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libavcodec: Implementation of AAC_fixed_decoder (LC-module) [3/4]

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Add fixed point implementation

Signed-off-by: Nedeljko Babic <nedeljko.babic@imgtec.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Djordje Pesut 2015-06-30 11:53:05 +02:00 committed by Michael Niedermayer
parent 08be74ac81
commit b04f46cb4b
5 changed files with 809 additions and 146 deletions
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80
libavcodec/aac.h
View File

@ -41,21 +41,53 @@
#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 "libavutil/float_dsp.h"
#include "libavutil/fixed_dsp.h"
#include "avcodec.h"
#if !USE_FIXED
#include "imdct15.h"
#endif
#include "fft.h"
#include "mpeg4audio.h"
#include "sbr.h"
@ -149,12 +181,12 @@ typedef struct OutputConfiguration {
* Predictor State
*/
typedef struct PredictorState {
float cor0;
float cor1;
float var0;
float var1;
float r0;
float r1;
AAC_FLOAT cor0;
AAC_FLOAT cor1;
AAC_FLOAT var0;
AAC_FLOAT var1;
AAC_FLOAT r0;
AAC_FLOAT r1;
} PredictorState;
#define MAX_PREDICTORS 672
@ -175,7 +207,7 @@ typedef struct PredictorState {
typedef struct LongTermPrediction {
int8_t present;
int16_t lag;
float coef;
INTFLOAT coef;
int8_t used[MAX_LTP_LONG_SFB];
} LongTermPrediction;
@ -209,7 +241,7 @@ typedef struct TemporalNoiseShaping {
int length[8][4];
int direction[8][4];
int order[8][4];
float coef[8][4][TNS_MAX_ORDER];
INTFLOAT coef[8][4][TNS_MAX_ORDER];
} TemporalNoiseShaping;
/**
@ -246,7 +278,7 @@ typedef struct ChannelCoupling {
int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel;
* [2] list of gains for left channel; [3] lists of gains for both channels
*/
float gain[16][120];
INTFLOAT gain[16][120];
} ChannelCoupling;
/**
@ -258,18 +290,18 @@ typedef struct SingleChannelElement {
Pulse pulse;
enum BandType band_type[128]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
INTFLOAT sf[120]; ///< scalefactors
int sf_idx[128]; ///< scalefactor indices (used by encoder)
uint8_t zeroes[128]; ///< band is not coded (used by encoder)
float is_ener[128]; ///< Intensity stereo pos (used by encoder)
float pns_ener[128]; ///< Noise energy values (used by encoder)
DECLARE_ALIGNED(32, float, pcoeffs)[1024]; ///< coefficients for IMDCT, pristine
DECLARE_ALIGNED(32, float, coeffs)[1024]; ///< coefficients for IMDCT, maybe processed
DECLARE_ALIGNED(32, float, saved)[1536]; ///< overlap
DECLARE_ALIGNED(32, float, ret_buf)[2048]; ///< PCM output buffer
DECLARE_ALIGNED(16, float, ltp_state)[3072]; ///< time signal for LTP
DECLARE_ALIGNED(32, INTFLOAT, pcoeffs)[1024]; ///< coefficients for IMDCT, pristine
DECLARE_ALIGNED(32, INTFLOAT, coeffs)[1024]; ///< coefficients for IMDCT, maybe processed
DECLARE_ALIGNED(32, INTFLOAT, saved)[1536]; ///< overlap
DECLARE_ALIGNED(32, INTFLOAT, ret_buf)[2048]; ///< PCM output buffer
DECLARE_ALIGNED(16, INTFLOAT, ltp_state)[3072]; ///< time signal for LTP
PredictorState predictor_state[MAX_PREDICTORS];
float *ret; ///< PCM output
INTFLOAT *ret; ///< PCM output
} SingleChannelElement;
/**
@ -316,7 +348,7 @@ struct AACContext {
* (We do not want to have these on the stack.)
* @{
*/
DECLARE_ALIGNED(32, float, buf_mdct)[1024];
DECLARE_ALIGNED(32, INTFLOAT, buf_mdct)[1024];
/** @} */
/**
@ -327,8 +359,12 @@ struct AACContext {
FFTContext mdct_small;
FFTContext mdct_ld;
FFTContext mdct_ltp;
#if USE_FIXED
AVFixedDSPContext *fdsp;
#else
IMDCT15Context *mdct480;
AVFloatDSPContext *fdsp;
#endif /* USE_FIXED */
int random_state;
/** @} */
@ -348,7 +384,7 @@ struct AACContext {
int dmono_mode; ///< 0->not dmono, 1->use first channel, 2->use second channel
/** @} */
DECLARE_ALIGNED(32, float, temp)[128];
DECLARE_ALIGNED(32, INTFLOAT, temp)[128];
OutputConfiguration oc[2];
int warned_num_aac_frames;
@ -356,11 +392,13 @@ struct AACContext {
/* aacdec functions pointers */
void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
void (*apply_tns)(float coef[1024], TemporalNoiseShaping *tns,
void (*apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
IndividualChannelStream *ics, int decode);
void (*windowing_and_mdct_ltp)(AACContext *ac, float *out,
float *in, IndividualChannelStream *ics);
void (*windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out,
INTFLOAT *in, IndividualChannelStream *ics);
void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
void (*vector_pow43)(int *coefs, int len);
void (*subband_scale)(int *dst, int *src, int scale, int offset, int len);
};

5
libavcodec/aacdec.c
View File

@ -32,6 +32,11 @@
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#define FFT_FLOAT 1
#define FFT_FIXED_32 0
#define USE_FIXED 0
#define CONFIG_FIXED 0
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "avcodec.h"

444
libavcodec/aacdec_fixed.c Normal file
View File

@ -0,0 +1,444 @@
/*
* 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 decoder fixed-point implementation
*
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov 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 decoder
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*
* Fixed point implementation
* @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
*/
#define FFT_FLOAT 0
#define FFT_FIXED_32 1
#define USE_FIXED 1
#define CONFIG_FIXED 1
#include "libavutil/fixed_dsp.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "fft.h"
#include "lpc.h"
#include "kbdwin.h"
#include "sinewin.h"
#include "aac.h"
#include "aactab.h"
#include "aacdectab.h"
#include "cbrt_tablegen.h"
#include "sbr.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
#include "aacadtsdec.h"
#include "libavutil/intfloat.h"
#include <math.h>
#include <string.h>
static av_always_inline void reset_predict_state(PredictorState *ps)
{
ps->r0.mant = 0;
ps->r0.exp = 0;
ps->r1.mant = 0;
ps->r1.exp = 0;
ps->cor0.mant = 0;
ps->cor0.exp = 0;
ps->cor1.mant = 0;
ps->cor1.exp = 0;
ps->var0.mant = 0x20000000;
ps->var0.exp = 1;
ps->var1.mant = 0x20000000;
ps->var1.exp = 1;
}
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) << 1));
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) << 1));
sign <<= nz & 1;
nz >>= 1;
dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) << 1));
sign <<= nz & 1;
nz >>= 1;
dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) << 1));
sign <<= nz & 1;
nz >>= 1;
dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) << 1));
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)cbrt_tab[-coef];
else
coef = (int)cbrt_tab[coef];
coefs[i] = coef;
}
}
static void subband_scale(int *dst, int *src, int scale, int offset, int len)
{
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 > 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 {
s = s + 32;
round = 1 << (s-1);
for (i=0; i<len; i++) {
out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
dst[i] = out * ssign;
}
}
}
static void noise_scale(int *coefs, int scale, int band_energy, int len)
{
int ssign = scale < 0 ? -1 : 1;
int s = FFABS(scale);
unsigned int round;
int i, out, c = exp2tab[s & 3];
int nlz = 0;
while (band_energy > 0x7fff) {
band_energy >>= 1;
nlz++;
}
c /= band_energy;
s = 21 + nlz - (s >> 2);
if (s > 0) {
round = 1 << (s-1);
for (i=0; i<len; i++) {
out = (int)(((int64_t)coefs[i] * c) >> 32);
coefs[i] = ((int)(out+round) >> s) * ssign;
}
}
else {
s = s + 32;
round = 1 << (s-1);
for (i=0; i<len; i++) {
out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
coefs[i] = out * ssign;
}
}
}
static av_always_inline SoftFloat flt16_round(SoftFloat pf)
{
SoftFloat tmp;
int s;
tmp.exp = pf.exp;
s = pf.mant >> 31;
tmp.mant = (pf.mant ^ s) - s;
tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
tmp.mant = (tmp.mant ^ s) - s;
return tmp;
}
static av_always_inline SoftFloat flt16_even(SoftFloat pf)
{
SoftFloat tmp;
int s;
tmp.exp = pf.exp;
s = pf.mant >> 31;
tmp.mant = (pf.mant ^ s) - s;
tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
tmp.mant = (tmp.mant ^ s) - s;
return tmp;
}
static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
{
SoftFloat pun;
int s;
pun.exp = pf.exp;
s = pf.mant >> 31;
pun.mant = (pf.mant ^ s) - s;
pun.mant = pun.mant & 0xFFC00000U;
pun.mant = (pun.mant ^ s) - s;
return pun;
}
static av_always_inline void predict(PredictorState *ps, int *coef,
int output_enable)
{
const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64
const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32
SoftFloat e0, e1;
SoftFloat pv;
SoftFloat k1, k2;
SoftFloat r0 = ps->r0, r1 = ps->r1;
SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
SoftFloat var0 = ps->var0, var1 = ps->var1;
SoftFloat tmp;
if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
}
else {
k1.mant = 0;
k1.exp = 0;
}
if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
}
else {
k2.mant = 0;
k2.exp = 0;
}
tmp = av_mul_sf(k1, r0);
pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
if (output_enable) {
int shift = 28 - pv.exp;
if (shift < 31)
*coef += (pv.mant + (1 << (shift - 1))) >> shift;
}
e0 = av_int2sf(*coef, 2);
e1 = av_sub_sf(e0, tmp);
ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
tmp.exp--;
ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
tmp.exp--;
ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
ps->r0 = flt16_trunc(av_mul_sf(a, e0));
}
static const int cce_scale_fixed[8] = {
Q30(1.0), //2^(0/8)
Q30(1.0905077327), //2^(1/8)
Q30(1.1892071150), //2^(2/8)
Q30(1.2968395547), //2^(3/8)
Q30(1.4142135624), //2^(4/8)
Q30(1.5422108254), //2^(5/8)
Q30(1.6817928305), //2^(6/8)
Q30(1.8340080864), //2^(7/8)
};
/**
* Apply dependent channel coupling (applied before IMDCT).
*
* @param index index into coupling gain array
*/
static void apply_dependent_coupling_fixed(AACContext *ac,
SingleChannelElement *target,
ChannelElement *cce, int index)
{
IndividualChannelStream *ics = &cce->ch[0].ics;
const uint16_t *offsets = ics->swb_offset;
int *dest = target->coeffs;
const int *src = cce->ch[0].coeffs;
int g, i, group, k, idx = 0;
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
av_log(ac->avctx, AV_LOG_ERROR,
"Dependent coupling is not supported together with LTP\n");
return;
}
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb; i++, idx++) {
if (cce->ch[0].band_type[idx] != ZERO_BT) {
const int gain = cce->coup.gain[index][idx];
int shift, round, c, tmp;
if (gain < 0) {
c = -cce_scale_fixed[-gain & 7];
shift = (-gain-1024) >> 3;
}
else {
c = cce_scale_fixed[gain & 7];
shift = (gain-1024) >> 3;
}
if (shift < 0) {
shift = -shift;
round = 1 << (shift - 1);
for (group = 0; group < ics->group_len[g]; group++) {
for (k = offsets[i]; k < offsets[i + 1]; k++) {
tmp = (int)(((int64_t)src[group * 128 + k] * c + \
(int64_t)0x1000000000) >> 37);
dest[group * 128 + k] += (tmp + round) >> shift;
}
}
}
else {
for (group = 0; group < ics->group_len[g]; group++) {
for (k = offsets[i]; k < offsets[i + 1]; k++) {
tmp = (int)(((int64_t)src[group * 128 + k] * c + \
(int64_t)0x1000000000) >> 37);
dest[group * 128 + k] += tmp << shift;
}
}
}
}
}
dest += ics->group_len[g] * 128;
src += ics->group_len[g] * 128;
}
}
/**
* Apply independent channel coupling (applied after IMDCT).
*
* @param index index into coupling gain array
*/
static void apply_independent_coupling_fixed(AACContext *ac,
SingleChannelElement *target,
ChannelElement *cce, int index)
{
int i, c, shift, round, tmp;
const int gain = cce->coup.gain[index][0];
const int *src = cce->ch[0].ret;
int *dest = target->ret;
const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
c = cce_scale_fixed[gain & 7];
shift = (gain-1024) >> 3;
if (shift < 0) {
shift = -shift;
round = 1 << (shift - 1);
for (i = 0; i < len; i++) {
tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
dest[i] += (tmp + round) >> shift;
}
}
else {
for (i = 0; i < len; i++) {
tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
dest[i] += tmp << shift;
}
}
}
#include "aacdec_template.c"
AVCodec ff_aac_fixed_decoder = {
.name = "aac_fixed",
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AAC,
.priv_data_size = sizeof(AACContext),
.init = aac_decode_init,
.close = aac_decode_close,
.decode = aac_decode_frame,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
},
.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
.flush = flush,
};

421
libavcodec/aacdec_template.c
View File

@ -8,6 +8,10 @@
* 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
@ -30,6 +34,10 @@
* AAC decoder
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*
* AAC decoder fixed-point implementation
* @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
* @author Nedeljko Babic ( nedeljko.babic imgtec com )
*/
/*
@ -173,7 +181,7 @@ static int frame_configure_elements(AVCodecContext *avctx)
/* map output channel pointers to AVFrame data */
for (ch = 0; ch < avctx->channels; ch++) {
if (ac->output_element[ch])
ac->output_element[ch]->ret = (float *)ac->frame->extended_data[ch];
ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
}
return 0;
@ -866,8 +874,14 @@ static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
m4ac->ps = 0;
m4ac->sbr = 0;
#if USE_FIXED
if (get_bits1(gb)) { // frameLengthFlag
avpriv_request_sample(avctx, "960/120 MDCT window");
return AVERROR_PATCHWELCOME;
}
#else
m4ac->frame_length_short = get_bits1(gb);
#endif
res_flags = get_bits(gb, 3);
if (res_flags) {
avpriv_report_missing_feature(avctx,
@ -1053,8 +1067,11 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
aacdec_init(ac);
#if USE_FIXED
avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
#else
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
#endif /* USE_FIXED */
if (avctx->extradata_size > 0) {
if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
@ -1111,7 +1128,11 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ff_aac_sbr_init();
#if USE_FIXED
ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & CODEC_FLAG_BITEXACT);
#else
ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
#endif /* USE_FIXED */
if (!ac->fdsp) {
return AVERROR(ENOMEM);
}
@ -1130,22 +1151,23 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
sizeof(ff_aac_scalefactor_code[0]),
352);
ff_mdct_init(&ac->mdct, 11, 1, 1.0 / (32768.0 * 1024.0));
ff_mdct_init(&ac->mdct_ld, 10, 1, 1.0 / (32768.0 * 512.0));
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0 / (32768.0 * 128.0));
ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0 * 32768.0);
AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
#if !USE_FIXED
ret = ff_imdct15_init(&ac->mdct480, 5);
if (ret < 0)
return ret;
#endif
// window initialization
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
ff_init_ff_sine_windows(10);
ff_init_ff_sine_windows( 9);
ff_init_ff_sine_windows( 7);
AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024);
AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128);
AAC_RENAME(ff_init_ff_sine_windows)(10);
AAC_RENAME(ff_init_ff_sine_windows)( 9);
AAC_RENAME(ff_init_ff_sine_windows)( 7);
cbrt_tableinit();
AAC_RENAME(cbrt_tableinit)();
return 0;
}
@ -1366,7 +1388,7 @@ static int decode_band_types(AACContext *ac, enum BandType band_type[120],
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb,
unsigned int global_gain,
IndividualChannelStream *ics,
enum BandType band_type[120],
@ -1381,7 +1403,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
int run_end = band_type_run_end[idx];
if (band_type[idx] == ZERO_BT) {
for (; i < run_end; i++, idx++)
sf[idx] = 0.0;
sf[idx] = FIXR(0.);
} else if ((band_type[idx] == INTENSITY_BT) ||
(band_type[idx] == INTENSITY_BT2)) {
for (; i < run_end; i++, idx++) {
@ -1393,7 +1415,11 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
"Clipped intensity stereo position (%d -> %d)",
offset[2], clipped_offset);
}
#if USE_FIXED
sf[idx] = 100 - clipped_offset;
#else
sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
#endif /* USE_FIXED */
}
} else if (band_type[idx] == NOISE_BT) {
for (; i < run_end; i++, idx++) {
@ -1408,7 +1434,11 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
"Clipped noise gain (%d -> %d)",
offset[1], clipped_offset);
}
#if USE_FIXED
sf[idx] = -(100 + clipped_offset);
#else
sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
#endif /* USE_FIXED */
}
} else {
for (; i < run_end; i++, idx++) {
@ -1418,7 +1448,11 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
"Scalefactor (%d) out of range.\n", offset[0]);
return AVERROR_INVALIDDATA;
}
#if USE_FIXED
sf[idx] = -offset[0];
#else
sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
#endif /* USE_FIXED */
}
}
}
@ -1524,8 +1558,8 @@ static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
GetBitContext *gb, const float sf[120],
static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
GetBitContext *gb, const INTFLOAT sf[120],
int pulse_present, const Pulse *pulse,
const IndividualChannelStream *ics,
enum BandType band_type[120])
@ -1533,49 +1567,63 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
int i, k, g, idx = 0;
const int c = 1024 / ics->num_windows;
const uint16_t *offsets = ics->swb_offset;
float *coef_base = coef;
INTFLOAT *coef_base = coef;
for (g = 0; g < ics->num_windows; g++)
memset(coef + g * 128 + offsets[ics->max_sfb], 0,
sizeof(float) * (c - offsets[ics->max_sfb]));
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;
float *cfo = coef + offsets[i];
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 < g_len; group++, cfo+=128) {
memset(cfo, 0, off_len * sizeof(float));
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 < g_len; group++, cfo+=128) {
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
#if !USE_FIXED
float scale;
float band_energy;
#endif /* !USE_FIXED */
INTFLOAT band_energy;
for (k = 0; k < off_len; k++) {
ac->random_state = lcg_random(ac->random_state);
#if USE_FIXED
cfo[k] = ac->random_state >> 3;
#else
cfo[k] = ac->random_state;
#endif /* USE_FIXED */
}
#if USE_FIXED
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
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 uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
OPEN_READER(re, gb);
switch (cbt_m1 >> 1) {
case 0:
for (group = 0; group < g_len; group++, cfo+=128) {
float *cf = cfo;
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
@ -1585,14 +1633,18 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_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 < g_len; group++, cfo+=128) {
float *cf = cfo;
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
@ -1607,14 +1659,18 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
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 < g_len; group++, cfo+=128) {
float *cf = cfo;
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
@ -1624,15 +1680,19 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
UPDATE_CACHE(re, gb);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_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 < g_len; group++, cfo+=128) {
float *cf = cfo;
for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
INTFLOAT *cf = cfo;
int len = off_len;
do {
@ -1647,15 +1707,24 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
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 < g_len; group++, cfo+=128) {
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 {
@ -1699,18 +1768,33 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
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++ = 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 */
}
}
@ -1723,19 +1807,48 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
if (pulse_present) {
idx = 0;
for (i = 0; i < pulse->num_pulse; i++) {
float co = coef_base[ pulse->pos[i] ];
INTFLOAT co = coef_base[ pulse->pos[i] ];
while (offsets[idx + 1] <= pulse->pos[i])
idx++;
if (band_type[idx] != NOISE_BT && sf[idx]) {
float ico = -pulse->amp[i];
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);
}
}
}
coef += g_len << 7;
}
#endif /* USE_FIXED */
return 0;
}
@ -1784,7 +1897,7 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
Pulse pulse;
TemporalNoiseShaping *tns = &sce->tns;
IndividualChannelStream *ics = &sce->ics;
float *out = sce->coeffs;
INTFLOAT *out = sce->coeffs;
int global_gain, eld_syntax, er_syntax, pulse_present = 0;
int ret;
@ -1858,8 +1971,8 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
{
const IndividualChannelStream *ics = &cpe->ch[0].ics;
float *ch0 = cpe->ch[0].coeffs;
float *ch1 = cpe->ch[1].coeffs;
INTFLOAT *ch0 = cpe->ch[0].coeffs;
INTFLOAT *ch1 = cpe->ch[1].coeffs;
int g, i, group, idx = 0;
const uint16_t *offsets = ics->swb_offset;
for (g = 0; g < ics->num_window_groups; g++) {
@ -1867,10 +1980,17 @@ static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
if (cpe->ms_mask[idx] &&
cpe->ch[0].band_type[idx] < NOISE_BT &&
cpe->ch[1].band_type[idx] < NOISE_BT) {
#if USE_FIXED
for (group = 0; group < ics->group_len[g]; group++) {
ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
ch1 + group * 128 + offsets[i],
offsets[i+1] - offsets[i]);
#else
for (group = 0; group < ics->group_len[g]; group++) {
ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
ch1 + group * 128 + offsets[i],
offsets[i+1] - offsets[i]);
#endif /* USE_FIXED */
}
}
}
@ -1891,11 +2011,11 @@ static void apply_intensity_stereo(AACContext *ac,
{
const IndividualChannelStream *ics = &cpe->ch[1].ics;
SingleChannelElement *sce1 = &cpe->ch[1];
float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
const uint16_t *offsets = ics->swb_offset;
int g, group, i, idx = 0;
int c;
float scale;
INTFLOAT scale;
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb;) {
if (sce1->band_type[idx] == INTENSITY_BT ||
@ -1907,10 +2027,18 @@ static void apply_intensity_stereo(AACContext *ac,
c *= 1 - 2 * cpe->ms_mask[idx];
scale = c * sce1->sf[idx];
for (group = 0; group < ics->group_len[g]; group++)
#if USE_FIXED
ac->subband_scale(coef1 + group * 128 + offsets[i],
coef0 + group * 128 + offsets[i],
scale,
23,
offsets[i + 1] - offsets[i]);
#else
ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
coef0 + group * 128 + offsets[i],
scale,
offsets[i + 1] - offsets[i]);
#endif /* USE_FIXED */
}
} else {
int bt_run_end = sce1->band_type_run_end[idx];
@ -1986,7 +2114,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
int num_gain = 0;
int c, g, sfb, ret;
int sign;
float scale;
INTFLOAT scale;
SingleChannelElement *sce = &che->ch[0];
ChannelCoupling *coup = &che->coup;
@ -2006,7 +2134,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
sign = get_bits(gb, 1);
scale = cce_scale[get_bits(gb, 2)];
scale = AAC_RENAME(cce_scale)[get_bits(gb, 2)];
if ((ret = decode_ics(ac, sce, gb, 0, 0)))
return ret;
@ -2015,11 +2143,11 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
int idx = 0;
int cge = 1;
int gain = 0;
float gain_cache = 1.0;
INTFLOAT gain_cache = FIXR10(1.);
if (c) {
cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
gain_cache = powf(scale, -gain);
gain_cache = GET_GAIN(scale, gain);
}
if (coup->coupling_point == AFTER_IMDCT) {
coup->gain[c][0] = gain_cache;
@ -2036,7 +2164,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
s -= 2 * (t & 0x1);
t >>= 1;
}
gain_cache = powf(scale, -t) * s;
gain_cache = GET_GAIN(scale, t) * s;
}
}
coup->gain[c][idx] = gain_cache;
@ -2210,14 +2338,14 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
* @param decode 1 if tool is used normally, 0 if tool is used in LTP.
* @param coef spectral coefficients
*/
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
IndividualChannelStream *ics, int decode)
{
const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
int w, filt, m, i;
int bottom, top, order, start, end, size, inc;
float lpc[TNS_MAX_ORDER];
float tmp[TNS_MAX_ORDER+1];
INTFLOAT lpc[TNS_MAX_ORDER];
INTFLOAT tmp[TNS_MAX_ORDER+1];
for (w = 0; w < ics->num_windows; w++) {
bottom = ics->num_swb;
@ -2247,13 +2375,13 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
// ar filter
for (m = 0; m < size; m++, start += inc)
for (i = 1; i <= FFMIN(m, order); i++)
coef[start] -= coef[start - i * inc] * lpc[i - 1];
coef[start] -= AAC_MUL26(coef[start - i * inc], lpc[i - 1]);
} else {
// ma filter
for (m = 0; m < size; m++, start += inc) {
tmp[0] = coef[start];
for (i = 1; i <= FFMIN(m, order); i++)
coef[start] += tmp[i] * lpc[i - 1];
coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
for (i = order; i > 0; i--)
tmp[i] = tmp[i - 1];
}
@ -2266,25 +2394,25 @@ static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
* Apply windowing and MDCT to obtain the spectral
* coefficient from the predicted sample by LTP.
*/
static void windowing_and_mdct_ltp(AACContext *ac, float *out,
float *in, IndividualChannelStream *ics)
static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out,
INTFLOAT *in, IndividualChannelStream *ics)
{
const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
} else {
memset(in, 0, 448 * sizeof(float));
memset(in, 0, 448 * sizeof(*in));
ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
}
if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
} else {
ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
memset(in + 1024 + 576, 0, 448 * sizeof(float));
memset(in + 1024 + 576, 0, 448 * sizeof(*in));
}
ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
}
@ -2299,15 +2427,15 @@ static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
int i, sfb;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
float *predTime = sce->ret;
float *predFreq = ac->buf_mdct;
INTFLOAT *predTime = sce->ret;
INTFLOAT *predFreq = ac->buf_mdct;
int16_t num_samples = 2048;
if (ltp->lag < 1024)
num_samples = ltp->lag + 1024;
for (i = 0; i < num_samples; i++)
predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
memset(&predTime[i], 0, (2048 - i) * sizeof(float));
predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
@ -2327,28 +2455,31 @@ static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
float *saved = sce->saved;
float *saved_ltp = sce->coeffs;
const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
INTFLOAT *saved = sce->saved;
INTFLOAT *saved_ltp = sce->coeffs;
const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
int i;
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy(saved_ltp, saved, 512 * sizeof(float));
memset(saved_ltp + 576, 0, 448 * sizeof(float));
memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
for (i = 0; i < 64; i++)
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
memset(saved_ltp + 576, 0, 448 * sizeof(float));
memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
for (i = 0; i < 64; i++)
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
} else { // LONG_STOP or ONLY_LONG
ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
for (i = 0; i < 512; i++)
saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
}
memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
@ -2362,22 +2493,27 @@ static void update_ltp(AACContext *ac, SingleChannelElement *sce)
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
float *in = sce->coeffs;
float *out = sce->ret;
float *saved = sce->saved;
const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
float *buf = ac->buf_mdct;
float *temp = ac->temp;
INTFLOAT *in = sce->coeffs;
INTFLOAT *out = sce->ret;
INTFLOAT *saved = sce->saved;
const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
INTFLOAT *buf = ac->buf_mdct;
INTFLOAT *temp = ac->temp;
int i;
// imdct
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
for (i = 0; i < 1024; i += 128)
ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
} else
} else {
ac->mdct.imdct_half(&ac->mdct, buf, in);
#if USE_FIXED
for (i=0; i<1024; i++)
buf[i] = (buf[i] + 4) >> 3;
#endif /* USE_FIXED */
}
/* window overlapping
* NOTE: To simplify the overlapping code, all 'meaningless' short to long
@ -2389,7 +2525,7 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
(ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
} else {
memcpy( out, saved, 448 * sizeof(float));
memcpy( out, saved, 448 * sizeof(*out));
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
@ -2397,65 +2533,73 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
} else {
ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
memcpy( out + 576, buf + 64, 448 * sizeof(float));
memcpy( out + 576, buf + 64, 448 * sizeof(*out));
}
}
// buffer update
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
memcpy( saved, temp + 64, 64 * sizeof(float));
memcpy( saved, temp + 64, 64 * sizeof(*saved));
ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
memcpy( saved, buf + 512, 448 * sizeof(float));
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
memcpy( saved, buf + 512, 448 * sizeof(*saved));
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
} else { // LONG_STOP or ONLY_LONG
memcpy( saved, buf + 512, 512 * sizeof(float));
memcpy( saved, buf + 512, 512 * sizeof(*saved));
}
}
static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
float *in = sce->coeffs;
float *out = sce->ret;
float *saved = sce->saved;
float *buf = ac->buf_mdct;
INTFLOAT *in = sce->coeffs;
INTFLOAT *out = sce->ret;
INTFLOAT *saved = sce->saved;
INTFLOAT *buf = ac->buf_mdct;
#if USE_FIXED
int i;
#endif /* USE_FIXED */
// imdct
ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
#if USE_FIXED
for (i = 0; i < 1024; i++)
buf[i] = (buf[i] + 2) >> 2;
#endif /* USE_FIXED */
// window overlapping
if (ics->use_kb_window[1]) {
// AAC LD uses a low overlap sine window instead of a KBD window
memcpy(out, saved, 192 * sizeof(float));
ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
memcpy( out + 320, buf + 64, 192 * sizeof(float));
memcpy(out, saved, 192 * sizeof(*out));
ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
memcpy( out + 320, buf + 64, 192 * sizeof(*out));
} else {
ac->fdsp->vector_fmul_window(out, saved, buf, ff_sine_512, 256);
ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
}
// buffer update
memcpy(saved, buf + 256, 256 * sizeof(float));
memcpy(saved, buf + 256, 256 * sizeof(*saved));
}
static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
{
float *in = sce->coeffs;
float *out = sce->ret;
float *saved = sce->saved;
float *buf = ac->buf_mdct;
INTFLOAT *in = sce->coeffs;
INTFLOAT *out = sce->ret;
INTFLOAT *saved = sce->saved;
INTFLOAT *buf = ac->buf_mdct;
int i;
const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
const int n2 = n >> 1;
const int n4 = n >> 2;
const float *const window = n == 480 ? ff_aac_eld_window_480 :
ff_aac_eld_window_512;
const INTFLOAT *const window = n == 480 ? ff_aac_eld_window_480 :
AAC_RENAME(ff_aac_eld_window_512);
// Inverse transform, mapped to the conventional IMDCT by
// Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
@ -2463,14 +2607,22 @@ static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
// International Conference on Audio, Language and Image Processing, ICALIP 2008.
// URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
for (i = 0; i < n2; i+=2) {
float temp;
INTFLOAT temp;
temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
}
#if !USE_FIXED
if (n == 480)
ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960));
else
#endif
ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
#if USE_FIXED
for (i = 0; i < 1024; i++)
buf[i] = (buf[i] + 1) >> 1;
#endif /* USE_FIXED */
for (i = 0; i < n; i+=2) {
buf[i] = -buf[i];
}
@ -2482,26 +2634,26 @@ static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
// The spec says to use samples [0..511] but the reference decoder uses
// samples [128..639].
for (i = n4; i < n2; i ++) {
out[i - n4] = buf[n2 - 1 - i] * window[i - n4] +
saved[ i + n2] * window[i + n - n4] +
-saved[ n + n2 - 1 - i] * window[i + 2*n - n4] +
-saved[2*n + n2 + i] * window[i + 3*n - n4];
out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
}
for (i = 0; i < n2; i ++) {
out[n4 + i] = buf[i] * window[i + n2 - n4] +
-saved[ n - 1 - i] * window[i + n2 + n - n4] +
-saved[ n + i] * window[i + n2 + 2*n - n4] +
saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
}
for (i = 0; i < n4; i ++) {
out[n2 + n4 + i] = buf[ i + n2] * window[i + n - n4] +
-saved[ n2 - 1 - i] * window[i + 2*n - n4] +
-saved[ n + n2 + i] * window[i + 3*n - n4];
out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
}
// buffer update
memmove(saved + n, saved, 2 * n * sizeof(float));
memcpy( saved, buf, n * sizeof(float));
memmove(saved + n, saved, 2 * n * sizeof(*saved));
memcpy( saved, buf, n * sizeof(*saved));
}
/**
@ -2540,7 +2692,7 @@ static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
}
/**
* Convert spectral data to float samples, applying all supported tools as appropriate.
* Convert spectral data to samples, applying all supported tools as appropriate.
*/
static void spectral_to_sample(AACContext *ac)
{
@ -2561,7 +2713,7 @@ static void spectral_to_sample(AACContext *ac)
ChannelElement *che = ac->che[type][i];
if (che && che->present) {
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, AAC_RENAME(apply_dependent_coupling));
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
if (che->ch[0].ics.predictor_present) {
if (che->ch[0].ics.ltp.present)
@ -2575,7 +2727,7 @@ static void spectral_to_sample(AACContext *ac)
if (che->ch[1].tns.present)
ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
if (type <= TYPE_CPE)
apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, AAC_RENAME(apply_dependent_coupling));
if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
imdct_and_window(ac, &che->ch[0]);
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
@ -2590,7 +2742,18 @@ static void spectral_to_sample(AACContext *ac)
}
}
if (type <= TYPE_CCE)
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, AAC_RENAME(apply_independent_coupling));
#if USE_FIXED
{
int j;
/* preparation for resampler */
for(j = 0; j<2048; j++){
che->ch[0].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[0].ret[j]<<7)+0x8000;
che->ch[1].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[1].ret[j]<<7)+0x8000;
}
}
#endif /* USE_FIXED */
che->present = 0;
} else if (che) {
av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
@ -2999,7 +3162,9 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
ff_mdct_end(&ac->mdct_small);
ff_mdct_end(&ac->mdct_ld);
ff_mdct_end(&ac->mdct_ltp);
#if !USE_FIXED
ff_imdct15_uninit(&ac->mdct480);
#endif
av_freep(&ac->fdsp);
return 0;
}
@ -3011,9 +3176,15 @@ static void aacdec_init(AACContext *c)
c->apply_tns = apply_tns;
c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
c->update_ltp = update_ltp;
#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);
#endif /* !USE_FIXED */
}
/**
* AVOptions for Japanese DTV specific extensions (ADTS only)

5
libavcodec/mdct_template.c
View File

@ -81,8 +81,13 @@ av_cold int ff_mdct_init(FFTContext *s, int nbits, int inverse, double scale)
scale = sqrt(fabs(scale));
for(i=0;i<n4;i++) {
alpha = 2 * M_PI * (i + theta) / n;
#if FFT_FIXED_32
s->tcos[i*tstep] = (FFTSample)floor(-cos(alpha) * 2147483648.0 + 0.5);
s->tsin[i*tstep] = (FFTSample)floor(-sin(alpha) * 2147483648.0 + 0.5);
#else
s->tcos[i*tstep] = FIX15(-cos(alpha) * scale);
s->tsin[i*tstep] = FIX15(-sin(alpha) * scale);
#endif
}
return 0;
fail: