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FFmpeg/libavcodec/aacdec_fixed.c

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/*
* 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
#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_data.h"
#include "sbr.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
#include "aacadtsdec.h"
#include "profiles.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;
}
static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) }; // 2^0, 2^0.25, 2^0.5, 2^0.75
static inline int *DEC_SPAIR(int *dst, unsigned idx)
{
dst[0] = (idx & 15) - 4;
dst[1] = (idx >> 4 & 15) - 4;
return dst + 2;
}
static inline int *DEC_SQUAD(int *dst, unsigned idx)
{
dst[0] = (idx & 3) - 1;
dst[1] = (idx >> 2 & 3) - 1;
dst[2] = (idx >> 4 & 3) - 1;
dst[3] = (idx >> 6 & 3) - 1;
return dst + 4;
}
static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
{
dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
return dst + 2;
}
static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
{
unsigned nz = idx >> 12;
dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
return dst + 4;
}
static void vector_pow43(int *coefs, int len)
{
int i, coef;
for (i=0; i<len; i++) {
coef = coefs[i];
if (coef < 0)
coef = -(int)ff_cbrt_tab_fixed[-coef];
else
coef = (int)ff_cbrt_tab_fixed[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 > 31) {
for (i=0; i<len; i++) {
dst[i] = 0;
}
} else if (s > 0) {
round = 1 << (s-1);
for (i=0; i<len; i++) {
out = (int)(((int64_t)src[i] * c) >> 32);
dst[i] = ((int)(out+round) >> s) * ssign;
}
} else if (s > -32) {
s = s + 32;
round = 1U << (s-1);
for (i=0; i<len; i++) {
out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
dst[i] = out * (unsigned)ssign;
}
} else {
av_log(NULL, AV_LOG_ERROR, "Overflow in subband_scale()\n");
}
}
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 > 31) {
for (i=0; i<len; i++) {
coefs[i] = 0;
}
} else if (s >= 0) {
round = s ? 1 << (s-1) : 0;
for (i=0; i<len; i++) {
out = (int)(((int64_t)coefs[i] * c) >> 32);
coefs[i] = ((int)(out+round) >> s) * 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 < -31) {
// Nothing to do
} else 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 * (1 << 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 < -31) {
return;
} else 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 = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
.channel_layouts = aac_channel_layout,
.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
.flush = flush,
};