mirror of https://git.ffmpeg.org/ffmpeg.git
474 lines
14 KiB
C
474 lines
14 KiB
C
/*
|
|
* 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 USE_FIXED 1
|
|
#define TX_TYPE AV_TX_INT32_MDCT
|
|
|
|
#include "libavutil/fixed_dsp.h"
|
|
#include "libavutil/opt.h"
|
|
#include "avcodec.h"
|
|
#include "codec_internal.h"
|
|
#include "get_bits.h"
|
|
#include "lpc.h"
|
|
#include "kbdwin.h"
|
|
#include "sinewin_fixed_tablegen.h"
|
|
|
|
#include "aac.h"
|
|
#include "aactab.h"
|
|
#include "aacdectab.h"
|
|
#include "adts_header.h"
|
|
#include "cbrt_data.h"
|
|
#include "sbr.h"
|
|
#include "aacsbr.h"
|
|
#include "mpeg4audio.h"
|
|
#include "profiles.h"
|
|
#include "libavutil/intfloat.h"
|
|
|
|
#include <math.h>
|
|
#include <string.h>
|
|
|
|
DECLARE_ALIGNED(32, static int, AAC_RENAME2(aac_kbd_long_1024))[1024];
|
|
DECLARE_ALIGNED(32, static int, AAC_RENAME2(aac_kbd_short_128))[128];
|
|
DECLARE_ALIGNED(32, static int, AAC_RENAME2(aac_kbd_long_960))[960];
|
|
DECLARE_ALIGNED(32, static int, AAC_RENAME2(aac_kbd_short_120))[120];
|
|
|
|
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) & 8191];
|
|
else
|
|
coef = (int)ff_cbrt_tab_fixed[ coef & 8191];
|
|
coefs[i] = coef;
|
|
}
|
|
}
|
|
|
|
static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
|
|
{
|
|
int ssign = scale < 0 ? -1 : 1;
|
|
int s = FFABS(scale);
|
|
unsigned int round;
|
|
int i, out, c = exp2tab[s & 3];
|
|
|
|
s = offset - (s >> 2);
|
|
|
|
if (s > 31) {
|
|
for (i=0; i<len; i++) {
|
|
dst[i] = 0;
|
|
}
|
|
} else if (s > 0) {
|
|
round = 1 << (s-1);
|
|
for (i=0; i<len; i++) {
|
|
out = (int)(((int64_t)src[i] * c) >> 32);
|
|
dst[i] = ((int)(out+round) >> s) * ssign;
|
|
}
|
|
} else if (s > -32) {
|
|
s = s + 32;
|
|
round = 1U << (s-1);
|
|
for (i=0; i<len; i++) {
|
|
out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
|
|
dst[i] = out * (unsigned)ssign;
|
|
}
|
|
} else {
|
|
av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
|
|
}
|
|
}
|
|
|
|
static void noise_scale(int *coefs, int scale, int band_energy, int len)
|
|
{
|
|
int s = -scale;
|
|
unsigned int round;
|
|
int i, out, c = exp2tab[s & 3];
|
|
int nlz = 0;
|
|
|
|
av_assert0(s >= 0);
|
|
while (band_energy > 0x7fff) {
|
|
band_energy >>= 1;
|
|
nlz++;
|
|
}
|
|
c /= band_energy;
|
|
s = 21 + nlz - (s >> 2);
|
|
|
|
if (s > 31) {
|
|
for (i=0; i<len; i++) {
|
|
coefs[i] = 0;
|
|
}
|
|
} else if (s >= 0) {
|
|
round = s ? 1 << (s-1) : 0;
|
|
for (i=0; i<len; i++) {
|
|
out = (int)(((int64_t)coefs[i] * c) >> 32);
|
|
coefs[i] = -((int)(out+round) >> s);
|
|
}
|
|
}
|
|
else {
|
|
s = s + 32;
|
|
if (s > 0) {
|
|
round = 1 << (s-1);
|
|
for (i=0; i<len; i++) {
|
|
out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
|
|
coefs[i] = -out;
|
|
}
|
|
} else {
|
|
for (i=0; i<len; i++)
|
|
coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
|
|
}
|
|
}
|
|
}
|
|
|
|
static av_always_inline SoftFloat flt16_round(SoftFloat pf)
|
|
{
|
|
SoftFloat tmp;
|
|
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) {
|
|
if (shift > 0) {
|
|
*coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
|
|
} else
|
|
*coef += (unsigned)pv.mant << -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 + (int64_t)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 * (1U << 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;
|
|
unsigned 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 * (1U << shift);
|
|
}
|
|
}
|
|
}
|
|
|
|
#include "aacdec_template.c"
|
|
|
|
const FFCodec ff_aac_fixed_decoder = {
|
|
.p.name = "aac_fixed",
|
|
CODEC_LONG_NAME("AAC (Advanced Audio Coding)"),
|
|
.p.type = AVMEDIA_TYPE_AUDIO,
|
|
.p.id = AV_CODEC_ID_AAC,
|
|
.priv_data_size = sizeof(AACContext),
|
|
.init = aac_decode_init,
|
|
.close = aac_decode_close,
|
|
FF_CODEC_DECODE_CB(aac_decode_frame),
|
|
.p.sample_fmts = (const enum AVSampleFormat[]) {
|
|
AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
|
|
},
|
|
.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
|
|
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
|
|
CODEC_OLD_CHANNEL_LAYOUTS_ARRAY(ff_aac_channel_layout)
|
|
.p.ch_layouts = ff_aac_ch_layout,
|
|
.p.priv_class = &aac_decoder_class,
|
|
.p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
|
|
.flush = flush,
|
|
};
|