/* Copyright (c) 2007-2008 CSIRO

Copyright (c) 2007-2010 Xiph.Org Foundation

Copyright (c) 2008 Gregory Maxwell

Written by Jean-Marc Valin and Gregory Maxwell */

/*

Redistribution and use in source and binary forms, with or without

modification, are permitted provided that the following conditions

are met:

- Redistributions of source code must retain the above copyright

notice, this list of conditions and the following disclaimer.

- 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.

THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS

``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 COPYRIGHT OWNER

OR CONTRIBUTORS 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.

*/

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#define CELT_DECODER_C

#include "cpu_support.h"

#include "os_support.h"

#include "mdct.h"

#include <math.h>

#include "celt.h"

#include "pitch.h"

#include "bands.h"

#include "modes.h"

#include "entcode.h"

#include "quant_bands.h"

#include "rate.h"

#include "stack_alloc.h"

#include "mathops.h"

#include "float_cast.h"

#include <stdarg.h>

#include "celt_lpc.h"

#include "vq.h"

/* The maximum pitch lag to allow in the pitch-based PLC. It's possible to save

CPU time in the PLC pitch search by making this smaller than MAX_PERIOD. The

current value corresponds to a pitch of 66.67 Hz. */

#define PLC_PITCH_LAG_MAX (720)

/* The minimum pitch lag to allow in the pitch-based PLC. This corresponds to a

pitch of 480 Hz. */

#define PLC_PITCH_LAG_MIN (100)

#if defined(SMALL_FOOTPRINT) && defined(FIXED_POINT)

#define NORM_ALIASING_HACK

#endif

/**********************************************************************/

/* */

/* DECODER */

/* */

/**********************************************************************/

#define DECODE_BUFFER_SIZE 2048

/** Decoder state

@brief Decoder state

*/

struct OpusCustomDecoder {

};

#if defined(ENABLE_HARDENING) || defined(ENABLE_ASSERTIONS)

/* Make basic checks on the CELT state to ensure we don't end

up writing all over memory. */

void validate_celt_decoder(CELTDecoder *st)

{

#ifndef CUSTOM_MODES

celt_assert(st->mode == opus_custom_mode_create(48000, 960, NULL));

celt_assert(st->overlap == 120);

#endif

celt_assert(st->channels == 1 || st->channels == 2);

celt_assert(st->stream_channels == 1 || st->stream_channels == 2);

celt_assert(st->downsample > 0);

celt_assert(st->start == 0 || st->start == 17);

celt_assert(st->start < st->end);

celt_assert(st->end <= 21);

#ifdef OPUS_ARCHMASK

celt_assert(st->arch >= 0);

celt_assert(st->arch <= OPUS_ARCHMASK);

#endif

celt_assert(st->last_pitch_index <= PLC_PITCH_LAG_MAX);

celt_assert(st->last_pitch_index >= PLC_PITCH_LAG_MIN || st->last_pitch_index == 0);

celt_assert(st->postfilter_period < MAX_PERIOD);

celt_assert(st->postfilter_period >= COMBFILTER_MINPERIOD || st->postfilter_period == 0);

celt_assert(st->postfilter_period_old < MAX_PERIOD);

celt_assert(st->postfilter_period_old >= COMBFILTER_MINPERIOD || st->postfilter_period_old == 0);

celt_assert(st->postfilter_tapset <= 2);

celt_assert(st->postfilter_tapset >= 0);

celt_assert(st->postfilter_tapset_old <= 2);

celt_assert(st->postfilter_tapset_old >= 0);

}

#endif

int celt_decoder_get_size(int channels)

{

const CELTMode *mode = opus_custom_mode_create(48000, 960, NULL);

return opus_custom_decoder_get_size(mode, channels);

}

OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_get_size(const CELTMode *mode, int channels)

{

int size = sizeof(struct CELTDecoder)

+ (channels*(DECODE_BUFFER_SIZE+mode->overlap)-1)*sizeof(celt_sig)

+ channels*LPC_ORDER*sizeof(opus_val16)

+ 4*2*mode->nbEBands*sizeof(opus_val16);

return size;

}

#ifdef CUSTOM_MODES

CELTDecoder *opus_custom_decoder_create(const CELTMode *mode, int channels, int *error)

{

int ret;

CELTDecoder *st = (CELTDecoder *)opus_alloc(opus_custom_decoder_get_size(mode, channels));

ret = opus_custom_decoder_init(st, mode, channels);

if (ret != OPUS_OK)

{

opus_custom_decoder_destroy(st);

st = NULL;

}

if (error)

*error = ret;

return st;

}

#endif /* CUSTOM_MODES */

int celt_decoder_init(CELTDecoder *st, opus_int32 sampling_rate, int channels)

{

int ret;

ret = opus_custom_decoder_init(st, opus_custom_mode_create(48000, 960, NULL), channels);

if (ret != OPUS_OK)

return ret;

st->downsample = resampling_factor(sampling_rate);

if (st->downsample==0)

return OPUS_BAD_ARG;

else

return OPUS_OK;

}

OPUS_CUSTOM_NOSTATIC int opus_custom_decoder_init(CELTDecoder *st, const CELTMode *mode, int channels)

{

if (channels < 0 || channels > 2)

return OPUS_BAD_ARG;

if (st==NULL)

return OPUS_ALLOC_FAIL;

OPUS_CLEAR((char*)st, opus_custom_decoder_get_size(mode, channels));

st->mode = mode;

st->overlap = mode->overlap;

st->stream_channels = st->channels = channels;

st->downsample = 1;

st->start = 0;

st->end = st->mode->effEBands;

st->signalling = 1;

#ifndef DISABLE_UPDATE_DRAFT

st->disable_inv = channels == 1;

#else

st->disable_inv = 0;

#endif

st->arch = opus_select_arch();

opus_custom_decoder_ctl(st, OPUS_RESET_STATE);

return OPUS_OK;

}

#ifdef CUSTOM_MODES

void opus_custom_decoder_destroy(CELTDecoder *st)

{

opus_free(st);

}

#endif /* CUSTOM_MODES */

#ifndef CUSTOM_MODES

/* Special case for stereo with no downsampling and no accumulation. This is

quite common and we can make it faster by processing both channels in the

same loop, reducing overhead due to the dependency loop in the IIR filter. */

static void deemphasis_stereo_simple(celt_sig *in[], opus_val16 *pcm, int N, const opus_val16 coef0,

celt_sig *mem)

{

celt_sig * OPUS_RESTRICT x0;

celt_sig * OPUS_RESTRICT x1;

celt_sig m0, m1;

int j;

x0=in[0];

x1=in[1];

m0 = mem[0];

m1 = mem[1];

for (j=0;j<N;j++)

{

celt_sig tmp0, tmp1;

/* Add VERY_SMALL to x[] first to reduce dependency chain. */

tmp0 = x0[j] + VERY_SMALL + m0;

tmp1 = x1[j] + VERY_SMALL + m1;

m0 = MULT16_32_Q15(coef0, tmp0);

m1 = MULT16_32_Q15(coef0, tmp1);

pcm[2*j ] = SCALEOUT(SIG2WORD16(tmp0));

pcm[2*j+1] = SCALEOUT(SIG2WORD16(tmp1));

}

mem[0] = m0;

mem[1] = m1;

}

#endif

#ifndef RESYNTH

static

#endif

void deemphasis(celt_sig *in[], opus_val16 *pcm, int N, int C, int downsample, const opus_val16 *coef,

celt_sig *mem, int accum)

{

int c;

int Nd;

int apply_downsampling=0;

opus_val16 coef0;

VARDECL(celt_sig, scratch);

SAVE_STACK;

#ifndef CUSTOM_MODES

/* Short version for common case. */

if (downsample == 1 && C == 2 && !accum)

{

deemphasis_stereo_simple(in, pcm, N, coef[0], mem);

return;

}

#endif

#ifndef FIXED_POINT

(void)accum;

celt_assert(accum==0);

#endif

ALLOC(scratch, N, celt_sig);

coef0 = coef[0];

Nd = N/downsample;

c=0; do {

int j;

celt_sig * OPUS_RESTRICT x;

opus_val16 * OPUS_RESTRICT y;

celt_sig m = mem[c];

x =in[c];

y = pcm+c;

#ifdef CUSTOM_MODES

if (coef[1] != 0)

{

opus_val16 coef1 = coef[1];

opus_val16 coef3 = coef[3];

for (j=0;j<N;j++)

{

celt_sig tmp = x[j] + m + VERY_SMALL;

m = MULT16_32_Q15(coef0, tmp)

- MULT16_32_Q15(coef1, x[j]);

tmp = SHL32(MULT16_32_Q15(coef3, tmp), 2);

scratch[j] = tmp;

}

apply_downsampling=1;

} else

#endif

if (downsample>1)

{

/* Shortcut for the standard (non-custom modes) case */

for (j=0;j<N;j++)

{

celt_sig tmp = x[j] + VERY_SMALL + m;

m = MULT16_32_Q15(coef0, tmp);

scratch[j] = tmp;

}

apply_downsampling=1;

} else {

/* Shortcut for the standard (non-custom modes) case */

#ifdef FIXED_POINT

if (accum)

{

for (j=0;j<N;j++)

{

celt_sig tmp = x[j] + m + VERY_SMALL;

m = MULT16_32_Q15(coef0, tmp);

y[j*C] = SAT16(ADD32(y[j*C], SCALEOUT(SIG2WORD16(tmp))));

}

} else

#endif

{

for (j=0;j<N;j++)

{

celt_sig tmp = x[j] + VERY_SMALL + m;

m = MULT16_32_Q15(coef0, tmp);

y[j*C] = SCALEOUT(SIG2WORD16(tmp));

}

}

}

mem[c] = m;

if (apply_downsampling)

{

/* Perform down-sampling */

#ifdef FIXED_POINT

if (accum)

{

for (j=0;j<Nd;j++)

y[j*C] = SAT16(ADD32(y[j*C], SCALEOUT(SIG2WORD16(scratch[j*downsample]))));

} else

#endif

{

for (j=0;j<Nd;j++)

y[j*C] = SCALEOUT(SIG2WORD16(scratch[j*downsample]));

}

}

} while (++c<C);

RESTORE_STACK;

}

#ifndef RESYNTH

static

#endif

void celt_synthesis(const CELTMode *mode, celt_norm *X, celt_sig * out_syn[],

opus_val16 *oldBandE, int start, int effEnd, int C, int CC,

int isTransient, int LM, int downsample,

int silence, int arch)

{

int c, i;

int M;

int b;

int B;

int N, NB;

int shift;

int nbEBands;

int overlap;

VARDECL(celt_sig, freq);

SAVE_STACK;

overlap = mode->overlap;

nbEBands = mode->nbEBands;

N = mode->shortMdctSize<<LM;

ALLOC(freq, N, celt_sig); /**< Interleaved signal MDCTs */

M = 1<<LM;

if (isTransient)

{

B = M;

NB = mode->shortMdctSize;

shift = mode->maxLM;

} else {

B = 1;

NB = mode->shortMdctSize<<LM;

shift = mode->maxLM-LM;

}

if (CC==2&&C==1)

{

/* Copying a mono streams to two channels */

celt_sig *freq2;

denormalise_bands(mode, X, freq, oldBandE, start, effEnd, M,

downsample, silence);

/* Store a temporary copy in the output buffer because the IMDCT destroys its input. */

freq2 = out_syn[1]+overlap/2;

OPUS_COPY(freq2, freq, N);

for (b=0;b<B;b++)

clt_mdct_backward(&mode->mdct, &freq2[b], out_syn[0]+NB*b, mode->window, overlap, shift, B, arch);

for (b=0;b<B;b++)

clt_mdct_backward(&mode->mdct, &freq[b], out_syn[1]+NB*b, mode->window, overlap, shift, B, arch);

} else if (CC==1&&C==2)

{

/* Downmixing a stereo stream to mono */

celt_sig *freq2;

freq2 = out_syn[0]+overlap/2;

denormalise_bands(mode, X, freq, oldBandE, start, effEnd, M,

downsample, silence);

/* Use the output buffer as temp array before downmixing. */

denormalise_bands(mode, X+N, freq2, oldBandE+nbEBands, start, effEnd, M,

downsample, silence);

for (i=0;i<N;i++)

freq[i] = ADD32(HALF32(freq[i]), HALF32(freq2[i]));

for (b=0;b<B;b++)

clt_mdct_backward(&mode->mdct, &freq[b], out_syn[0]+NB*b, mode->window, overlap, shift, B, arch);

} else {

/* Normal case (mono or stereo) */

c=0; do {

denormalise_bands(mode, X+c*N, freq, oldBandE+c*nbEBands, start, effEnd, M,

downsample, silence);

for (b=0;b<B;b++)

clt_mdct_backward(&mode->mdct, &freq[b], out_syn[c]+NB*b, mode->window, overlap, shift, B, arch);

} while (++c<CC);

}

/* Saturate IMDCT output so that we can't overflow in the pitch postfilter

or in the */

c=0; do {

for (i=0;i<N;i++)

out_syn[c][i] = SATURATE(out_syn[c][i], SIG_SAT);

} while (++c<CC);

RESTORE_STACK;

}

static void tf_decode(int start, int end, int isTransient, int *tf_res, int LM, ec_dec *dec)

{

int i, curr, tf_select;

int tf_select_rsv;

int tf_changed;

int logp;

opus_uint32 budget;

opus_uint32 tell;

budget = dec->storage*8;

tell = ec_tell(dec);

logp = isTransient ? 2 : 4;

tf_select_rsv = LM>0 && tell+logp+1<=budget;

budget -= tf_select_rsv;

tf_changed = curr = 0;

for (i=start;i<end;i++)

{

if (tell+logp<=budget)

{

curr ^= ec_dec_bit_logp(dec, logp);

tell = ec_tell(dec);

tf_changed |= curr;

}

tf_res[i] = curr;

logp = isTransient ? 4 : 5;

}

tf_select = 0;

if (tf_select_rsv &&

tf_select_table[LM][4*isTransient+0+tf_changed] !=

tf_select_table[LM][4*isTransient+2+tf_changed])

{

tf_select = ec_dec_bit_logp(dec, 1);

}

for (i=start;i<end;i++)

{

tf_res[i] = tf_select_table[LM][4*isTransient+2*tf_select+tf_res[i]];

}

}

static int celt_plc_pitch_search(celt_sig *decode_mem[2], int C, int arch)

{

int pitch_index;

VARDECL( opus_val16, lp_pitch_buf );

SAVE_STACK;

ALLOC( lp_pitch_buf, DECODE_BUFFER_SIZE>>1, opus_val16 );

pitch_downsample(decode_mem, lp_pitch_buf,

DECODE_BUFFER_SIZE, C, arch);

pitch_search(lp_pitch_buf+(PLC_PITCH_LAG_MAX>>1), lp_pitch_buf,

DECODE_BUFFER_SIZE-PLC_PITCH_LAG_MAX,

PLC_PITCH_LAG_MAX-PLC_PITCH_LAG_MIN, &pitch_index, arch);

pitch_index = PLC_PITCH_LAG_MAX-pitch_index;

RESTORE_STACK;

return pitch_index;

}

static void celt_decode_lost(CELTDecoder * OPUS_RESTRICT st, int N, int LM)

{

int c;

int i;

const int C = st->channels;

celt_sig *decode_mem[2];

celt_sig *out_syn[2];

opus_val16 *lpc;

opus_val16 *oldBandE, *oldLogE, *oldLogE2, *backgroundLogE;

const OpusCustomMode *mode;

int nbEBands;

int overlap;

int start;

int loss_count;

int noise_based;

const opus_int16 *eBands;

SAVE_STACK;

mode = st->mode;

nbEBands = mode->nbEBands;

overlap = mode->overlap;

eBands = mode->eBands;

c=0; do {

decode_mem[c] = st->_decode_mem + c*(DECODE_BUFFER_SIZE+overlap);

out_syn[c] = decode_mem[c]+DECODE_BUFFER_SIZE-N;

} while (++c<C);

lpc = (opus_val16*)(st->_decode_mem+(DECODE_BUFFER_SIZE+overlap)*C);

oldBandE = lpc+C*LPC_ORDER;

oldLogE = oldBandE + 2*nbEBands;

oldLogE2 = oldLogE + 2*nbEBands;

backgroundLogE = oldLogE2 + 2*nbEBands;

loss_count = st->loss_count;

start = st->start;

noise_based = loss_count >= 5 || start != 0 || st->skip_plc;

if (noise_based)

{

/* Noise-based PLC/CNG */

#ifdef NORM_ALIASING_HACK

celt_norm *X;

#else

VARDECL(celt_norm, X);

#endif

opus_uint32 seed;

int end;

int effEnd;

opus_val16 decay;

end = st->end;

effEnd = IMAX(start, IMIN(end, mode->effEBands));

#ifdef NORM_ALIASING_HACK

/* This is an ugly hack that breaks aliasing rules and would be easily broken,

but it saves almost 4kB of stack. */

X = (celt_norm*)(out_syn[C-1]+overlap/2);

#else

ALLOC(X, C*N, celt_norm); /**< Interleaved normalised MDCTs */

#endif

/* Energy decay */

decay = loss_count==0 ? QCONST16(1.5f, DB_SHIFT) : QCONST16(.5f, DB_SHIFT);

c=0; do

{

for (i=start;i<end;i++)

oldBandE[c*nbEBands+i] = MAX16(backgroundLogE[c*nbEBands+i], oldBandE[c*nbEBands+i] - decay);

} while (++c<C);

seed = st->rng;

for (c=0;c<C;c++)

{

for (i=start;i<effEnd;i++)

{

int j;

int boffs;

int blen;

boffs = N*c+(eBands[i]<<LM);

blen = (eBands[i+1]-eBands[i])<<LM;

for (j=0;j<blen;j++)

{

seed = celt_lcg_rand(seed);

X[boffs+j] = (celt_norm)((opus_int32)seed>>20);

}

renormalise_vector(X+boffs, blen, Q15ONE, st->arch);

}

}

st->rng = seed;

c=0; do {

OPUS_MOVE(decode_mem[c], decode_mem[c]+N,

DECODE_BUFFER_SIZE-N+(overlap>>1));

} while (++c<C);

celt_synthesis(mode, X, out_syn, oldBandE, start, effEnd, C, C, 0, LM, st->downsample, 0, st->arch);

} else {

int exc_length;

/* Pitch-based PLC */

const opus_val16 *window;

opus_val16 *exc;

opus_val16 fade = Q15ONE;

int pitch_index;

VARDECL(opus_val32, etmp);

VARDECL(opus_val16, _exc);

VARDECL(opus_val16, fir_tmp);

if (loss_count == 0)

{

st->last_pitch_index = pitch_index = celt_plc_pitch_search(decode_mem, C, st->arch);

} else {

pitch_index = st->last_pitch_index;

fade = QCONST16(.8f,15);

}

/* We want the excitation for 2 pitch periods in order to look for a

decaying signal, but we can't get more than MAX_PERIOD. */

exc_length = IMIN(2*pitch_index, MAX_PERIOD);

ALLOC(etmp, overlap, opus_val32);

ALLOC(_exc, MAX_PERIOD+LPC_ORDER, opus_val16);

ALLOC(fir_tmp, exc_length, opus_val16);

exc = _exc+LPC_ORDER;

window = mode->window;

c=0; do {

opus_val16 decay;

opus_val16 attenuation;

opus_val32 S1=0;

celt_sig *buf;

int extrapolation_offset;

int extrapolation_len;

int j;

buf = decode_mem[c];

for (i=0;i<MAX_PERIOD+LPC_ORDER;i++)

exc[i-LPC_ORDER] = ROUND16(buf[DECODE_BUFFER_SIZE-MAX_PERIOD-LPC_ORDER+i], SIG_SHIFT);

if (loss_count == 0)

{

opus_val32 ac[LPC_ORDER+1];

/* Compute LPC coefficients for the last MAX_PERIOD samples before

the first loss so we can work in the excitation-filter domain. */

_celt_autocorr(exc, ac, window, overlap,

LPC_ORDER, MAX_PERIOD, st->arch);

/* Add a noise floor of -40 dB. */

#ifdef FIXED_POINT

ac[0] += SHR32(ac[0],13);

#else

ac[0] *= 1.0001f;

#endif

/* Use lag windowing to stabilize the Levinson-Durbin recursion. */

for (i=1;i<=LPC_ORDER;i++)

{

/*ac[i] *= exp(-.5*(2*M_PI*.002*i)*(2*M_PI*.002*i));*/

#ifdef FIXED_POINT

ac[i] -= MULT16_32_Q15(2*i*i, ac[i]);

#else

ac[i] -= ac[i]*(0.008f*0.008f)*i*i;

#endif

}

_celt_lpc(lpc+c*LPC_ORDER, ac, LPC_ORDER);

#ifdef FIXED_POINT

/* For fixed-point, apply bandwidth expansion until we can guarantee that

no overflow can happen in the IIR filter. This means:

32768*sum(abs(filter)) < 2^31 */

while (1) {

opus_val16 tmp=Q15ONE;

opus_val32 sum=QCONST16(1., SIG_SHIFT);

for (i=0;i<LPC_ORDER;i++)

sum += ABS16(lpc[c*LPC_ORDER+i]);

if (sum < 65535) break;

for (i=0;i<LPC_ORDER;i++)

{

tmp = MULT16_16_Q15(QCONST16(.99f,15), tmp);

lpc[c*LPC_ORDER+i] = MULT16_16_Q15(lpc[c*LPC_ORDER+i], tmp);

}

}

#endif

}

/* Initialize the LPC history with the samples just before the start

of the region for which we're computing the excitation. */

{

/* Compute the excitation for exc_length samples before the loss. We need the copy

because celt_fir() cannot filter in-place. */

celt_fir(exc+MAX_PERIOD-exc_length, lpc+c*LPC_ORDER,

fir_tmp, exc_length, LPC_ORDER, st->arch);

OPUS_COPY(exc+MAX_PERIOD-exc_length, fir_tmp, exc_length);

}

/* Check if the waveform is decaying, and if so how fast.

We do this to avoid adding energy when concealing in a segment

with decaying energy. */

{

opus_val32 E1=1, E2=1;

int decay_length;

#ifdef FIXED_POINT

int shift = IMAX(0,2*celt_zlog2(celt_maxabs16(&exc[MAX_PERIOD-exc_length], exc_length))-20);

#endif

decay_length = exc_length>>1;

for (i=0;i<decay_length;i++)

{

opus_val16 e;

e = exc[MAX_PERIOD-decay_length+i];

E1 += SHR32(MULT16_16(e, e), shift);

e = exc[MAX_PERIOD-2*decay_length+i];

E2 += SHR32(MULT16_16(e, e), shift);

}

E1 = MIN32(E1, E2);

decay = celt_sqrt(frac_div32(SHR32(E1, 1), E2));

}

/* Move the decoder memory one frame to the left to give us room to

add the data for the new frame. We ignore the overlap that extends

past the end of the buffer, because we aren't going to use it. */

OPUS_MOVE(buf, buf+N, DECODE_BUFFER_SIZE-N);

/* Extrapolate from the end of the excitation with a period of

"pitch_index", scaling down each period by an additional factor of

"decay". */

extrapolation_offset = MAX_PERIOD-pitch_index;

/* We need to extrapolate enough samples to cover a complete MDCT

window (including overlap/2 samples on both sides). */

extrapolation_len = N+overlap;

/* We also apply fading if this is not the first loss. */

attenuation = MULT16_16_Q15(fade, decay);

for (i=j=0;i<extrapolation_len;i++,j++)

{

opus_val16 tmp;

if (j >= pitch_index) {

j -= pitch_index;

attenuation = MULT16_16_Q15(attenuation, decay);

}

buf[DECODE_BUFFER_SIZE-N+i] =

SHL32(EXTEND32(MULT16_16_Q15(attenuation,

exc[extrapolation_offset+j])), SIG_SHIFT);

/* Compute the energy of the previously decoded signal whose

excitation we're copying. */

tmp = ROUND16(

buf[DECODE_BUFFER_SIZE-MAX_PERIOD-N+extrapolation_offset+j],

SIG_SHIFT);

S1 += SHR32(MULT16_16(tmp, tmp), 10);

}

{

opus_val16 lpc_mem[LPC_ORDER];

/* Copy the last decoded samples (prior to the overlap region) to

synthesis filter memory so we can have a continuous signal. */

for (i=0;i<LPC_ORDER;i++)

lpc_mem[i] = ROUND16(buf[DECODE_BUFFER_SIZE-N-1-i], SIG_SHIFT);

/* Apply the synthesis filter to convert the excitation back into

the signal domain. */

celt_iir(buf+DECODE_BUFFER_SIZE-N, lpc+c*LPC_ORDER,

buf+DECODE_BUFFER_SIZE-N, extrapolation_len, LPC_ORDER,

lpc_mem, st->arch);

#ifdef FIXED_POINT

for (i=0; i < extrapolation_len; i++)

buf[DECODE_BUFFER_SIZE-N+i] = SATURATE(buf[DECODE_BUFFER_SIZE-N+i], SIG_SAT);

#endif

}

/* Check if the synthesis energy is higher than expected, which can

happen with the signal changes during our window. If so,

attenuate. */

{

opus_val32 S2=0;

for (i=0;i<extrapolation_len;i++)

{

opus_val16 tmp = ROUND16(buf[DECODE_BUFFER_SIZE-N+i], SIG_SHIFT);

S2 += SHR32(MULT16_16(tmp, tmp), 10);

}

/* This checks for an "explosion" in the synthesis. */

#ifdef FIXED_POINT

if (!(S1 > SHR32(S2,2)))

#else

/* The float test is written this way to catch NaNs in the output

of the IIR filter at the same time. */

if (!(S1 > 0.2f*S2))

#endif

{

for (i=0;i<extrapolation_len;i++)

buf[DECODE_BUFFER_SIZE-N+i] = 0;

} else if (S1 < S2)

{

opus_val16 ratio = celt_sqrt(frac_div32(SHR32(S1,1)+1,S2+1));

for (i=0;i<overlap;i++)

{

opus_val16 tmp_g = Q15ONE

- MULT16_16_Q15(window[i], Q15ONE-ratio);

buf[DECODE_BUFFER_SIZE-N+i] =

MULT16_32_Q15(tmp_g, buf[DECODE_BUFFER_SIZE-N+i]);

}

for (i=overlap;i<extrapolation_len;i++)

{

buf[DECODE_BUFFER_SIZE-N+i] =

MULT16_32_Q15(ratio, buf[DECODE_BUFFER_SIZE-N+i]);

}

}

}

/* Apply the pre-filter to the MDCT overlap for the next frame because

the post-filter will be re-applied in the decoder after the MDCT

overlap. */

comb_filter(etmp, buf+DECODE_BUFFER_SIZE,

st->postfilter_period, st->postfilter_period, overlap,

-st->postfilter_gain, -st->postfilter_gain,

st->postfilter_tapset, st->postfilter_tapset, NULL, 0, st->arch);

/* Simulate TDAC on the concealed audio so that it blends with the

MDCT of the next frame. */

for (i=0;i<overlap/2;i++)

{

buf[DECODE_BUFFER_SIZE+i] =

MULT16_32_Q15(window[i], etmp[overlap-1-i])

+ MULT16_32_Q15(window[overlap-i-1], etmp[i]);

}

} while (++c<C);

}

st->loss_count = loss_count+1;

RESTORE_STACK;

}

int celt_decode_with_ec(CELTDecoder * OPUS_RESTRICT st, const unsigned char *data,

int len, opus_val16 * OPUS_RESTRICT pcm, int frame_size, ec_dec *dec, int accum)

{

int c, i, N;

int spread_decision;

opus_int32 bits;

ec_dec _dec;

#ifdef NORM_ALIASING_HACK

celt_norm *X;

#else

VARDECL(celt_norm, X);

#endif

VARDECL(int, fine_quant);

VARDECL(int, pulses);

VARDECL(int, cap);

VARDECL(int, offsets);

VARDECL(int, fine_priority);

VARDECL(int, tf_res);

VARDECL(unsigned char, collapse_masks);

celt_sig *decode_mem[2];

celt_sig *out_syn[2];

opus_val16 *lpc;

opus_val16 *oldBandE, *oldLogE, *oldLogE2, *backgroundLogE;

int shortBlocks;

int isTransient;

int intra_ener;

const int CC = st->channels;

int LM, M;

int start;

int end;

int effEnd;

int codedBands;

int alloc_trim;

int postfilter_pitch;

opus_val16 postfilter_gain;

int intensity=0;

int dual_stereo=0;

opus_int32 total_bits;

opus_int32 balance;

opus_int32 tell;

int dynalloc_logp;

int postfilter_tapset;

int anti_collapse_rsv;

int anti_collapse_on=0;

int silence;

int C = st->stream_channels;

const OpusCustomMode *mode;

int nbEBands;

int overlap;

const opus_int16 *eBands;

ALLOC_STACK;

VALIDATE_CELT_DECODER(st);

mode = st->mode;

nbEBands = mode->nbEBands;

overlap = mode->overlap;

eBands = mode->eBands;

start = st->start;

end = st->end;

frame_size *= st->downsample;

lpc = (opus_val16*)(st->_decode_mem+(DECODE_BUFFER_SIZE+overlap)*CC);

oldBandE = lpc+CC*LPC_ORDER;

oldLogE = oldBandE + 2*nbEBands;

oldLogE2 = oldLogE + 2*nbEBands;

backgroundLogE = oldLogE2 + 2*nbEBands;

#ifdef CUSTOM_MODES

if (st->signalling && data!=NULL)

{

int data0=data[0];

/* Convert "standard mode" to Opus header */

if (mode->Fs==48000 && mode->shortMdctSize==120)

{

data0 = fromOpus(data0);

if (data0<0)

return OPUS_INVALID_PACKET;

}

st->end = end = IMAX(1, mode->effEBands-2*(data0>>5));

LM = (data0>>3)&0x3;

C = 1 + ((data0>>2)&0x1);

data++;

len--;

if (LM>mode->maxLM)

return OPUS_INVALID_PACKET;

if (frame_size < mode->shortMdctSize<<LM)

return OPUS_BUFFER_TOO_SMALL;

else

frame_size = mode->shortMdctSize<<LM;

} else {

#else

{

#endif

for (LM=0;LM<=mode->maxLM;LM++)

if (mode->shortMdctSize<<LM==frame_size)

break;

if (LM>mode->maxLM)

return OPUS_BAD_ARG;

}

M=1<<LM;

if (len<0 || len>1275 || pcm==NULL)

return OPUS_BAD_ARG;

N = M*mode->shortMdctSize;

c=0; do {

decode_mem[c] = st->_decode_mem + c*(DECODE_BUFFER_SIZE+overlap);

out_syn[c] = decode_mem[c]+DECODE_BUFFER_SIZE-N;

} while (++c<CC);

effEnd = end;

if (effEnd > mode->effEBands)

effEnd = mode->effEBands;

if (data == NULL || len<=1)

{

celt_decode_lost(st, N, LM);

deemphasis(out_syn, pcm, N, CC, st->downsample, mode->preemph, st->preemph_memD, accum);

RESTORE_STACK;

return frame_size/st->downsample;

}

/* Check if there are at least two packets received consecutively before

* turning on the pitch-based PLC */

st->skip_plc = st->loss_count != 0;

if (dec == NULL)

{

ec_dec_init(&_dec,(unsigned char*)data,len);

dec = &_dec;

}

if (C==1)

{

for (i=0;i<nbEBands;i++)

oldBandE[i]=MAX16(oldBandE[i],oldBandE[nbEBands+i]);

}

total_bits = len*8;

tell = ec_tell(dec);

if (tell >= total_bits)

silence = 1;

else if (tell==1)

silence = ec_dec_bit_logp(dec, 15);

else

silence = 0;

if (silence)

{

/* Pretend we've read all the remaining bits */

tell = len*8;

dec->nbits_total+=tell-ec_tell(dec);

}

postfilter_gain = 0;

postfilter_pitch = 0;

postfilter_tapset = 0;

if (start==0 && tell+16 <= total_bits)

{

if(ec_dec_bit_logp(dec, 1))

{

int qg, octave;

octave = ec_dec_uint(dec, 6);

postfilter_pitch = (16<<octave)+ec_dec_bits(dec, 4+octave)-1;

qg = ec_dec_bits(dec, 3);

if (ec_tell(dec)+2<=total_bits)

postfilter_tapset = ec_dec_icdf(dec, tapset_icdf, 2);

postfilter_gain = QCONST16(.09375f,15)*(qg+1);

}

tell = ec_tell(dec);

}

if (LM > 0 && tell+3 <= total_bits)

{

isTransient = ec_dec_bit_logp(dec, 3);

tell = ec_tell(dec);

}

else

isTransient = 0;

if (isTransient)

shortBlocks = M;

else

shortBlocks = 0;

/* Decode the global flags (first symbols in the stream) */