim; k++) {
SoftFloat gain_boost, gain_max;
SoftFloat sum[2] = { { 0, 0}, { 0, 0 } };
for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
const SoftFloat temp = av_div_sf(sbr->e_origmapped[e][m],
av_add_sf(FLOAT_1, sbr->q_mapped[e][m]));
sbr->q_m[e][m] = av_sqrt_sf(av_mul_sf(temp, sbr->q_mapped[e][m]));
sbr->s_m[e][m] = av_sqrt_sf(av_mul_sf(temp, av_int2sf(ch_data->s_indexmapped[e + 1][m], 0)));
if (!sbr->s_mapped[e][m]) {
if (delta) {
sbr->gain[e][m] = av_sqrt_sf(av_div_sf(sbr->e_origmapped[e][m],
av_mul_sf(av_add_sf(FLOAT_1, sbr->e_curr[e][m]),
av_add_sf(FLOAT_1, sbr->q_mapped[e][m]))));
} else {
sbr->gain[e][m] = av_sqrt_sf(av_div_sf(sbr->e_origmapped[e][m],
av_add_sf(FLOAT_1, sbr->e_curr[e][m])));
}
} else {
sbr->gain[e][m] = av_sqrt_sf(
av_div_sf(
av_mul_sf(sbr->e_origmapped[e][m], sbr->q_mapped[e][m]),
av_mul_sf(
av_add_sf(FLOAT_1, sbr->e_curr[e][m]),
av_add_sf(FLOAT_1, sbr->q_mapped[e][m]))));
}
}
for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
sum[0] = av_add_sf(sum[0], sbr->e_origmapped[e][m]);
sum[1] = av_add_sf(sum[1], sbr->e_curr[e][m]);
}
gain_max = av_mul_sf(limgain[sbr->bs_limiter_gains],
av_sqrt_sf(
av_div_sf(
av_add_sf(FLOAT_EPSILON, sum[0]),
av_add_sf(FLOAT_EPSILON, sum[1]))));
if (av_gt_sf(gain_max, FLOAT_100000))
gain_max = FLOAT_100000;
for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
SoftFloat q_m_max = av_div_sf(
av_mul_sf(sbr->q_m[e][m], gain_max),
sbr->gain[e][m]);
if (av_gt_sf(sbr->q_m[e][m], q_m_max))
sbr->q_m[e][m] = q_m_max;
if (av_gt_sf(sbr->gain[e][m], gain_max))
sbr->gain[e][m] = gain_max;
}
sum[0] = sum[1] = FLOAT_0;
for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
sum[0] = av_add_sf(sum[0], sbr->e_origmapped[e][m]);
sum[1] = av_add_sf(sum[1],
av_mul_sf(
av_mul_sf(sbr->e_curr[e][m],
sbr->gain[e][m]),
sbr->gain[e][m]));
sum[1] = av_add_sf(sum[1],
av_mul_sf(sbr->s_m[e][m], sbr->s_m[e][m]));
if (delta && !sbr->s_m[e][m].mant)
sum[1] = av_add_sf(sum[1],
av_mul_sf(sbr->q_m[e][m], sbr->q_m[e][m]));
}
gain_boost = av_sqrt_sf(
av_div_sf(
av_add_sf(FLOAT_EPSILON, sum[0]),
av_add_sf(FLOAT_EPSILON, sum[1])));
if (av_gt_sf(gain_boost, FLOAT_1584893192))
gain_boost = FLOAT_1584893192;

for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
sbr->gain[e][m] = av_mul_sf(sbr->gain[e][m], gain_boost);
sbr->q_m[e][m] = av_mul_sf(sbr->q_m[e][m], gain_boost);
sbr->s_m[e][m] = av_mul_sf(sbr->s_m[e][m], gain_boost);
}
}
}
}

/// Assembling HF Signals (14496-3 sp04 p220)
static void sbr_hf_assemble(int Y1[38][64][2],
const int X_high[64][40][2],
SpectralBandReplication *sbr, SBRData *ch_data,
const int e_a[2])
{
int e, i, j, m;
const int h_SL = 4 * !sbr->bs_smoothing_mode;
const int kx = sbr->kx[1];
const int m_max = sbr->m[1];
static const SoftFloat h_smooth[5] = {
{ 715827883, -1 },
{ 647472402, -1 },
{ 937030863, -2 },
{ 989249804, -3 },
{ 546843842, -4 },
};
SoftFloat (*g_temp)[48] = ch_data->g_temp, (*q_temp)[48] = ch_data->q_temp;
int indexnoise = ch_data->f_indexnoise;
int indexsine = ch_data->f_indexsine;

if (sbr->reset) {
for (i = 0; i < h_SL; i++) {
memcpy(g_temp[i + 2*ch_data->t_env[0]], sbr->gain[0], m_max * sizeof(sbr->gain[0][0]));
memcpy(q_temp[i + 2*ch_data->t_env[0]], sbr->q_m[0], m_max * sizeof(sbr->q_m[0][0]));
}
} else if (h_SL) {
for (i = 0; i < 4; i++) {
memcpy(g_temp[i + 2 * ch_data->t_env[0]],
g_temp[i + 2 * ch_data->t_env_num_env_old],
sizeof(g_temp[0]));
memcpy(q_temp[i + 2 * ch_data->t_env[0]],
q_temp[i + 2 * ch_data->t_env_num_env_old],
sizeof(q_temp[0]));
}
}

for (e = 0; e < ch_data->bs_num_env; e++) {
for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
memcpy(g_temp[h_SL + i], sbr->gain[e], m_max * sizeof(sbr->gain[0][0]));
memcpy(q_temp[h_SL + i], sbr->q_m[e], m_max * sizeof(sbr->q_m[0][0]));
}
}

for (e = 0; e < ch_data->bs_num_env; e++) {
for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
SoftFloat g_filt_tab[48];
SoftFloat q_filt_tab[48];
SoftFloat *g_filt, *q_filt;

if (h_SL && e != e_a[0] && e != e_a[1]) {
g_filt = g_filt_tab;
q_filt = q_filt_tab;
for (m = 0; m < m_max; m++) {
const int idx1 = i + h_SL;
g_filt[m].mant = g_filt[m].exp = 0;
q_filt[m].mant = q_filt[m].exp = 0;
for (j = 0; j <= h_SL; j++) {
g_filt[m] = av_add_sf(g_filt[m],
av_mul_sf(g_temp[idx1 - j][m],
h_smooth[j]));
q_filt[m] = av_add_sf(q_filt[m],
av_mul_sf(q_temp[idx1 - j][m],
h_smooth[j]));
}
}
} else {
g_filt = g_temp[i + h_SL];
q_filt = q_temp[i];
}

sbr->dsp.hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
i + ENVELOPE_ADJUSTMENT_OFFSET);

if (e != e_a[0] && e != e_a[1]) {
sbr->dsp.hf_apply_noise[indexsine](Y1[i] + kx, sbr->s_m[e],
q_filt, indexnoise,
kx, m_max);
} else {
int idx = indexsine&1;
int A = (1-((indexsine+(kx & 1))&2));
int B = (A^(-idx)) + idx;
int *out = &Y1[i][kx][idx];
int shift, round;

SoftFloat *in = sbr->s_m[e];
for (m = 0; m+1 < m_max; m+=2) {
shift = 22 - in[m ].exp;
round = 1 << (shift-1);
out[2*m ] += (in[m ].mant * A + round) >> shift;

shift = 22 - in[m+1].exp;
round = 1 << (shift-1);
out[2*m+2] += (in[m+1].mant * B + round) >> shift;
}
if(m_max&1)
{
shift = 22 - in[m ].exp;
round = 1 << (shift-1);

out[2*m ] += (in[m ].mant * A + round) >> shift;
}
}
indexnoise = (indexnoise + m_max) & 0x1ff;
indexsine = (indexsine + 1) & 3;
}
}
ch_data->f_indexnoise = indexnoise;
ch_data->f_indexsine = indexsine;
}

#include "aacsbr_template.c"

+ 168
- 43
libavcodec/aacsbr_template.c View File

@@ -3,6 +3,10 @@
* Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
* Copyright (c) 2009-2010 Alex Converse <alex.converse@gmail.com>
*
* Fixed point code
* Copyright (c) 2013
* MIPS Technologies, Inc., California.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
@@ -24,9 +28,11 @@
* @file
* AAC Spectral Band Replication decoding functions
* @author Robert Swain ( rob opendot cl )
* @author Stanislav Ocovaj ( stanislav.ocovaj@imgtec.com )
* @author Zoran Basaric ( zoran.basaric@imgtec.com )
*/

av_cold void ff_aac_sbr_init(void)
av_cold void AAC_RENAME(ff_aac_sbr_init)(void)
{
static const struct {
const void *sbr_codes, *sbr_bits;
@@ -72,7 +78,7 @@ static void sbr_turnoff(SpectralBandReplication *sbr) {
memset(&sbr->spectrum_params, -1, sizeof(SpectrumParameters));
}

av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
av_cold void AAC_RENAME(ff_aac_sbr_ctx_init)(AACContext *ac, SpectralBandReplication *sbr)
{
if(sbr->mdct.mdct_bits)
return;
@@ -83,17 +89,17 @@ av_cold void ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
/* SBR requires samples to be scaled to +/-32768.0 to work correctly.
* mdct scale factors are adjusted to scale up from +/-1.0 at analysis
* and scale back down at synthesis. */
ff_mdct_init(&sbr->mdct, 7, 1, 1.0 / (64 * 32768.0));
ff_mdct_init(&sbr->mdct_ana, 7, 1, -2.0 * 32768.0);
AAC_RENAME_32(ff_mdct_init)(&sbr->mdct, 7, 1, 1.0 / (64 * 32768.0));
AAC_RENAME_32(ff_mdct_init)(&sbr->mdct_ana, 7, 1, -2.0 * 32768.0);
ff_ps_ctx_init(&sbr->ps);
ff_sbrdsp_init(&sbr->dsp);
AAC_RENAME(ff_sbrdsp_init)(&sbr->dsp);
aacsbr_func_ptr_init(&sbr->c);
}

av_cold void ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
av_cold void AAC_RENAME(ff_aac_sbr_ctx_close)(SpectralBandReplication *sbr)
{
ff_mdct_end(&sbr->mdct);
ff_mdct_end(&sbr->mdct_ana);
AAC_RENAME_32(ff_mdct_end)(&sbr->mdct);
AAC_RENAME_32(ff_mdct_end)(&sbr->mdct_ana);
}

static int qsort_comparison_function_int16(const void *a, const void *b)
@@ -115,10 +121,10 @@ static void sbr_make_f_tablelim(SpectralBandReplication *sbr)
{
int k;
if (sbr->bs_limiter_bands > 0) {
static const float bands_warped[3] = { 1.32715174233856803909f, //2^(0.49/1.2)
1.18509277094158210129f, //2^(0.49/2)
1.11987160404675912501f }; //2^(0.49/3)
const float lim_bands_per_octave_warped = bands_warped[sbr->bs_limiter_bands - 1];
static const INTFLOAT bands_warped[3] = { Q23(1.32715174233856803909f), //2^(0.49/1.2)
Q23(1.18509277094158210129f), //2^(0.49/2)
Q23(1.11987160404675912501f) }; //2^(0.49/3)
const INTFLOAT lim_bands_per_octave_warped = bands_warped[sbr->bs_limiter_bands - 1];
int16_t patch_borders[7];
uint16_t *in = sbr->f_tablelim + 1, *out = sbr->f_tablelim;

@@ -138,7 +144,11 @@ static void sbr_make_f_tablelim(SpectralBandReplication *sbr)

sbr->n_lim = sbr->n[0] + sbr->num_patches - 1;
while (out < sbr->f_tablelim + sbr->n_lim) {
#if USE_FIXED
if ((*in << 23) >= *out * lim_bands_per_octave_warped) {
#else
if (*in >= *out * lim_bands_per_octave_warped) {
#endif /* USE_FIXED */
*++out = *in++;
} else if (*in == *out ||
!in_table_int16(patch_borders, sbr->num_patches, *in)) {
@@ -344,6 +354,9 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
int two_regions, num_bands_0;
int vdk0_max, vdk1_min;
int16_t vk0[49];
#if USE_FIXED
int tmp, nz = 0;
#endif /* USE_FIXED */

if (49 * sbr->k[2] > 110 * sbr->k[0]) {
two_regions = 1;
@@ -353,7 +366,19 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,
sbr->k[1] = sbr->k[2];
}

#if USE_FIXED
tmp = (sbr->k[1] << 23) / sbr->k[0];
while (tmp < 0x40000000) {
tmp <<= 1;
nz++;
}
tmp = fixed_log(tmp - 0x80000000);
tmp = (int)(((int64_t)tmp * CONST_RECIP_LN2 + 0x20000000) >> 30);
tmp = (((tmp + 0x80) >> 8) + ((8 - nz) << 23)) * half_bands;
num_bands_0 = ((tmp + 0x400000) >> 23) * 2;
#else
num_bands_0 = lrintf(half_bands * log2f(sbr->k[1] / (float)sbr->k[0])) * 2;
#endif /* USE_FIXED */

if (num_bands_0 <= 0) { // Requirements (14496-3 sp04 p205)
av_log(ac->avctx, AV_LOG_ERROR, "Invalid num_bands_0: %d\n", num_bands_0);
@@ -378,11 +403,27 @@ static int sbr_make_f_master(AACContext *ac, SpectralBandReplication *sbr,

if (two_regions) {
int16_t vk1[49];
#if USE_FIXED
int num_bands_1;

tmp = (sbr->k[2] << 23) / sbr->k[1];
nz = 0;
while (tmp < 0x40000000) {
tmp <<= 1;
nz++;
}
tmp = fixed_log(tmp - 0x80000000);
tmp = (int)(((int64_t)tmp * CONST_RECIP_LN2 + 0x20000000) >> 30);
tmp = (((tmp + 0x80) >> 8) + ((8 - nz) << 23)) * half_bands;
if (spectrum->bs_alter_scale)
tmp = (int)(((int64_t)tmp * CONST_076923 + 0x40000000) >> 31);
num_bands_1 = ((tmp + 0x400000) >> 23) * 2;
#else
float invwarp = spectrum->bs_alter_scale ? 0.76923076923076923077f
: 1.0f; // bs_alter_scale = {0,1}
int num_bands_1 = lrintf(half_bands * invwarp *
log2f(sbr->k[2] / (float)sbr->k[1])) * 2;

#endif /* USE_FIXED */
make_bands(vk1+1, sbr->k[1], sbr->k[2], num_bands_1);

vdk1_min = array_min_int16(vk1 + 1, num_bands_1);
@@ -487,6 +528,9 @@ static int sbr_hf_calc_npatches(AACContext *ac, SpectralBandReplication *sbr)
static int sbr_make_f_derived(AACContext *ac, SpectralBandReplication *sbr)
{
int k, temp;
#if USE_FIXED
int nz = 0;
#endif /* USE_FIXED */

sbr->n[1] = sbr->n_master - sbr->spectrum_params.bs_xover_band;
sbr->n[0] = (sbr->n[1] + 1) >> 1;
@@ -511,9 +555,24 @@ static int sbr_make_f_derived(AACContext *ac, SpectralBandReplication *sbr)
temp = sbr->n[1] & 1;
for (k = 1; k <= sbr->n[0]; k++)
sbr->f_tablelow[k] = sbr->f_tablehigh[2 * k - temp];
#if USE_FIXED
temp = (sbr->k[2] << 23) / sbr->kx[1];
while (temp < 0x40000000) {
temp <<= 1;
nz++;
}
temp = fixed_log(temp - 0x80000000);
temp = (int)(((int64_t)temp * CONST_RECIP_LN2 + 0x20000000) >> 30);
temp = (((temp + 0x80) >> 8) + ((8 - nz) << 23)) * sbr->spectrum_params.bs_noise_bands;

sbr->n_q = (temp + 0x400000) >> 23;
if (sbr->n_q < 1)
sbr->n_q = 1;
#else
sbr->n_q = FFMAX(1, lrintf(sbr->spectrum_params.bs_noise_bands *
log2f(sbr->k[2] / (float)sbr->kx[1]))); // 0 <= bs_noise_bands <= 3
#endif /* USE_FIXED */

if (sbr->n_q > 5) {
av_log(ac->avctx, AV_LOG_ERROR, "Too many noise floor scale factors: %d\n", sbr->n_q);
return -1;
@@ -770,6 +829,31 @@ static void read_sbr_envelope(SpectralBandReplication *sbr, GetBitContext *gb,
}
}

#if USE_FIXED
for (i = 0; i < ch_data->bs_num_env; i++) {
if (ch_data->bs_df_env[i]) {
// bs_freq_res[0] == bs_freq_res[bs_num_env] from prev frame
if (ch_data->bs_freq_res[i + 1] == ch_data->bs_freq_res[i]) {
for (j = 0; j < sbr->n[ch_data->bs_freq_res[i + 1]]; j++)
ch_data->env_facs[i + 1][j].mant = ch_data->env_facs[i][j].mant + delta * (get_vlc2(gb, t_huff, 9, 3) - t_lav);
} else if (ch_data->bs_freq_res[i + 1]) {
for (j = 0; j < sbr->n[ch_data->bs_freq_res[i + 1]]; j++) {
k = (j + odd) >> 1; // find k such that f_tablelow[k] <= f_tablehigh[j] < f_tablelow[k + 1]
ch_data->env_facs[i + 1][j].mant = ch_data->env_facs[i][k].mant + delta * (get_vlc2(gb, t_huff, 9, 3) - t_lav);
}
} else {
for (j = 0; j < sbr->n[ch_data->bs_freq_res[i + 1]]; j++) {
k = j ? 2*j - odd : 0; // find k such that f_tablehigh[k] == f_tablelow[j]
ch_data->env_facs[i + 1][j].mant = ch_data->env_facs[i][k].mant + delta * (get_vlc2(gb, t_huff, 9, 3) - t_lav);
}
}
} else {
ch_data->env_facs[i + 1][0].mant = delta * get_bits(gb, bits); // bs_env_start_value_balance
for (j = 1; j < sbr->n[ch_data->bs_freq_res[i + 1]]; j++)
ch_data->env_facs[i + 1][j].mant = ch_data->env_facs[i + 1][j - 1].mant + delta * (get_vlc2(gb, f_huff, 9, 3) - f_lav);
}
}
#else
for (i = 0; i < ch_data->bs_num_env; i++) {
if (ch_data->bs_df_env[i]) {
// bs_freq_res[0] == bs_freq_res[bs_num_env] from prev frame
@@ -793,6 +877,7 @@ static void read_sbr_envelope(SpectralBandReplication *sbr, GetBitContext *gb,
ch_data->env_facs[i + 1][j] = ch_data->env_facs[i + 1][j - 1] + delta * (get_vlc2(gb, f_huff, 9, 3) - f_lav);
}
}
#endif /* USE_FIXED */

//assign 0th elements of env_facs from last elements
memcpy(ch_data->env_facs[0], ch_data->env_facs[ch_data->bs_num_env],
@@ -819,6 +904,18 @@ static void read_sbr_noise(SpectralBandReplication *sbr, GetBitContext *gb,
f_lav = vlc_sbr_lav[F_HUFFMAN_ENV_3_0DB];
}

#if USE_FIXED
for (i = 0; i < ch_data->bs_num_noise; i++) {
if (ch_data->bs_df_noise[i]) {
for (j = 0; j < sbr->n_q; j++)
ch_data->noise_facs[i + 1][j].mant = ch_data->noise_facs[i][j].mant + delta * (get_vlc2(gb, t_huff, 9, 2) - t_lav);
} else {
ch_data->noise_facs[i + 1][0].mant = delta * get_bits(gb, 5); // bs_noise_start_value_balance or bs_noise_start_value_level
for (j = 1; j < sbr->n_q; j++)
ch_data->noise_facs[i + 1][j].mant = ch_data->noise_facs[i + 1][j - 1].mant + delta * (get_vlc2(gb, f_huff, 9, 3) - f_lav);
}
}
#else
for (i = 0; i < ch_data->bs_num_noise; i++) {
if (ch_data->bs_df_noise[i]) {
for (j = 0; j < sbr->n_q; j++)
@@ -829,6 +926,7 @@ static void read_sbr_noise(SpectralBandReplication *sbr, GetBitContext *gb,
ch_data->noise_facs[i + 1][j] = ch_data->noise_facs[i + 1][j - 1] + delta * (get_vlc2(gb, f_huff, 9, 3) - f_lav);
}
}
#endif /* USE_FIXED */

//assign 0th elements of noise_facs from last elements
memcpy(ch_data->noise_facs[0], ch_data->noise_facs[ch_data->bs_num_noise],
@@ -992,7 +1090,7 @@ static void sbr_reset(AACContext *ac, SpectralBandReplication *sbr)
*
* @return Returns number of bytes consumed from the TYPE_FIL element.
*/
int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
int AAC_RENAME(ff_decode_sbr_extension)(AACContext *ac, SpectralBandReplication *sbr,
GetBitContext *gb_host, int crc, int cnt, int id_aac)
{
unsigned int num_sbr_bits = 0, num_align_bits;
@@ -1044,9 +1142,13 @@ int ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr,
* @param W array of complex-valued samples split into subbands
*/
#ifndef sbr_qmf_analysis
#if USE_FIXED
static void sbr_qmf_analysis(AVFixedDSPContext *dsp, FFTContext *mdct,
#else
static void sbr_qmf_analysis(AVFloatDSPContext *dsp, FFTContext *mdct,
SBRDSPContext *sbrdsp, const float *in, float *x,
float z[320], float W[2][32][32][2], int buf_idx)
#endif /* USE_FIXED */
SBRDSPContext *sbrdsp, const INTFLOAT *in, INTFLOAT *x,
INTFLOAT z[320], INTFLOAT W[2][32][32][2], int buf_idx)
{
int i;
memcpy(x , x+1024, (320-32)*sizeof(x[0]));
@@ -1069,19 +1171,23 @@ static void sbr_qmf_analysis(AVFloatDSPContext *dsp, FFTContext *mdct,
*/
#ifndef sbr_qmf_synthesis
static void sbr_qmf_synthesis(FFTContext *mdct,
#if USE_FIXED
SBRDSPContext *sbrdsp, AVFixedDSPContext *dsp,
#else
SBRDSPContext *sbrdsp, AVFloatDSPContext *dsp,
float *out, float X[2][38][64],
float mdct_buf[2][64],
float *v0, int *v_off, const unsigned int div)
#endif /* USE_FIXED */
INTFLOAT *out, INTFLOAT X[2][38][64],
INTFLOAT mdct_buf[2][64],
INTFLOAT *v0, int *v_off, const unsigned int div)
{
int i, n;
const float *sbr_qmf_window = div ? sbr_qmf_window_ds : sbr_qmf_window_us;
const INTFLOAT *sbr_qmf_window = div ? sbr_qmf_window_ds : sbr_qmf_window_us;
const int step = 128 >> div;
float *v;
INTFLOAT *v;
for (i = 0; i < 32; i++) {
if (*v_off < step) {
int saved_samples = (1280 - 128) >> div;
memcpy(&v0[SBR_SYNTHESIS_BUF_SIZE - saved_samples], v0, saved_samples * sizeof(float));
memcpy(&v0[SBR_SYNTHESIS_BUF_SIZE - saved_samples], v0, saved_samples * sizeof(INTFLOAT));
*v_off = SBR_SYNTHESIS_BUF_SIZE - saved_samples - step;
} else {
*v_off -= step;
@@ -1117,7 +1223,7 @@ static void sbr_qmf_synthesis(FFTContext *mdct,

/// Generate the subband filtered lowband
static int sbr_lf_gen(AACContext *ac, SpectralBandReplication *sbr,
float X_low[32][40][2], const float W[2][32][32][2],
INTFLOAT X_low[32][40][2], const INTFLOAT W[2][32][32][2],
int buf_idx)
{
int i, k;
@@ -1142,9 +1248,9 @@ static int sbr_lf_gen(AACContext *ac, SpectralBandReplication *sbr,

/// High Frequency Generator (14496-3 sp04 p215)
static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
float X_high[64][40][2], const float X_low[32][40][2],
const float (*alpha0)[2], const float (*alpha1)[2],
const float bw_array[5], const uint8_t *t_env,
INTFLOAT X_high[64][40][2], const INTFLOAT X_low[32][40][2],
const INTFLOAT (*alpha0)[2], const INTFLOAT (*alpha1)[2],
const INTFLOAT bw_array[5], const uint8_t *t_env,
int bs_num_env)
{
int j, x;
@@ -1176,9 +1282,9 @@ static int sbr_hf_gen(AACContext *ac, SpectralBandReplication *sbr,
}

/// Generate the subband filtered lowband
static int sbr_x_gen(SpectralBandReplication *sbr, float X[2][38][64],
const float Y0[38][64][2], const float Y1[38][64][2],
const float X_low[32][40][2], int ch)
static int sbr_x_gen(SpectralBandReplication *sbr, INTFLOAT X[2][38][64],
const INTFLOAT Y0[38][64][2], const INTFLOAT Y1[38][64][2],
const INTFLOAT X_low[32][40][2], int ch)
{
int k, i;
const int i_f = 32;
@@ -1270,7 +1376,7 @@ static int sbr_mapping(AACContext *ac, SpectralBandReplication *sbr,
}

/// Estimation of current envelope (14496-3 sp04 p218)
static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
static void sbr_env_estimate(AAC_FLOAT (*e_curr)[48], INTFLOAT X_high[64][40][2],
SpectralBandReplication *sbr, SBRData *ch_data)
{
int e, m;
@@ -1278,13 +1384,21 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],

if (sbr->bs_interpol_freq) {
for (e = 0; e < ch_data->bs_num_env; e++) {
#if USE_FIXED
const SoftFloat recip_env_size = av_int2sf(0x20000000 / (ch_data->t_env[e + 1] - ch_data->t_env[e]), 30);
#else
const float recip_env_size = 0.5f / (ch_data->t_env[e + 1] - ch_data->t_env[e]);
#endif /* USE_FIXED */
int ilb = ch_data->t_env[e] * 2 + ENVELOPE_ADJUSTMENT_OFFSET;
int iub = ch_data->t_env[e + 1] * 2 + ENVELOPE_ADJUSTMENT_OFFSET;

for (m = 0; m < sbr->m[1]; m++) {
float sum = sbr->dsp.sum_square(X_high[m+kx1] + ilb, iub - ilb);
AAC_FLOAT sum = sbr->dsp.sum_square(X_high[m+kx1] + ilb, iub - ilb);
#if USE_FIXED
e_curr[e][m] = av_mul_sf(sum, recip_env_size);
#else
e_curr[e][m] = sum * recip_env_size;
#endif /* USE_FIXED */
}
}
} else {
@@ -1297,6 +1411,14 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
const uint16_t *table = ch_data->bs_freq_res[e + 1] ? sbr->f_tablehigh : sbr->f_tablelow;

for (p = 0; p < sbr->n[ch_data->bs_freq_res[e + 1]]; p++) {
#if USE_FIXED
SoftFloat sum = { 0, 0 };
const SoftFloat den = av_int2sf(0x20000000 / (env_size * (table[p + 1] - table[p])), 29);
for (k = table[p]; k < table[p + 1]; k++) {
sum = av_add_sf(sum, sbr->dsp.sum_square(X_high[k] + ilb, iub - ilb));
}
sum = av_mul_sf(sum, den);
#else
float sum = 0.0f;
const int den = env_size * (table[p + 1] - table[p]);

@@ -1304,6 +1426,7 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
sum += sbr->dsp.sum_square(X_high[k] + ilb, iub - ilb);
}
sum /= den;
#endif /* USE_FIXED */
for (k = table[p]; k < table[p + 1]; k++) {
e_curr[e][k - kx1] = sum;
}
@@ -1312,8 +1435,8 @@ static void sbr_env_estimate(float (*e_curr)[48], float X_high[64][40][2],
}
}

void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
float* L, float* R)
void AAC_RENAME(ff_sbr_apply)(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
INTFLOAT* L, INTFLOAT* R)
{
int downsampled = ac->oc[1].m4ac.ext_sample_rate < sbr->sample_rate;
int ch;
@@ -1339,21 +1462,21 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
for (ch = 0; ch < nch; ch++) {
/* decode channel */
sbr_qmf_analysis(ac->fdsp, &sbr->mdct_ana, &sbr->dsp, ch ? R : L, sbr->data[ch].analysis_filterbank_samples,
(float*)sbr->qmf_filter_scratch,
(INTFLOAT*)sbr->qmf_filter_scratch,
sbr->data[ch].W, sbr->data[ch].Ypos);
sbr->c.sbr_lf_gen(ac, sbr, sbr->X_low,
(const float (*)[32][32][2]) sbr->data[ch].W,
(const INTFLOAT (*)[32][32][2]) sbr->data[ch].W,
sbr->data[ch].Ypos);
sbr->data[ch].Ypos ^= 1;
if (sbr->start) {
sbr->c.sbr_hf_inverse_filter(&sbr->dsp, sbr->alpha0, sbr->alpha1,
(const float (*)[40][2]) sbr->X_low, sbr->k[0]);
(const INTFLOAT (*)[40][2]) sbr->X_low, sbr->k[0]);
sbr_chirp(sbr, &sbr->data[ch]);
av_assert0(sbr->data[ch].bs_num_env > 0);
sbr_hf_gen(ac, sbr, sbr->X_high,
(const float (*)[40][2]) sbr->X_low,
(const float (*)[2]) sbr->alpha0,
(const float (*)[2]) sbr->alpha1,
(const INTFLOAT (*)[40][2]) sbr->X_low,
(const INTFLOAT (*)[2]) sbr->alpha0,
(const INTFLOAT (*)[2]) sbr->alpha1,
sbr->data[ch].bw_array, sbr->data[ch].t_env,
sbr->data[ch].bs_num_env);

@@ -1363,7 +1486,7 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,
sbr_env_estimate(sbr->e_curr, sbr->X_high, sbr, &sbr->data[ch]);
sbr_gain_calc(ac, sbr, &sbr->data[ch], sbr->data[ch].e_a);
sbr->c.sbr_hf_assemble(sbr->data[ch].Y[sbr->data[ch].Ypos],
(const float (*)[40][2]) sbr->X_high,
(const INTFLOAT (*)[40][2]) sbr->X_high,
sbr, &sbr->data[ch],
sbr->data[ch].e_a);
}
@@ -1371,9 +1494,9 @@ void ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac,

/* synthesis */
sbr->c.sbr_x_gen(sbr, sbr->X[ch],
(const float (*)[64][2]) sbr->data[ch].Y[1-sbr->data[ch].Ypos],
(const float (*)[64][2]) sbr->data[ch].Y[ sbr->data[ch].Ypos],
(const float (*)[40][2]) sbr->X_low, ch);
(const INTFLOAT (*)[64][2]) sbr->data[ch].Y[1-sbr->data[ch].Ypos],
(const INTFLOAT (*)[64][2]) sbr->data[ch].Y[ sbr->data[ch].Ypos],
(const INTFLOAT (*)[40][2]) sbr->X_low, ch);
}

if (ac->oc[1].m4ac.ps == 1) {
@@ -1405,6 +1528,8 @@ static void aacsbr_func_ptr_init(AACSBRContext *c)
c->sbr_x_gen = sbr_x_gen;
c->sbr_hf_inverse_filter = sbr_hf_inverse_filter;

#if !USE_FIXED
if(ARCH_MIPS)
ff_aacsbr_func_ptr_init_mips(c);
#endif
}

+ 10
- 5
libavcodec/lpc.h View File

@@ -25,6 +25,7 @@
#include <stdint.h>
#include "libavutil/avassert.h"
#include "libavutil/lls.h"
#include "aac_defines.h"

#define ORDER_METHOD_EST 0
#define ORDER_METHOD_2LEVEL 1
@@ -111,11 +112,15 @@ void ff_lpc_init_x86(LPCContext *s);
*/
void ff_lpc_end(LPCContext *s);

#if USE_FIXED
#define LPC_TYPE int
#else
#ifdef LPC_USE_DOUBLE
#define LPC_TYPE double
#else
#define LPC_TYPE float
#endif
#endif // USE_FIXED

/**
* Schur recursion.
@@ -152,7 +157,7 @@ static inline void compute_ref_coefs(const LPC_TYPE *autoc, int max_order,
* Levinson-Durbin recursion.
* Produce LPC coefficients from autocorrelation data.
*/
static inline int compute_lpc_coefs(const LPC_TYPE *autoc, int max_order,
static inline int AAC_RENAME(compute_lpc_coefs)(const LPC_TYPE *autoc, int max_order,
LPC_TYPE *lpc, int lpc_stride, int fail,
int normalize)
{
@@ -169,14 +174,14 @@ static inline int compute_lpc_coefs(const LPC_TYPE *autoc, int max_order,
return -1;

for(i=0; i<max_order; i++) {
LPC_TYPE r = -autoc[i];
LPC_TYPE r = AAC_SRA_R(-autoc[i], 5);

if (normalize) {
for(j=0; j<i; j++)
r -= lpc_last[j] * autoc[i-j-1];

r /= err;
err *= 1.0 - (r * r);
err *= FIXR(1.0) - (r * r);
}

lpc[i] = r;
@@ -184,8 +189,8 @@ static inline int compute_lpc_coefs(const LPC_TYPE *autoc, int max_order,
for(j=0; j < (i+1)>>1; j++) {
LPC_TYPE f = lpc_last[ j];
LPC_TYPE b = lpc_last[i-1-j];
lpc[ j] = f + r * b;
lpc[i-1-j] = b + r * f;
lpc[ j] = f + AAC_MUL26(r, b);
lpc[i-1-j] = b + AAC_MUL26(r, f);
}

if (fail && err < 0)


+ 39
- 39
libavcodec/sbr.h View File

@@ -66,9 +66,9 @@ typedef struct SBRData {
*/
unsigned bs_frame_class;
unsigned bs_add_harmonic_flag;
unsigned bs_num_env;
AAC_SIGNE bs_num_env;
uint8_t bs_freq_res[7];
unsigned bs_num_noise;
AAC_SIGNE bs_num_noise;
uint8_t bs_df_env[5];
uint8_t bs_df_noise[2];
uint8_t bs_invf_mode[2][5];
@@ -80,25 +80,25 @@ typedef struct SBRData {
* @name State variables
* @{
*/
DECLARE_ALIGNED(32, float, synthesis_filterbank_samples)[SBR_SYNTHESIS_BUF_SIZE];
DECLARE_ALIGNED(32, float, analysis_filterbank_samples) [1312];
DECLARE_ALIGNED(32, INTFLOAT, synthesis_filterbank_samples)[SBR_SYNTHESIS_BUF_SIZE];
DECLARE_ALIGNED(32, INTFLOAT, analysis_filterbank_samples) [1312];
int synthesis_filterbank_samples_offset;
///l_APrev and l_A
int e_a[2];
///Chirp factors
float bw_array[5];
INTFLOAT bw_array[5];
///QMF values of the original signal
float W[2][32][32][2];
INTFLOAT W[2][32][32][2];
///QMF output of the HF adjustor
int Ypos;
DECLARE_ALIGNED(16, float, Y)[2][38][64][2];
DECLARE_ALIGNED(16, float, g_temp)[42][48];
float q_temp[42][48];
DECLARE_ALIGNED(16, INTFLOAT, Y)[2][38][64][2];
DECLARE_ALIGNED(16, AAC_FLOAT, g_temp)[42][48];
AAC_FLOAT q_temp[42][48];
uint8_t s_indexmapped[8][48];
///Envelope scalefactors
float env_facs[6][48];
AAC_FLOAT env_facs[6][48];
///Noise scalefactors
float noise_facs[3][5];
AAC_FLOAT noise_facs[3][5];
///Envelope time borders
uint8_t t_env[8];
///Envelope time border of the last envelope of the previous frame
@@ -117,18 +117,18 @@ typedef struct SpectralBandReplication SpectralBandReplication;
*/
typedef struct AACSBRContext {
int (*sbr_lf_gen)(AACContext *ac, SpectralBandReplication *sbr,
float X_low[32][40][2], const float W[2][32][32][2],
INTFLOAT X_low[32][40][2], const INTFLOAT W[2][32][32][2],
int buf_idx);
void (*sbr_hf_assemble)(float Y1[38][64][2],
const float X_high[64][40][2],
void (*sbr_hf_assemble)(INTFLOAT Y1[38][64][2],
const INTFLOAT X_high[64][40][2],
SpectralBandReplication *sbr, SBRData *ch_data,
const int e_a[2]);
int (*sbr_x_gen)(SpectralBandReplication *sbr, float X[2][38][64],
const float Y0[38][64][2], const float Y1[38][64][2],
const float X_low[32][40][2], int ch);
int (*sbr_x_gen)(SpectralBandReplication *sbr, INTFLOAT X[2][38][64],
const INTFLOAT Y0[38][64][2], const INTFLOAT Y1[38][64][2],
const INTFLOAT X_low[32][40][2], int ch);
void (*sbr_hf_inverse_filter)(SBRDSPContext *dsp,
float (*alpha0)[2], float (*alpha1)[2],
const float X_low[32][40][2], int k0);
INTFLOAT (*alpha0)[2], INTFLOAT (*alpha1)[2],
const INTFLOAT X_low[32][40][2], int k0);
} AACSBRContext;

/**
@@ -151,23 +151,23 @@ struct SpectralBandReplication {
unsigned bs_smoothing_mode;
/** @} */
unsigned bs_coupling;
unsigned k[5]; ///< k0, k1, k2
AAC_SIGNE k[5]; ///< k0, k1, k2
///kx', and kx respectively, kx is the first QMF subband where SBR is used.
///kx' is its value from the previous frame
unsigned kx[2];
AAC_SIGNE kx[2];
///M' and M respectively, M is the number of QMF subbands that use SBR.
unsigned m[2];
AAC_SIGNE m[2];
unsigned kx_and_m_pushed;
///The number of frequency bands in f_master
unsigned n_master;
AAC_SIGNE n_master;
SBRData data[2];
PSContext ps;
///N_Low and N_High respectively, the number of frequency bands for low and high resolution
unsigned n[2];
AAC_SIGNE n[2];
///Number of noise floor bands
unsigned n_q;
AAC_SIGNE n_q;
///Number of limiter bands
unsigned n_lim;
AAC_SIGNE n_lim;
///The master QMF frequency grouping
uint16_t f_master[49];
///Frequency borders for low resolution SBR
@@ -178,33 +178,33 @@ struct SpectralBandReplication {
uint16_t f_tablenoise[6];
///Frequency borders for the limiter
uint16_t f_tablelim[30];
unsigned num_patches;
AAC_SIGNE num_patches;
uint8_t patch_num_subbands[6];
uint8_t patch_start_subband[6];
///QMF low frequency input to the HF generator
DECLARE_ALIGNED(16, float, X_low)[32][40][2];
DECLARE_ALIGNED(16, INTFLOAT, X_low)[32][40][2];
///QMF output of the HF generator
DECLARE_ALIGNED(16, float, X_high)[64][40][2];
DECLARE_ALIGNED(16, INTFLOAT, X_high)[64][40][2];
///QMF values of the reconstructed signal
DECLARE_ALIGNED(16, float, X)[2][2][38][64];
DECLARE_ALIGNED(16, INTFLOAT, X)[2][2][38][64];
///Zeroth coefficient used to filter the subband signals
DECLARE_ALIGNED(16, float, alpha0)[64][2];
DECLARE_ALIGNED(16, INTFLOAT, alpha0)[64][2];
///First coefficient used to filter the subband signals
DECLARE_ALIGNED(16, float, alpha1)[64][2];
DECLARE_ALIGNED(16, INTFLOAT, alpha1)[64][2];
///Dequantized envelope scalefactors, remapped
float e_origmapped[7][48];
AAC_FLOAT e_origmapped[7][48];
///Dequantized noise scalefactors, remapped
float q_mapped[7][48];
AAC_FLOAT q_mapped[7][48];
///Sinusoidal presence, remapped
uint8_t s_mapped[7][48];
///Estimated envelope
float e_curr[7][48];
AAC_FLOAT e_curr[7][48];
///Amplitude adjusted noise scalefactors
float q_m[7][48];
AAC_FLOAT q_m[7][48];
///Sinusoidal levels
float s_m[7][48];
float gain[7][48];
DECLARE_ALIGNED(32, float, qmf_filter_scratch)[5][64];
AAC_FLOAT s_m[7][48];
AAC_FLOAT gain[7][48];
DECLARE_ALIGNED(32, INTFLOAT, qmf_filter_scratch)[5][64];
FFTContext mdct_ana;
FFTContext mdct;
SBRDSPContext dsp;


+ 3
- 0
libavcodec/sbrdsp.c View File

@@ -20,6 +20,9 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

#define USE_FIXED 0

#include "aac.h"
#include "config.h"
#include "libavutil/attributes.h"
#include "libavutil/intfloat.h"


+ 19
- 17
libavcodec/sbrdsp.h View File

@@ -22,29 +22,31 @@
#define AVCODEC_SBRDSP_H

#include <stdint.h>
#include "aac_defines.h"
#include "libavutil/softfloat.h"

typedef struct SBRDSPContext {
void (*sum64x5)(float *z);
float (*sum_square)(float (*x)[2], int n);
void (*neg_odd_64)(float *x);
void (*qmf_pre_shuffle)(float *z);
void (*qmf_post_shuffle)(float W[32][2], const float *z);
void (*qmf_deint_neg)(float *v, const float *src);
void (*qmf_deint_bfly)(float *v, const float *src0, const float *src1);
void (*autocorrelate)(const float x[40][2], float phi[3][2][2]);
void (*hf_gen)(float (*X_high)[2], const float (*X_low)[2],
const float alpha0[2], const float alpha1[2],
float bw, int start, int end);
void (*hf_g_filt)(float (*Y)[2], const float (*X_high)[40][2],
const float *g_filt, int m_max, intptr_t ixh);
void (*hf_apply_noise[4])(float (*Y)[2], const float *s_m,
const float *q_filt, int noise,
void (*sum64x5)(INTFLOAT *z);
AAC_FLOAT (*sum_square)(INTFLOAT (*x)[2], int n);
void (*neg_odd_64)(INTFLOAT *x);
void (*qmf_pre_shuffle)(INTFLOAT *z);
void (*qmf_post_shuffle)(INTFLOAT W[32][2], const INTFLOAT *z);
void (*qmf_deint_neg)(INTFLOAT *v, const INTFLOAT *src);
void (*qmf_deint_bfly)(INTFLOAT *v, const INTFLOAT *src0, const INTFLOAT *src1);
void (*autocorrelate)(const INTFLOAT x[40][2], AAC_FLOAT phi[3][2][2]);
void (*hf_gen)(INTFLOAT (*X_high)[2], const INTFLOAT (*X_low)[2],
const INTFLOAT alpha0[2], const INTFLOAT alpha1[2],
INTFLOAT bw, int start, int end);
void (*hf_g_filt)(INTFLOAT (*Y)[2], const INTFLOAT (*X_high)[40][2],
const AAC_FLOAT *g_filt, int m_max, intptr_t ixh);
void (*hf_apply_noise[4])(INTFLOAT (*Y)[2], const AAC_FLOAT *s_m,
const AAC_FLOAT *q_filt, int noise,
int kx, int m_max);
} SBRDSPContext;

extern const float ff_sbr_noise_table[][2];
extern const INTFLOAT AAC_RENAME(ff_sbr_noise_table)[][2];

void ff_sbrdsp_init(SBRDSPContext *s);
void AAC_RENAME(ff_sbrdsp_init)(SBRDSPContext *s);
void ff_sbrdsp_init_arm(SBRDSPContext *s);
void ff_sbrdsp_init_x86(SBRDSPContext *s);
void ff_sbrdsp_init_mips(SBRDSPContext *s);


+ 286
- 0
libavcodec/sbrdsp_fixed.c View File

@@ -0,0 +1,286 @@
/*
* AAC Spectral Band Replication decoding functions
* Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
* Copyright (c) 2009-2010 Alex Converse <alex.converse@gmail.com>
*
* 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
*
* Note: Rounding-to-nearest used unless otherwise stated
*
*/

#define USE_FIXED 1

#include "aac.h"
#include "config.h"
#include "libavutil/attributes.h"
#include "libavutil/intfloat.h"
#include "sbrdsp.h"

static SoftFloat sbr_sum_square_c(int (*x)[2], int n)
{
SoftFloat ret;
int64_t accu = 0;
int i, nz, round;

for (i = 0; i < n; i += 2) {
accu += (int64_t)x[i + 0][0] * x[i + 0][0];
accu += (int64_t)x[i + 0][1] * x[i + 0][1];
accu += (int64_t)x[i + 1][0] * x[i + 1][0];
accu += (int64_t)x[i + 1][1] * x[i + 1][1];
}

i = (int)(accu >> 32);
if (i == 0) {
nz = 1;
} else {
nz = 0;
while (FFABS(i) < 0x40000000) {
i <<= 1;
nz++;
}
nz = 32 - nz;
}

round = 1 << (nz-1);
i = (int)((accu + round) >> nz);
i >>= 1;
ret = av_int2sf(i, 15 - nz);

return ret;
}

static void sbr_neg_odd_64_c(int *x)
{
int i;
for (i = 1; i < 64; i += 2)
x[i] = -x[i];
}

static void sbr_qmf_pre_shuffle_c(int *z)
{
int k;
z[64] = z[0];
z[65] = z[1];
for (k = 1; k < 32; k++) {
z[64+2*k ] = -z[64 - k];
z[64+2*k+1] = z[ k + 1];
}
}

static void sbr_qmf_post_shuffle_c(int W[32][2], const int *z)
{
int k;
for (k = 0; k < 32; k++) {
W[k][0] = -z[63-k];
W[k][1] = z[k];
}
}

static void sbr_qmf_deint_neg_c(int *v, const int *src)
{
int i;
for (i = 0; i < 32; i++) {
v[ i] = ( src[63 - 2*i ] + 0x10) >> 5;
v[63 - i] = (-src[63 - 2*i - 1] + 0x10) >> 5;
}
}

static av_always_inline SoftFloat autocorr_calc(int64_t accu)
{
int nz, mant, expo, round;
int i = (int)(accu >> 32);
if (i == 0) {
nz = 1;
} else {
nz = 0;
while (FFABS(i) < 0x40000000) {
i <<= 1;
nz++;
}
nz = 32-nz;
}

round = 1 << (nz-1);
mant = (int)((accu + round) >> nz);
mant = (mant + 0x40)>>7;
mant <<= 6;
expo = nz + 15;
return av_int2sf(mant, 30 - expo);
}

static av_always_inline void autocorrelate(const int x[40][2], SoftFloat phi[3][2][2], int lag)
{
int i;
int64_t real_sum, imag_sum;
int64_t accu_re = 0, accu_im = 0;

if (lag) {
for (i = 1; i < 38; i++) {
accu_re += (int64_t)x[i][0] * x[i+lag][0];
accu_re += (int64_t)x[i][1] * x[i+lag][1];
accu_im += (int64_t)x[i][0] * x[i+lag][1];
accu_im -= (int64_t)x[i][1] * x[i+lag][0];
}

real_sum = accu_re;
imag_sum = accu_im;

accu_re += (int64_t)x[ 0][0] * x[lag][0];
accu_re += (int64_t)x[ 0][1] * x[lag][1];
accu_im += (int64_t)x[ 0][0] * x[lag][1];
accu_im -= (int64_t)x[ 0][1] * x[lag][0];

phi[2-lag][1][0] = autocorr_calc(accu_re);
phi[2-lag][1][1] = autocorr_calc(accu_im);

if (lag == 1) {
accu_re = real_sum;
accu_im = imag_sum;
accu_re += (int64_t)x[38][0] * x[39][0];
accu_re += (int64_t)x[38][1] * x[39][1];
accu_im += (int64_t)x[38][0] * x[39][1];
accu_im -= (int64_t)x[38][1] * x[39][0];

phi[0][0][0] = autocorr_calc(accu_re);
phi[0][0][1] = autocorr_calc(accu_im);
}
} else {
for (i = 1; i < 38; i++) {
accu_re += (int64_t)x[i][0] * x[i][0];
accu_re += (int64_t)x[i][1] * x[i][1];
}
real_sum = accu_re;
accu_re += (int64_t)x[ 0][0] * x[ 0][0];
accu_re += (int64_t)x[ 0][1] * x[ 0][1];

phi[2][1][0] = autocorr_calc(accu_re);

accu_re = real_sum;
accu_re += (int64_t)x[38][0] * x[38][0];
accu_re += (int64_t)x[38][1] * x[38][1];

phi[1][0][0] = autocorr_calc(accu_re);
}
}

static void sbr_autocorrelate_c(const int x[40][2], SoftFloat phi[3][2][2])
{
autocorrelate(x, phi, 0);
autocorrelate(x, phi, 1);
autocorrelate(x, phi, 2);
}

static void sbr_hf_gen_c(int (*X_high)[2], const int (*X_low)[2],
const int alpha0[2], const int alpha1[2],
int bw, int start, int end)
{
int alpha[4];
int i;
int64_t accu;

accu = (int64_t)alpha0[0] * bw;
alpha[2] = (int)((accu + 0x40000000) >> 31);
accu = (int64_t)alpha0[1] * bw;
alpha[3] = (int)((accu + 0x40000000) >> 31);
accu = (int64_t)bw * bw;
bw = (int)((accu + 0x40000000) >> 31);
accu = (int64_t)alpha1[0] * bw;
alpha[0] = (int)((accu + 0x40000000) >> 31);
accu = (int64_t)alpha1[1] * bw;
alpha[1] = (int)((accu + 0x40000000) >> 31);

for (i = start; i < end; i++) {
accu = (int64_t)X_low[i][0] * 0x20000000;
accu += (int64_t)X_low[i - 2][0] * alpha[0];
accu -= (int64_t)X_low[i - 2][1] * alpha[1];
accu += (int64_t)X_low[i - 1][0] * alpha[2];
accu -= (int64_t)X_low[i - 1][1] * alpha[3];
X_high[i][0] = (int)((accu + 0x10000000) >> 29);

accu = (int64_t)X_low[i][1] * 0x20000000;
accu += (int64_t)X_low[i - 2][1] * alpha[0];
accu += (int64_t)X_low[i - 2][0] * alpha[1];
accu += (int64_t)X_low[i - 1][1] * alpha[2];
accu += (int64_t)X_low[i - 1][0] * alpha[3];
X_high[i][1] = (int)((accu + 0x10000000) >> 29);
}
}

static void sbr_hf_g_filt_c(int (*Y)[2], const int (*X_high)[40][2],
const SoftFloat *g_filt, int m_max, intptr_t ixh)
{
int m, r;
int64_t accu;

for (m = 0; m < m_max; m++) {
r = 1 << (22-g_filt[m].exp);
accu = (int64_t)X_high[m][ixh][0] * ((g_filt[m].mant + 0x40)>>7);
Y[m][0] = (int)((accu + r) >> (23-g_filt[m].exp));

accu = (int64_t)X_high[m][ixh][1] * ((g_filt[m].mant + 0x40)>>7);
Y[m][1] = (int)((accu + r) >> (23-g_filt[m].exp));
}
}

static av_always_inline void sbr_hf_apply_noise(int (*Y)[2],
const SoftFloat *s_m,
const SoftFloat *q_filt,
int noise,
int phi_sign0,
int phi_sign1,
int m_max)
{
int m;

for (m = 0; m < m_max; m++) {
int y0 = Y[m][0];
int y1 = Y[m][1];
noise = (noise + 1) & 0x1ff;
if (s_m[m].mant) {
int shift, round;

shift = 22 - s_m[m].exp;
if (shift < 30) {
round = 1 << (shift-1);
y0 += (s_m[m].mant * phi_sign0 + round) >> shift;
y1 += (s_m[m].mant * phi_sign1 + round) >> shift;
}
} else {
int shift, round, tmp;
int64_t accu;

shift = 22 - q_filt[m].exp;
if (shift < 30) {
round = 1 << (shift-1);

accu = (int64_t)q_filt[m].mant * ff_sbr_noise_table_fixed[noise][0];
tmp = (int)((accu + 0x40000000) >> 31);
y0 += (tmp + round) >> shift;

accu = (int64_t)q_filt[m].mant * ff_sbr_noise_table_fixed[noise][1];
tmp = (int)((accu + 0x40000000) >> 31);
y1 += (tmp + round) >> shift;
}
}
Y[m][0] = y0;
Y[m][1] = y1;
phi_sign1 = -phi_sign1;
}
}

#include "sbrdsp_template.c"

+ 22
- 20
libavcodec/sbrdsp_template.c View File

@@ -20,55 +20,55 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

static void sbr_sum64x5_c(float *z)
static void sbr_sum64x5_c(INTFLOAT *z)
{
int k;
for (k = 0; k < 64; k++) {
float f = z[k] + z[k + 64] + z[k + 128] + z[k + 192] + z[k + 256];
INTFLOAT f = z[k] + z[k + 64] + z[k + 128] + z[k + 192] + z[k + 256];
z[k] = f;
}
}

static void sbr_qmf_deint_bfly_c(float *v, const float *src0, const float *src1)
static void sbr_qmf_deint_bfly_c(INTFLOAT *v, const INTFLOAT *src0, const INTFLOAT *src1)
{
int i;
for (i = 0; i < 64; i++) {
v[ i] = src0[i] - src1[63 - i];
v[127 - i] = src0[i] + src1[63 - i];
v[ i] = AAC_SRA_R((src0[i] - src1[63 - i]), 5);
v[127 - i] = AAC_SRA_R((src0[i] + src1[63 - i]), 5);
}
}

static void sbr_hf_apply_noise_0(float (*Y)[2], const float *s_m,
const float *q_filt, int noise,
static void sbr_hf_apply_noise_0(INTFLOAT (*Y)[2], const AAC_FLOAT *s_m,
const AAC_FLOAT *q_filt, int noise,
int kx, int m_max)
{
sbr_hf_apply_noise(Y, s_m, q_filt, noise, 1.0, 0.0, m_max);
sbr_hf_apply_noise(Y, s_m, q_filt, noise, (INTFLOAT)1.0, (INTFLOAT)0.0, m_max);
}

static void sbr_hf_apply_noise_1(float (*Y)[2], const float *s_m,
const float *q_filt, int noise,
static void sbr_hf_apply_noise_1(INTFLOAT (*Y)[2], const AAC_FLOAT *s_m,
const AAC_FLOAT *q_filt, int noise,
int kx, int m_max)
{
float phi_sign = 1 - 2 * (kx & 1);
sbr_hf_apply_noise(Y, s_m, q_filt, noise, 0.0, phi_sign, m_max);
INTFLOAT phi_sign = 1 - 2 * (kx & 1);
sbr_hf_apply_noise(Y, s_m, q_filt, noise, (INTFLOAT)0.0, phi_sign, m_max);
}

static void sbr_hf_apply_noise_2(float (*Y)[2], const float *s_m,
const float *q_filt, int noise,
static void sbr_hf_apply_noise_2(INTFLOAT (*Y)[2], const AAC_FLOAT *s_m,
const AAC_FLOAT *q_filt, int noise,
int kx, int m_max)
{
sbr_hf_apply_noise(Y, s_m, q_filt, noise, -1.0, 0.0, m_max);
sbr_hf_apply_noise(Y, s_m, q_filt, noise, (INTFLOAT)-1.0, (INTFLOAT)0.0, m_max);
}

static void sbr_hf_apply_noise_3(float (*Y)[2], const float *s_m,
const float *q_filt, int noise,
static void sbr_hf_apply_noise_3(INTFLOAT (*Y)[2], const AAC_FLOAT *s_m,
const AAC_FLOAT *q_filt, int noise,
int kx, int m_max)
{
float phi_sign = 1 - 2 * (kx & 1);
sbr_hf_apply_noise(Y, s_m, q_filt, noise, 0.0, -phi_sign, m_max);
INTFLOAT phi_sign = 1 - 2 * (kx & 1);
sbr_hf_apply_noise(Y, s_m, q_filt, noise, (INTFLOAT)0.0, -phi_sign, m_max);
}

av_cold void ff_sbrdsp_init(SBRDSPContext *s)
av_cold void AAC_RENAME(ff_sbrdsp_init)(SBRDSPContext *s)
{
s->sum64x5 = sbr_sum64x5_c;
s->sum_square = sbr_sum_square_c;
@@ -86,10 +86,12 @@ av_cold void ff_sbrdsp_init(SBRDSPContext *s)
s->hf_apply_noise[2] = sbr_hf_apply_noise_2;
s->hf_apply_noise[3] = sbr_hf_apply_noise_3;

#if !USE_FIXED
if (ARCH_ARM)
ff_sbrdsp_init_arm(s);
if (ARCH_X86)
ff_sbrdsp_init_x86(s);
if (ARCH_MIPS)
ff_sbrdsp_init_mips(s);
#endif /* !USE_FIXED */
}

+ 8
- 0
libavutil/softfloat.h View File

@@ -36,6 +36,14 @@ typedef struct SoftFloat{
int32_t exp;
}SoftFloat;

static const SoftFloat FLOAT_0 = { 0, 0};
static const SoftFloat FLOAT_05 = { 0x20000000, 0};
static const SoftFloat FLOAT_1 = { 0x20000000, 1};
static const SoftFloat FLOAT_EPSILON = { 0x29F16B12, -16};
static const SoftFloat FLOAT_1584893192 = { 0x32B771ED, 1};
static const SoftFloat FLOAT_100000 = { 0x30D40000, 17};
static const SoftFloat FLOAT_0999999 = { 0x3FFFFBCE, 0};

static inline av_const double av_sf2double(SoftFloat v) {
v.exp -= ONE_BITS +1;
if(v.exp > 0) return (double)v.mant * (double)(1 << v.exp);


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