Google Git
Sign in
nest-open-source / nest-cam / 4320010 / av / 5121e70f983699f03625822f4e53d0759305313b / . / av / media / libstagefright / codecs / amrwbenc / src / voAMRWBEnc.c
blob: b908ff8b537b4c1863613e2c87cf5d1fae895e44 [file] [log] [blame]
/*
** Copyright 2003-2010, VisualOn, Inc.
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
/***********************************************************************
* File: voAMRWBEnc.c *
* *
* Description: Performs the main encoder routine *
* Fixed-point C simulation of AMR WB ACELP coding *
* algorithm with 20 msspeech frames for *
* wideband speech signals. *
* *
************************************************************************/
#include <stdio.h>
#include <stdlib.h>
#include "typedef.h"
#include "basic_op.h"
#include "oper_32b.h"
#include "math_op.h"
#include "cnst.h"
#include "acelp.h"
#include "cod_main.h"
#include "bits.h"
#include "main.h"
#include "voAMRWB.h"
#include "mem_align.h"
#include "cmnMemory.h"
#define UNUSED(x) (void)(x)
#ifdef __cplusplus
extern "C" {
#endif
/* LPC interpolation coef {0.45, 0.8, 0.96, 1.0}; in Q15 */
static Word16 interpol_frac[NB_SUBFR] = {14746, 26214, 31457, 32767};
/* isp tables for initialization */
static Word16 isp_init[M] =
{
32138, 30274, 27246, 23170, 18205, 12540, 6393, 0,
-6393, -12540, -18205, -23170, -27246, -30274, -32138, 1475
};
static Word16 isf_init[M] =
{
1024, 2048, 3072, 4096, 5120, 6144, 7168, 8192,
9216, 10240, 11264, 12288, 13312, 14336, 15360, 3840
};
/* High Band encoding */
static const Word16 HP_gain[16] =
{
3624, 4673, 5597, 6479, 7425, 8378, 9324, 10264,
11210, 12206, 13391, 14844, 16770, 19655, 24289, 32728
};
/* Private function declaration */
static Word16 synthesis(
Word16 Aq[], /* A(z) : quantized Az */
Word16 exc[], /* (i) : excitation at 12kHz */
Word16 Q_new, /* (i) : scaling performed on exc */
Word16 synth16k[], /* (o) : 16kHz synthesis signal */
Coder_State * st /* (i/o) : State structure */
);
/* Codec some parameters initialization */
void Reset_encoder(void *st, Word16 reset_all)
{
Word16 i;
Coder_State *cod_state;
cod_state = (Coder_State *) st;
Set_zero(cod_state->old_exc, PIT_MAX + L_INTERPOL);
Set_zero(cod_state->mem_syn, M);
Set_zero(cod_state->past_isfq, M);
cod_state->mem_w0 = 0;
cod_state->tilt_code = 0;
cod_state->first_frame = 1;
Init_gp_clip(cod_state->gp_clip);
cod_state->L_gc_thres = 0;
if (reset_all != 0)
{
/* Static vectors to zero */
Set_zero(cod_state->old_speech, L_TOTAL - L_FRAME);
Set_zero(cod_state->old_wsp, (PIT_MAX / OPL_DECIM));
Set_zero(cod_state->mem_decim2, 3);
/* routines initialization */
Init_Decim_12k8(cod_state->mem_decim);
Init_HP50_12k8(cod_state->mem_sig_in);
Init_Levinson(cod_state->mem_levinson);
Init_Q_gain2(cod_state->qua_gain);
Init_Hp_wsp(cod_state->hp_wsp_mem);
/* isp initialization */
Copy(isp_init, cod_state->ispold, M);
Copy(isp_init, cod_state->ispold_q, M);
/* variable initialization */
cod_state->mem_preemph = 0;
cod_state->mem_wsp = 0;
cod_state->Q_old = 15;
cod_state->Q_max[0] = 15;
cod_state->Q_max[1] = 15;
cod_state->old_wsp_max = 0;
cod_state->old_wsp_shift = 0;
/* pitch ol initialization */
cod_state->old_T0_med = 40;
cod_state->ol_gain = 0;
cod_state->ada_w = 0;
cod_state->ol_wght_flg = 0;
for (i = 0; i < 5; i++)
{
cod_state->old_ol_lag[i] = 40;
}
Set_zero(cod_state->old_hp_wsp, (L_FRAME / 2) / OPL_DECIM + (PIT_MAX / OPL_DECIM));
Set_zero(cod_state->mem_syn_hf, M);
Set_zero(cod_state->mem_syn_hi, M);
Set_zero(cod_state->mem_syn_lo, M);
Init_HP50_12k8(cod_state->mem_sig_out);
Init_Filt_6k_7k(cod_state->mem_hf);
Init_HP400_12k8(cod_state->mem_hp400);
Copy(isf_init, cod_state->isfold, M);
cod_state->mem_deemph = 0;
cod_state->seed2 = 21845;
Init_Filt_6k_7k(cod_state->mem_hf2);
cod_state->gain_alpha = 32767;
cod_state->vad_hist = 0;
wb_vad_reset(cod_state->vadSt);
dtx_enc_reset(cod_state->dtx_encSt, isf_init);
}
return;
}
/*-----------------------------------------------------------------*
* Funtion coder *
* ~~~~~ *
* ->Main coder routine. *
* *
*-----------------------------------------------------------------*/
void coder(
Word16 * mode, /* input : used mode */
Word16 speech16k[], /* input : 320 new speech samples (at 16 kHz) */
Word16 prms[], /* output: output parameters */
Word16 * ser_size, /* output: bit rate of the used mode */
void *spe_state, /* i/o : State structure */
Word16 allow_dtx /* input : DTX ON/OFF */
)
{
/* Coder states */
Coder_State *st;
/* Speech vector */
Word16 old_speech[L_TOTAL];
Word16 *new_speech, *speech, *p_window;
/* Weighted speech vector */
Word16 old_wsp[L_FRAME + (PIT_MAX / OPL_DECIM)];
Word16 *wsp;
/* Excitation vector */
Word16 old_exc[(L_FRAME + 1) + PIT_MAX + L_INTERPOL];
Word16 *exc;
/* LPC coefficients */
Word16 r_h[M + 1], r_l[M + 1]; /* Autocorrelations of windowed speech */
Word16 rc[M]; /* Reflection coefficients. */
Word16 Ap[M + 1]; /* A(z) with spectral expansion */
Word16 ispnew[M]; /* immittance spectral pairs at 4nd sfr */
Word16 ispnew_q[M]; /* quantized ISPs at 4nd subframe */
Word16 isf[M]; /* ISF (frequency domain) at 4nd sfr */
Word16 *p_A, *p_Aq; /* ptr to A(z) for the 4 subframes */
Word16 A[NB_SUBFR * (M + 1)]; /* A(z) unquantized for the 4 subframes */
Word16 Aq[NB_SUBFR * (M + 1)]; /* A(z) quantized for the 4 subframes */
/* Other vectors */
Word16 xn[L_SUBFR]; /* Target vector for pitch search */
Word16 xn2[L_SUBFR]; /* Target vector for codebook search */
Word16 dn[L_SUBFR]; /* Correlation between xn2 and h1 */
Word16 cn[L_SUBFR]; /* Target vector in residual domain */
Word16 h1[L_SUBFR]; /* Impulse response vector */
Word16 h2[L_SUBFR]; /* Impulse response vector */
Word16 code[L_SUBFR]; /* Fixed codebook excitation */
Word16 y1[L_SUBFR]; /* Filtered adaptive excitation */
Word16 y2[L_SUBFR]; /* Filtered adaptive excitation */
Word16 error[M + L_SUBFR]; /* error of quantization */
Word16 synth[L_SUBFR]; /* 12.8kHz synthesis vector */
Word16 exc2[L_FRAME]; /* excitation vector */
Word16 buf[L_FRAME]; /* VAD buffer */
/* Scalars */
Word32 i, j, i_subfr, select, pit_flag, clip_gain, vad_flag;
Word16 codec_mode;
Word16 T_op, T_op2, T0, T0_min, T0_max, T0_frac, index;
Word16 gain_pit, gain_code, g_coeff[4], g_coeff2[4];
Word16 tmp, gain1, gain2, exp, Q_new, mu, shift, max;
Word16 voice_fac;
Word16 indice[8];
Word32 L_tmp, L_gain_code, L_max, L_tmp1;
Word16 code2[L_SUBFR]; /* Fixed codebook excitation */
Word16 stab_fac, fac, gain_code_lo;
Word16 corr_gain;
Word16 *vo_p0, *vo_p1, *vo_p2, *vo_p3;
st = (Coder_State *) spe_state;
*ser_size = nb_of_bits[*mode];
codec_mode = *mode;
/*--------------------------------------------------------------------------*
* Initialize pointers to speech vector. *
* *
* *
* |-------|-------|-------|-------|-------|-------| *
* past sp sf1 sf2 sf3 sf4 L_NEXT *
* <------- Total speech buffer (L_TOTAL) ------> *
* old_speech *
* <------- LPC analysis window (L_WINDOW) ------> *
* | <-- present frame (L_FRAME) ----> *
* p_window | <----- new speech (L_FRAME) ----> *
* | | *
* speech | *
* new_speech *
*--------------------------------------------------------------------------*/
new_speech = old_speech + L_TOTAL - L_FRAME - L_FILT; /* New speech */
speech = old_speech + L_TOTAL - L_FRAME - L_NEXT; /* Present frame */
p_window = old_speech + L_TOTAL - L_WINDOW;
exc = old_exc + PIT_MAX + L_INTERPOL;
wsp = old_wsp + (PIT_MAX / OPL_DECIM);
/* copy coder memory state into working space */
Copy(st->old_speech, old_speech, L_TOTAL - L_FRAME);
Copy(st->old_wsp, old_wsp, PIT_MAX / OPL_DECIM);
Copy(st->old_exc, old_exc, PIT_MAX + L_INTERPOL);
/*---------------------------------------------------------------*
* Down sampling signal from 16kHz to 12.8kHz *
* -> The signal is extended by L_FILT samples (padded to zero) *
* to avoid additional delay (L_FILT samples) in the coder. *
* The last L_FILT samples are approximated after decimation and *
* are used (and windowed) only in autocorrelations. *
*---------------------------------------------------------------*/
Decim_12k8(speech16k, L_FRAME16k, new_speech, st->mem_decim);
/* last L_FILT samples for autocorrelation window */
Copy(st->mem_decim, code, 2 * L_FILT16k);
Set_zero(error, L_FILT16k); /* set next sample to zero */
Decim_12k8(error, L_FILT16k, new_speech + L_FRAME, code);
/*---------------------------------------------------------------*
* Perform 50Hz HP filtering of input signal. *
*---------------------------------------------------------------*/
HP50_12k8(new_speech, L_FRAME, st->mem_sig_in);
/* last L_FILT samples for autocorrelation window */
Copy(st->mem_sig_in, code, 6);
HP50_12k8(new_speech + L_FRAME, L_FILT, code);
/*---------------------------------------------------------------*
* Perform fixed preemphasis through 1 - g z^-1 *
* Scale signal to get maximum of precision in filtering *
*---------------------------------------------------------------*/
mu = PREEMPH_FAC >> 1; /* Q15 --> Q14 */
/* get max of new preemphased samples (L_FRAME+L_FILT) */
L_tmp = new_speech[0] << 15;
L_tmp -= (st->mem_preemph * mu)<<1;
L_max = L_abs(L_tmp);
for (i = 1; i < L_FRAME + L_FILT; i++)
{
L_tmp = new_speech[i] << 15;
L_tmp -= (new_speech[i - 1] * mu)<<1;
L_tmp = L_abs(L_tmp);
if(L_tmp > L_max)
{
L_max = L_tmp;
}
}
/* get scaling factor for new and previous samples */
/* limit scaling to Q_MAX to keep dynamic for ringing in low signal */
/* limit scaling to Q_MAX also to avoid a[0]<1 in syn_filt_32 */
tmp = extract_h(L_max);
if (tmp == 0)
{
shift = Q_MAX;
} else
{
shift = norm_s(tmp) - 1;
if (shift < 0)
{
shift = 0;
}
if (shift > Q_MAX)
{
shift = Q_MAX;
}
}
Q_new = shift;
if (Q_new > st->Q_max[0])
{
Q_new = st->Q_max[0];
}
if (Q_new > st->Q_max[1])
{
Q_new = st->Q_max[1];
}
exp = (Q_new - st->Q_old);
st->Q_old = Q_new;
st->Q_max[1] = st->Q_max[0];
st->Q_max[0] = shift;
/* preemphasis with scaling (L_FRAME+L_FILT) */
tmp = new_speech[L_FRAME - 1];
for (i = L_FRAME + L_FILT - 1; i > 0; i--)
{
L_tmp = new_speech[i] << 15;
L_tmp -= (new_speech[i - 1] * mu)<<1;
L_tmp = (L_tmp << Q_new);
new_speech[i] = vo_round(L_tmp);
}
L_tmp = new_speech[0] << 15;
L_tmp -= (st->mem_preemph * mu)<<1;
L_tmp = (L_tmp << Q_new);
new_speech[0] = vo_round(L_tmp);
st->mem_preemph = tmp;
/* scale previous samples and memory */
Scale_sig(old_speech, L_TOTAL - L_FRAME - L_FILT, exp);
Scale_sig(old_exc, PIT_MAX + L_INTERPOL, exp);
Scale_sig(st->mem_syn, M, exp);
Scale_sig(st->mem_decim2, 3, exp);
Scale_sig(&(st->mem_wsp), 1, exp);
Scale_sig(&(st->mem_w0), 1, exp);
/*------------------------------------------------------------------------*
* Call VAD *
* Preemphesis scale down signal in low frequency and keep dynamic in HF.*
* Vad work slightly in futur (new_speech = speech + L_NEXT - L_FILT). *
*------------------------------------------------------------------------*/
Copy(new_speech, buf, L_FRAME);
#ifdef ASM_OPT /* asm optimization branch */
Scale_sig_opt(buf, L_FRAME, 1 - Q_new);
#else
Scale_sig(buf, L_FRAME, 1 - Q_new);
#endif
vad_flag = wb_vad(st->vadSt, buf); /* Voice Activity Detection */
if (vad_flag == 0)
{
st->vad_hist = (st->vad_hist + 1);
} else
{
st->vad_hist = 0;
}
/* DTX processing */
if (allow_dtx != 0)
{
/* Note that mode may change here */
tx_dtx_handler(st->dtx_encSt, vad_flag, mode);
*ser_size = nb_of_bits[*mode];
}
if(*mode != MRDTX)
{
Parm_serial(vad_flag, 1, &prms);
}
/*------------------------------------------------------------------------*
* Perform LPC analysis *
* ~~~~~~~~~~~~~~~~~~~~ *
* - autocorrelation + lag windowing *
* - Levinson-durbin algorithm to find a[] *
* - convert a[] to isp[] *
* - convert isp[] to isf[] for quantization *
* - quantize and code the isf[] *
* - convert isf[] to isp[] for interpolation *
* - find the interpolated ISPs and convert to a[] for the 4 subframes *
*------------------------------------------------------------------------*/
/* LP analysis centered at 4nd subframe */
Autocorr(p_window, M, r_h, r_l); /* Autocorrelations */
Lag_window(r_h, r_l); /* Lag windowing */
Levinson(r_h, r_l, A, rc, st->mem_levinson); /* Levinson Durbin */
Az_isp(A, ispnew, st->ispold); /* From A(z) to ISP */
/* Find the interpolated ISPs and convert to a[] for all subframes */
Int_isp(st->ispold, ispnew, interpol_frac, A);
/* update ispold[] for the next frame */
Copy(ispnew, st->ispold, M);
/* Convert ISPs to frequency domain 0..6400 */
Isp_isf(ispnew, isf, M);
/* check resonance for pitch clipping algorithm */
Gp_clip_test_isf(isf, st->gp_clip);
/*----------------------------------------------------------------------*
* Perform PITCH_OL analysis *
* ~~~~~~~~~~~~~~~~~~~~~~~~~ *
* - Find the residual res[] for the whole speech frame *
* - Find the weighted input speech wsp[] for the whole speech frame *
* - scale wsp[] to avoid overflow in pitch estimation *
* - Find open loop pitch lag for whole speech frame *
*----------------------------------------------------------------------*/
p_A = A;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
/* Weighting of LPC coefficients */
Weight_a(p_A, Ap, GAMMA1, M);
#ifdef ASM_OPT /* asm optimization branch */
Residu_opt(Ap, &speech[i_subfr], &wsp[i_subfr], L_SUBFR);
#else
Residu(Ap, &speech[i_subfr], &wsp[i_subfr], L_SUBFR);
#endif
p_A += (M + 1);
}
Deemph2(wsp, TILT_FAC, L_FRAME, &(st->mem_wsp));
/* find maximum value on wsp[] for 12 bits scaling */
max = 0;
for (i = 0; i < L_FRAME; i++)
{
tmp = abs_s(wsp[i]);
if(tmp > max)
{
max = tmp;
}
}
tmp = st->old_wsp_max;
if(max > tmp)
{
tmp = max; /* tmp = max(wsp_max, old_wsp_max) */
}
st->old_wsp_max = max;
shift = norm_s(tmp) - 3;
if (shift > 0)
{
shift = 0; /* shift = 0..-3 */
}
/* decimation of wsp[] to search pitch in LF and to reduce complexity */
LP_Decim2(wsp, L_FRAME, st->mem_decim2);
/* scale wsp[] in 12 bits to avoid overflow */
#ifdef ASM_OPT /* asm optimization branch */
Scale_sig_opt(wsp, L_FRAME / OPL_DECIM, shift);
#else
Scale_sig(wsp, L_FRAME / OPL_DECIM, shift);
#endif
/* scale old_wsp (warning: exp must be Q_new-Q_old) */
exp = exp + (shift - st->old_wsp_shift);
st->old_wsp_shift = shift;
Scale_sig(old_wsp, PIT_MAX / OPL_DECIM, exp);
Scale_sig(st->old_hp_wsp, PIT_MAX / OPL_DECIM, exp);
scale_mem_Hp_wsp(st->hp_wsp_mem, exp);
/* Find open loop pitch lag for whole speech frame */
if(*ser_size == NBBITS_7k)
{
/* Find open loop pitch lag for whole speech frame */
T_op = Pitch_med_ol(wsp, st, L_FRAME / OPL_DECIM);
} else
{
/* Find open loop pitch lag for first 1/2 frame */
T_op = Pitch_med_ol(wsp, st, (L_FRAME/2) / OPL_DECIM);
}
if(st->ol_gain > 19661) /* 0.6 in Q15 */
{
st->old_T0_med = Med_olag(T_op, st->old_ol_lag);
st->ada_w = 32767;
} else
{
st->ada_w = vo_mult(st->ada_w, 29491);
}
if(st->ada_w < 26214)
st->ol_wght_flg = 0;
else
st->ol_wght_flg = 1;
wb_vad_tone_detection(st->vadSt, st->ol_gain);
T_op *= OPL_DECIM;
if(*ser_size != NBBITS_7k)
{
/* Find open loop pitch lag for second 1/2 frame */
T_op2 = Pitch_med_ol(wsp + ((L_FRAME / 2) / OPL_DECIM), st, (L_FRAME/2) / OPL_DECIM);
if(st->ol_gain > 19661) /* 0.6 in Q15 */
{
st->old_T0_med = Med_olag(T_op2, st->old_ol_lag);
st->ada_w = 32767;
} else
{
st->ada_w = mult(st->ada_w, 29491);
}
if(st->ada_w < 26214)
st->ol_wght_flg = 0;
else
st->ol_wght_flg = 1;
wb_vad_tone_detection(st->vadSt, st->ol_gain);
T_op2 *= OPL_DECIM;
} else
{
T_op2 = T_op;
}
/*----------------------------------------------------------------------*
* DTX-CNG *
*----------------------------------------------------------------------*/
if(*mode == MRDTX) /* CNG mode */
{
/* Buffer isf's and energy */
#ifdef ASM_OPT /* asm optimization branch */
Residu_opt(&A[3 * (M + 1)], speech, exc, L_FRAME);
#else
Residu(&A[3 * (M + 1)], speech, exc, L_FRAME);
#endif
for (i = 0; i < L_FRAME; i++)
{
exc2[i] = shr(exc[i], Q_new);
}
L_tmp = 0;
for (i = 0; i < L_FRAME; i++)
L_tmp += (exc2[i] * exc2[i])<<1;
L_tmp >>= 1;
dtx_buffer(st->dtx_encSt, isf, L_tmp, codec_mode);
/* Quantize and code the ISFs */
dtx_enc(st->dtx_encSt, isf, exc2, &prms);
/* Convert ISFs to the cosine domain */
Isf_isp(isf, ispnew_q, M);
Isp_Az(ispnew_q, Aq, M, 0);
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
corr_gain = synthesis(Aq, &exc2[i_subfr], 0, &speech16k[i_subfr * 5 / 4], st);
}
Copy(isf, st->isfold, M);
/* reset speech coder memories */
Reset_encoder(st, 0);
/*--------------------------------------------------*
* Update signal for next frame. *
* -> save past of speech[] and wsp[]. *
*--------------------------------------------------*/
Copy(&old_speech[L_FRAME], st->old_speech, L_TOTAL - L_FRAME);
Copy(&old_wsp[L_FRAME / OPL_DECIM], st->old_wsp, PIT_MAX / OPL_DECIM);
return;
}
/*----------------------------------------------------------------------*
* ACELP *
*----------------------------------------------------------------------*/
/* Quantize and code the ISFs */
if (*ser_size <= NBBITS_7k)
{
Qpisf_2s_36b(isf, isf, st->past_isfq, indice, 4);
Parm_serial(indice[0], 8, &prms);
Parm_serial(indice[1], 8, &prms);
Parm_serial(indice[2], 7, &prms);
Parm_serial(indice[3], 7, &prms);
Parm_serial(indice[4], 6, &prms);
} else
{
Qpisf_2s_46b(isf, isf, st->past_isfq, indice, 4);
Parm_serial(indice[0], 8, &prms);
Parm_serial(indice[1], 8, &prms);
Parm_serial(indice[2], 6, &prms);
Parm_serial(indice[3], 7, &prms);
Parm_serial(indice[4], 7, &prms);
Parm_serial(indice[5], 5, &prms);
Parm_serial(indice[6], 5, &prms);
}
/* Check stability on isf : distance between old isf and current isf */
L_tmp = 0;
for (i = 0; i < M - 1; i++)
{
tmp = vo_sub(isf[i], st->isfold[i]);
L_tmp += (tmp * tmp)<<1;
}
tmp = extract_h(L_shl2(L_tmp, 8));
tmp = vo_mult(tmp, 26214); /* tmp = L_tmp*0.8/256 */
tmp = vo_sub(20480, tmp); /* 1.25 - tmp (in Q14) */
stab_fac = shl(tmp, 1);
if (stab_fac < 0)
{
stab_fac = 0;
}
Copy(isf, st->isfold, M);
/* Convert ISFs to the cosine domain */
Isf_isp(isf, ispnew_q, M);
if (st->first_frame != 0)
{
st->first_frame = 0;
Copy(ispnew_q, st->ispold_q, M);
}
/* Find the interpolated ISPs and convert to a[] for all subframes */
Int_isp(st->ispold_q, ispnew_q, interpol_frac, Aq);
/* update ispold[] for the next frame */
Copy(ispnew_q, st->ispold_q, M);
p_Aq = Aq;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
#ifdef ASM_OPT /* asm optimization branch */
Residu_opt(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
#else
Residu(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
#endif
p_Aq += (M + 1);
}
/* Buffer isf's and energy for dtx on non-speech frame */
if (vad_flag == 0)
{
for (i = 0; i < L_FRAME; i++)
{
exc2[i] = exc[i] >> Q_new;
}
L_tmp = 0;
for (i = 0; i < L_FRAME; i++) {
Word32 tmp = L_mult(exc2[i], exc2[i]); // (exc2[i] * exc2[i])<<1;
L_tmp = L_add(L_tmp, tmp);
}
L_tmp >>= 1;
dtx_buffer(st->dtx_encSt, isf, L_tmp, codec_mode);
}
/* range for closed loop pitch search in 1st subframe */
T0_min = T_op - 8;
if (T0_min < PIT_MIN)
{
T0_min = PIT_MIN;
}
T0_max = (T0_min + 15);
if(T0_max > PIT_MAX)
{
T0_max = PIT_MAX;
T0_min = T0_max - 15;
}
/*------------------------------------------------------------------------*
* Loop for every subframe in the analysis frame *
*------------------------------------------------------------------------*
* To find the pitch and innovation parameters. The subframe size is *
* L_SUBFR and the loop is repeated L_FRAME/L_SUBFR times. *
* - compute the target signal for pitch search *
* - compute impulse response of weighted synthesis filter (h1[]) *
* - find the closed-loop pitch parameters *
* - encode the pitch dealy *
* - find 2 lt prediction (with / without LP filter for lt pred) *
* - find 2 pitch gains and choose the best lt prediction. *
* - find target vector for codebook search *
* - update the impulse response h1[] for codebook search *
* - correlation between target vector and impulse response *
* - codebook search and encoding *
* - VQ of pitch and codebook gains *
* - find voicing factor and tilt of code for next subframe. *
* - update states of weighting filter *
* - find excitation and synthesis speech *
*------------------------------------------------------------------------*/
p_A = A;
p_Aq = Aq;
for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
{
pit_flag = i_subfr;
if ((i_subfr == 2 * L_SUBFR) && (*ser_size > NBBITS_7k))
{
pit_flag = 0;
/* range for closed loop pitch search in 3rd subframe */
T0_min = (T_op2 - 8);
if (T0_min < PIT_MIN)
{
T0_min = PIT_MIN;
}
T0_max = (T0_min + 15);
if (T0_max > PIT_MAX)
{
T0_max = PIT_MAX;
T0_min = (T0_max - 15);
}
}
/*-----------------------------------------------------------------------*
* *
* Find the target vector for pitch search: *
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ *
* *
* |------| res[n] *
* speech[n]---| A(z) |-------- *
* |------| | |--------| error[n] |------| *
* zero -- (-)--| 1/A(z) |-----------| W(z) |-- target *
* exc |--------| |------| *
* *
* Instead of subtracting the zero-input response of filters from *
* the weighted input speech, the above configuration is used to *
* compute the target vector. *
* *
*-----------------------------------------------------------------------*/
for (i = 0; i < M; i++)
{
error[i] = vo_sub(speech[i + i_subfr - M], st->mem_syn[i]);
}
#ifdef ASM_OPT /* asm optimization branch */
Residu_opt(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
#else
Residu(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR);
#endif
Syn_filt(p_Aq, &exc[i_subfr], error + M, L_SUBFR, error, 0);
Weight_a(p_A, Ap, GAMMA1, M);
#ifdef ASM_OPT /* asm optimization branch */
Residu_opt(Ap, error + M, xn, L_SUBFR);
#else
Residu(Ap, error + M, xn, L_SUBFR);
#endif
Deemph2(xn, TILT_FAC, L_SUBFR, &(st->mem_w0));
/*----------------------------------------------------------------------*
* Find approx. target in residual domain "cn[]" for inovation search. *
*----------------------------------------------------------------------*/
/* first half: xn[] --> cn[] */
Set_zero(code, M);
Copy(xn, code + M, L_SUBFR / 2);
tmp = 0;
Preemph2(code + M, TILT_FAC, L_SUBFR / 2, &tmp);
Weight_a(p_A, Ap, GAMMA1, M);
Syn_filt(Ap,code + M, code + M, L_SUBFR / 2, code, 0);
#ifdef ASM_OPT /* asm optimization branch */
Residu_opt(p_Aq,code + M, cn, L_SUBFR / 2);
#else
Residu(p_Aq,code + M, cn, L_SUBFR / 2);
#endif
/* second half: res[] --> cn[] (approximated and faster) */
Copy(&exc[i_subfr + (L_SUBFR / 2)], cn + (L_SUBFR / 2), L_SUBFR / 2);
/*---------------------------------------------------------------*
* Compute impulse response, h1[], of weighted synthesis filter *
*---------------------------------------------------------------*/
Set_zero(error, M + L_SUBFR);
Weight_a(p_A, error + M, GAMMA1, M);
vo_p0 = error+M;
vo_p3 = h1;
for (i = 0; i < L_SUBFR; i++)
{
L_tmp = *vo_p0 << 14; /* x4 (Q12 to Q14) */
vo_p1 = p_Aq + 1;
vo_p2 = vo_p0-1;
for (j = 1; j <= M/4; j++)
{
L_tmp = L_sub(L_tmp, *vo_p1++ * *vo_p2--);
L_tmp = L_sub(L_tmp, *vo_p1++ * *vo_p2--);
L_tmp = L_sub(L_tmp, *vo_p1++ * *vo_p2--);
L_tmp = L_sub(L_tmp, *vo_p1++ * *vo_p2--);
}
*vo_p3++ = *vo_p0++ = vo_round((L_tmp <<4));
}
/* deemph without division by 2 -> Q14 to Q15 */
tmp = 0;
Deemph2(h1, TILT_FAC, L_SUBFR, &tmp); /* h1 in Q14 */
/* h2 in Q12 for codebook search */
Copy(h1, h2, L_SUBFR);
/*---------------------------------------------------------------*
* scale xn[] and h1[] to avoid overflow in dot_product12() *
*---------------------------------------------------------------*/
#ifdef ASM_OPT /* asm optimization branch */
Scale_sig_opt(h2, L_SUBFR, -2);
Scale_sig_opt(xn, L_SUBFR, shift); /* scaling of xn[] to limit dynamic at 12 bits */
Scale_sig_opt(h1, L_SUBFR, 1 + shift); /* set h1[] in Q15 with scaling for convolution */
#else
Scale_sig(h2, L_SUBFR, -2);
Scale_sig(xn, L_SUBFR, shift); /* scaling of xn[] to limit dynamic at 12 bits */
Scale_sig(h1, L_SUBFR, 1 + shift); /* set h1[] in Q15 with scaling for convolution */
#endif
/*----------------------------------------------------------------------*
* Closed-loop fractional pitch search *
*----------------------------------------------------------------------*/
/* find closed loop fractional pitch lag */
if(*ser_size <= NBBITS_9k)
{
T0 = Pitch_fr4(&exc[i_subfr], xn, h1, T0_min, T0_max, &T0_frac,
pit_flag, PIT_MIN, PIT_FR1_8b, L_SUBFR);
/* encode pitch lag */
if (pit_flag == 0) /* if 1st/3rd subframe */
{
/*--------------------------------------------------------------*
* The pitch range for the 1st/3rd subframe is encoded with *
* 8 bits and is divided as follows: *
* PIT_MIN to PIT_FR1-1 resolution 1/2 (frac = 0 or 2) *
* PIT_FR1 to PIT_MAX resolution 1 (frac = 0) *
*--------------------------------------------------------------*/
if (T0 < PIT_FR1_8b)
{
index = ((T0 << 1) + (T0_frac >> 1) - (PIT_MIN<<1));
} else
{
index = ((T0 - PIT_FR1_8b) + ((PIT_FR1_8b - PIT_MIN)*2));
}
Parm_serial(index, 8, &prms);
/* find T0_min and T0_max for subframe 2 and 4 */
T0_min = (T0 - 8);
if (T0_min < PIT_MIN)
{
T0_min = PIT_MIN;
}
T0_max = T0_min + 15;
if (T0_max > PIT_MAX)
{
T0_max = PIT_MAX;
T0_min = (T0_max - 15);
}
} else
{ /* if subframe 2 or 4 */
/*--------------------------------------------------------------*
* The pitch range for subframe 2 or 4 is encoded with 5 bits: *
* T0_min to T0_max resolution 1/2 (frac = 0 or 2) *
*--------------------------------------------------------------*/
i = (T0 - T0_min);
index = (i << 1) + (T0_frac >> 1);
Parm_serial(index, 5, &prms);
}
} else
{
T0 = Pitch_fr4(&exc[i_subfr], xn, h1, T0_min, T0_max, &T0_frac,
pit_flag, PIT_FR2, PIT_FR1_9b, L_SUBFR);
/* encode pitch lag */
if (pit_flag == 0) /* if 1st/3rd subframe */
{
/*--------------------------------------------------------------*
* The pitch range for the 1st/3rd subframe is encoded with *
* 9 bits and is divided as follows: *
* PIT_MIN to PIT_FR2-1 resolution 1/4 (frac = 0,1,2 or 3) *
* PIT_FR2 to PIT_FR1-1 resolution 1/2 (frac = 0 or 1) *
* PIT_FR1 to PIT_MAX resolution 1 (frac = 0) *
*--------------------------------------------------------------*/
if (T0 < PIT_FR2)
{
index = ((T0 << 2) + T0_frac) - (PIT_MIN << 2);
} else if(T0 < PIT_FR1_9b)
{
index = ((((T0 << 1) + (T0_frac >> 1)) - (PIT_FR2<<1)) + ((PIT_FR2 - PIT_MIN)<<2));
} else
{
index = (((T0 - PIT_FR1_9b) + ((PIT_FR2 - PIT_MIN)<<2)) + ((PIT_FR1_9b - PIT_FR2)<<1));
}
Parm_serial(index, 9, &prms);
/* find T0_min and T0_max for subframe 2 and 4 */
T0_min = (T0 - 8);
if (T0_min < PIT_MIN)
{
T0_min = PIT_MIN;
}
T0_max = T0_min + 15;
if (T0_max > PIT_MAX)
{
T0_max = PIT_MAX;
T0_min = (T0_max - 15);
}
} else
{ /* if subframe 2 or 4 */
/*--------------------------------------------------------------*
* The pitch range for subframe 2 or 4 is encoded with 6 bits: *
* T0_min to T0_max resolution 1/4 (frac = 0,1,2 or 3) *
*--------------------------------------------------------------*/
i = (T0 - T0_min);
index = (i << 2) + T0_frac;
Parm_serial(index, 6, &prms);
}
}
/*-----------------------------------------------------------------*
* Gain clipping test to avoid unstable synthesis on frame erasure *
*-----------------------------------------------------------------*/
clip_gain = 0;
if((st->gp_clip[0] < 154) && (st->gp_clip[1] > 14746))
clip_gain = 1;
/*-----------------------------------------------------------------*
* - find unity gain pitch excitation (adaptive codebook entry) *
* with fractional interpolation. *
* - find filtered pitch exc. y1[]=exc[] convolved with h1[]) *
* - compute pitch gain1 *
*-----------------------------------------------------------------*/
/* find pitch exitation */
#ifdef ASM_OPT /* asm optimization branch */
pred_lt4_asm(&exc[i_subfr], T0, T0_frac, L_SUBFR + 1);
#else
Pred_lt4(&exc[i_subfr], T0, T0_frac, L_SUBFR + 1);
#endif
if (*ser_size > NBBITS_9k)
{
#ifdef ASM_OPT /* asm optimization branch */
Convolve_asm(&exc[i_subfr], h1, y1, L_SUBFR);
#else
Convolve(&exc[i_subfr], h1, y1, L_SUBFR);
#endif
gain1 = G_pitch(xn, y1, g_coeff, L_SUBFR);
/* clip gain if necessary to avoid problem at decoder */
if ((clip_gain != 0) && (gain1 > GP_CLIP))
{
gain1 = GP_CLIP;
}
/* find energy of new target xn2[] */
Updt_tar(xn, dn, y1, gain1, L_SUBFR); /* dn used temporary */
} else
{
gain1 = 0;
}
/*-----------------------------------------------------------------*
* - find pitch excitation filtered by 1st order LP filter. *
* - find filtered pitch exc. y2[]=exc[] convolved with h1[]) *
* - compute pitch gain2 *
*-----------------------------------------------------------------*/
/* find pitch excitation with lp filter */
vo_p0 = exc + i_subfr-1;
vo_p1 = code;
/* find pitch excitation with lp filter */
for (i = 0; i < L_SUBFR/2; i++)
{
L_tmp = 5898 * *vo_p0++;
L_tmp1 = 5898 * *vo_p0;
L_tmp += 20972 * *vo_p0++;
L_tmp1 += 20972 * *vo_p0++;
L_tmp1 += 5898 * *vo_p0--;
L_tmp += 5898 * *vo_p0;
*vo_p1++ = (L_tmp + 0x4000)>>15;
*vo_p1++ = (L_tmp1 + 0x4000)>>15;
}
#ifdef ASM_OPT /* asm optimization branch */
Convolve_asm(code, h1, y2, L_SUBFR);
#else
Convolve(code, h1, y2, L_SUBFR);
#endif
gain2 = G_pitch(xn, y2, g_coeff2, L_SUBFR);
/* clip gain if necessary to avoid problem at decoder */
if ((clip_gain != 0) && (gain2 > GP_CLIP))
{
gain2 = GP_CLIP;
}
/* find energy of new target xn2[] */
Updt_tar(xn, xn2, y2, gain2, L_SUBFR);
/*-----------------------------------------------------------------*
* use the best prediction (minimise quadratic error). *
*-----------------------------------------------------------------*/
select = 0;
if(*ser_size > NBBITS_9k)
{
L_tmp = 0L;
vo_p0 = dn;
vo_p1 = xn2;
for (i = 0; i < L_SUBFR/2; i++)
{
L_tmp = L_add(L_tmp, *vo_p0 * *vo_p0);
vo_p0++;
L_tmp = L_sub(L_tmp, *vo_p1 * *vo_p1);
vo_p1++;
L_tmp = L_add(L_tmp, *vo_p0 * *vo_p0);
vo_p0++;
L_tmp = L_sub(L_tmp, *vo_p1 * *vo_p1);
vo_p1++;
}
if (L_tmp <= 0)
{
select = 1;
}
Parm_serial(select, 1, &prms);
}
if (select == 0)
{
/* use the lp filter for pitch excitation prediction */
gain_pit = gain2;
Copy(code, &exc[i_subfr], L_SUBFR);
Copy(y2, y1, L_SUBFR);
Copy(g_coeff2, g_coeff, 4);
} else
{
/* no filter used for pitch excitation prediction */
gain_pit = gain1;
Copy(dn, xn2, L_SUBFR); /* target vector for codebook search */
}
/*-----------------------------------------------------------------*
* - update cn[] for codebook search *
*-----------------------------------------------------------------*/
Updt_tar(cn, cn, &exc[i_subfr], gain_pit, L_SUBFR);
#ifdef ASM_OPT /* asm optimization branch */
Scale_sig_opt(cn, L_SUBFR, shift); /* scaling of cn[] to limit dynamic at 12 bits */
#else
Scale_sig(cn, L_SUBFR, shift); /* scaling of cn[] to limit dynamic at 12 bits */
#endif
/*-----------------------------------------------------------------*
* - include fixed-gain pitch contribution into impulse resp. h1[] *
*-----------------------------------------------------------------*/
tmp = 0;
Preemph(h2, st->tilt_code, L_SUBFR, &tmp);
if (T0_frac > 2)
T0 = (T0 + 1);
Pit_shrp(h2, T0, PIT_SHARP, L_SUBFR);
/*-----------------------------------------------------------------*
* - Correlation between target xn2[] and impulse response h1[] *
* - Innovative codebook search *
*-----------------------------------------------------------------*/
cor_h_x(h2, xn2, dn);
if (*ser_size <= NBBITS_7k)
{
ACELP_2t64_fx(dn, cn, h2, code, y2, indice);
Parm_serial(indice[0], 12, &prms);
} else if(*ser_size <= NBBITS_9k)
{
ACELP_4t64_fx(dn, cn, h2, code, y2, 20, *ser_size, indice);
Parm_serial(indice[0], 5, &prms);
Parm_serial(indice[1], 5, &prms);
Parm_serial(indice[2], 5, &prms);
Parm_serial(indice[3], 5, &prms);
} else if(*ser_size <= NBBITS_12k)
{
ACELP_4t64_fx(dn, cn, h2, code, y2, 36, *ser_size, indice);
Parm_serial(indice[0], 9, &prms);
Parm_serial(indice[1], 9, &prms);
Parm_serial(indice[2], 9, &prms);
Parm_serial(indice[3], 9, &prms);
} else if(*ser_size <= NBBITS_14k)
{
ACELP_4t64_fx(dn, cn, h2, code, y2, 44, *ser_size, indice);
Parm_serial(indice[0], 13, &prms);
Parm_serial(indice[1], 13, &prms);
Parm_serial(indice[2], 9, &prms);
Parm_serial(indice[3], 9, &prms);
} else if(*ser_size <= NBBITS_16k)
{
ACELP_4t64_fx(dn, cn, h2, code, y2, 52, *ser_size, indice);
Parm_serial(indice[0], 13, &prms);
Parm_serial(indice[1], 13, &prms);
Parm_serial(indice[2], 13, &prms);
Parm_serial(indice[3], 13, &prms);
} else if(*ser_size <= NBBITS_18k)
{
ACELP_4t64_fx(dn, cn, h2, code, y2, 64, *ser_size, indice);
Parm_serial(indice[0], 2, &prms);
Parm_serial(indice[1], 2, &prms);
Parm_serial(indice[2], 2, &prms);
Parm_serial(indice[3], 2, &prms);
Parm_serial(indice[4], 14, &prms);
Parm_serial(indice[5], 14, &prms);
Parm_serial(indice[6], 14, &prms);
Parm_serial(indice[7], 14, &prms);
} else if(*ser_size <= NBBITS_20k)
{
ACELP_4t64_fx(dn, cn, h2, code, y2, 72, *ser_size, indice);
Parm_serial(indice[0], 10, &prms);
Parm_serial(indice[1], 10, &prms);
Parm_serial(indice[2], 2, &prms);
Parm_serial(indice[3], 2, &prms);
Parm_serial(indice[4], 10, &prms);
Parm_serial(indice[5], 10, &prms);
Parm_serial(indice[6], 14, &prms);
Parm_serial(indice[7], 14, &prms);
} else
{
ACELP_4t64_fx(dn, cn, h2, code, y2, 88, *ser_size, indice);
Parm_serial(indice[0], 11, &prms);
Parm_serial(indice[1], 11, &prms);
Parm_serial(indice[2], 11, &prms);
Parm_serial(indice[3], 11, &prms);
Parm_serial(indice[4], 11, &prms);
Parm_serial(indice[5], 11, &prms);
Parm_serial(indice[6], 11, &prms);
Parm_serial(indice[7], 11, &prms);
}
/*-------------------------------------------------------*
* - Add the fixed-gain pitch contribution to code[]. *
*-------------------------------------------------------*/
tmp = 0;
Preemph(code, st->tilt_code, L_SUBFR, &tmp);
Pit_shrp(code, T0, PIT_SHARP, L_SUBFR);
/*----------------------------------------------------------*
* - Compute the fixed codebook gain *
* - quantize fixed codebook gain *
*----------------------------------------------------------*/
if(*ser_size <= NBBITS_9k)
{
index = Q_gain2(xn, y1, Q_new + shift, y2, code, g_coeff, L_SUBFR, 6,
&gain_pit, &L_gain_code, clip_gain, st->qua_gain);
Parm_serial(index, 6, &prms);
} else
{
index = Q_gain2(xn, y1, Q_new + shift, y2, code, g_coeff, L_SUBFR, 7,
&gain_pit, &L_gain_code, clip_gain, st->qua_gain);
Parm_serial(index, 7, &prms);
}
/* test quantized gain of pitch for pitch clipping algorithm */
Gp_clip_test_gain_pit(gain_pit, st->gp_clip);
L_tmp = L_shl(L_gain_code, Q_new);
gain_code = extract_h(L_add(L_tmp, 0x8000));
/*----------------------------------------------------------*
* Update parameters for the next subframe. *
* - tilt of code: 0.0 (unvoiced) to 0.5 (voiced) *
*----------------------------------------------------------*/
/* find voice factor in Q15 (1=voiced, -1=unvoiced) */
Copy(&exc[i_subfr], exc2, L_SUBFR);
#ifdef ASM_OPT /* asm optimization branch */
Scale_sig_opt(exc2, L_SUBFR, shift);
#else
Scale_sig(exc2, L_SUBFR, shift);
#endif
voice_fac = voice_factor(exc2, shift, gain_pit, code, gain_code, L_SUBFR);
/* tilt of code for next subframe: 0.5=voiced, 0=unvoiced */
st->tilt_code = ((voice_fac >> 2) + 8192);
/*------------------------------------------------------*
* - Update filter's memory "mem_w0" for finding the *
* target vector in the next subframe. *
* - Find the total excitation *
* - Find synthesis speech to update mem_syn[]. *
*------------------------------------------------------*/
/* y2 in Q9, gain_pit in Q14 */
L_tmp = L_mult(gain_code, y2[L_SUBFR - 1]);
L_tmp = L_shl(L_tmp, (5 + shift));
L_tmp = L_negate(L_tmp);
L_tmp += (xn[L_SUBFR - 1] * 16384)<<1;
L_tmp -= (y1[L_SUBFR - 1] * gain_pit)<<1;
L_tmp = L_shl(L_tmp, (1 - shift));
st->mem_w0 = extract_h(L_add(L_tmp, 0x8000));
if (*ser_size >= NBBITS_24k)
Copy(&exc[i_subfr], exc2, L_SUBFR);
for (i = 0; i < L_SUBFR; i++)
{
Word32 tmp;
/* code in Q9, gain_pit in Q14 */
L_tmp = L_mult(gain_code, code[i]);
L_tmp = L_shl(L_tmp, 5);
tmp = L_mult(exc[i + i_subfr], gain_pit); // (exc[i + i_subfr] * gain_pit)<<1
L_tmp = L_add(L_tmp, tmp);
L_tmp = L_shl2(L_tmp, 1);
exc[i + i_subfr] = extract_h(L_add(L_tmp, 0x8000));
}
Syn_filt(p_Aq,&exc[i_subfr], synth, L_SUBFR, st->mem_syn, 1);
if(*ser_size >= NBBITS_24k)
{
/*------------------------------------------------------------*
* phase dispersion to enhance noise in low bit rate *
*------------------------------------------------------------*/
/* L_gain_code in Q16 */
VO_L_Extract(L_gain_code, &gain_code, &gain_code_lo);
/*------------------------------------------------------------*
* noise enhancer *
* ~~~~~~~~~~~~~~ *
* - Enhance excitation on noise. (modify gain of code) *
* If signal is noisy and LPC filter is stable, move gain *
* of code 1.5 dB toward gain of code threshold. *
* This decrease by 3 dB noise energy variation. *
*------------------------------------------------------------*/
tmp = (16384 - (voice_fac >> 1)); /* 1=unvoiced, 0=voiced */
fac = vo_mult(stab_fac, tmp);
L_tmp = L_gain_code;
if(L_tmp < st->L_gc_thres)
{
L_tmp = vo_L_add(L_tmp, Mpy_32_16(gain_code, gain_code_lo, 6226));
if(L_tmp > st->L_gc_thres)
{
L_tmp = st->L_gc_thres;
}
} else
{
L_tmp = Mpy_32_16(gain_code, gain_code_lo, 27536);
if(L_tmp < st->L_gc_thres)
{
L_tmp = st->L_gc_thres;
}
}
st->L_gc_thres = L_tmp;
L_gain_code = Mpy_32_16(gain_code, gain_code_lo, (32767 - fac));
VO_L_Extract(L_tmp, &gain_code, &gain_code_lo);
L_gain_code = vo_L_add(L_gain_code, Mpy_32_16(gain_code, gain_code_lo, fac));
/*------------------------------------------------------------*
* pitch enhancer *
* ~~~~~~~~~~~~~~ *
* - Enhance excitation on voice. (HP filtering of code) *
* On voiced signal, filtering of code by a smooth fir HP *
* filter to decrease energy of code in low frequency. *
*------------------------------------------------------------*/
tmp = ((voice_fac >> 3) + 4096); /* 0.25=voiced, 0=unvoiced */
L_tmp = L_deposit_h(code[0]);
L_tmp -= (code[1] * tmp)<<1;
code2[0] = vo_round(L_tmp);
for (i = 1; i < L_SUBFR - 1; i++)
{
L_tmp = L_deposit_h(code[i]);
L_tmp -= (code[i + 1] * tmp)<<1;
L_tmp -= (code[i - 1] * tmp)<<1;
code2[i] = vo_round(L_tmp);
}
L_tmp = L_deposit_h(code[L_SUBFR - 1]);
L_tmp -= (code[L_SUBFR - 2] * tmp)<<1;
code2[L_SUBFR - 1] = vo_round(L_tmp);
/* build excitation */
gain_code = vo_round(L_shl(L_gain_code, Q_new));
for (i = 0; i < L_SUBFR; i++)
{
L_tmp = L_mult(code2[i], gain_code);
L_tmp = L_shl(L_tmp, 5);
L_tmp = L_add(L_tmp, L_mult(exc2[i], gain_pit));
L_tmp = L_shl(L_tmp, 1);
exc2[i] = voround(L_tmp);
}
corr_gain = synthesis(p_Aq, exc2, Q_new, &speech16k[i_subfr * 5 / 4], st);
Parm_serial(corr_gain, 4, &prms);
}
p_A += (M + 1);
p_Aq += (M + 1);
} /* end of subframe loop */
/*--------------------------------------------------*
* Update signal for next frame. *
* -> save past of speech[], wsp[] and exc[]. *
*--------------------------------------------------*/
Copy(&old_speech[L_FRAME], st->old_speech, L_TOTAL - L_FRAME);
Copy(&old_wsp[L_FRAME / OPL_DECIM], st->old_wsp, PIT_MAX / OPL_DECIM);
Copy(&old_exc[L_FRAME], st->old_exc, PIT_MAX + L_INTERPOL);
return;
}
/*-----------------------------------------------------*
* Function synthesis() *
* *
* Synthesis of signal at 16kHz with HF extension. *
* *
*-----------------------------------------------------*/
static Word16 synthesis(
Word16 Aq[], /* A(z) : quantized Az */
Word16 exc[], /* (i) : excitation at 12kHz */
Word16 Q_new, /* (i) : scaling performed on exc */
Word16 synth16k[], /* (o) : 16kHz synthesis signal */
Coder_State * st /* (i/o) : State structure */
)
{
Word16 fac, tmp, exp;
Word16 ener, exp_ener;
Word32 L_tmp, i;
Word16 synth_hi[M + L_SUBFR], synth_lo[M + L_SUBFR];
Word16 synth[L_SUBFR];
Word16 HF[L_SUBFR16k]; /* High Frequency vector */
Word16 Ap[M + 1];
Word16 HF_SP[L_SUBFR16k]; /* High Frequency vector (from original signal) */
Word16 HP_est_gain, HP_calc_gain, HP_corr_gain;
Word16 dist_min, dist;
Word16 HP_gain_ind = 0;
Word16 gain1, gain2;
Word16 weight1, weight2;
/*------------------------------------------------------------*
* speech synthesis *
* ~~~~~~~~~~~~~~~~ *
* - Find synthesis speech corresponding to exc2[]. *
* - Perform fixed deemphasis and hp 50hz filtering. *
* - Oversampling from 12.8kHz to 16kHz. *
*------------------------------------------------------------*/
Copy(st->mem_syn_hi, synth_hi, M);
Copy(st->mem_syn_lo, synth_lo, M);
#ifdef ASM_OPT /* asm optimization branch */
Syn_filt_32_asm(Aq, M, exc, Q_new, synth_hi + M, synth_lo + M, L_SUBFR);
#else
Syn_filt_32(Aq, M, exc, Q_new, synth_hi + M, synth_lo + M, L_SUBFR);
#endif
Copy(synth_hi + L_SUBFR, st->mem_syn_hi, M);
Copy(synth_lo + L_SUBFR, st->mem_syn_lo, M);
#ifdef ASM_OPT /* asm optimization branch */
Deemph_32_asm(synth_hi + M, synth_lo + M, synth, &(st->mem_deemph));
#else
Deemph_32(synth_hi + M, synth_lo + M, synth, PREEMPH_FAC, L_SUBFR, &(st->mem_deemph));
#endif
HP50_12k8(synth, L_SUBFR, st->mem_sig_out);
/* Original speech signal as reference for high band gain quantisation */
for (i = 0; i < L_SUBFR16k; i++)
{
HF_SP[i] = synth16k[i];
}
/*------------------------------------------------------*
* HF noise synthesis *
* ~~~~~~~~~~~~~~~~~~ *
* - Generate HF noise between 5.5 and 7.5 kHz. *
* - Set energy of noise according to synthesis tilt. *
* tilt > 0.8 ==> - 14 dB (voiced) *
* tilt 0.5 ==> - 6 dB (voiced or noise) *
* tilt < 0.0 ==> 0 dB (noise) *
*------------------------------------------------------*/
/* generate white noise vector */
for (i = 0; i < L_SUBFR16k; i++)
{
HF[i] = Random(&(st->seed2))>>3;
}
/* energy of excitation */
#ifdef ASM_OPT /* asm optimization branch */
Scale_sig_opt(exc, L_SUBFR, -3);
Q_new = Q_new - 3;
ener = extract_h(Dot_product12_asm(exc, exc, L_SUBFR, &exp_ener));
#else
Scale_sig(exc, L_SUBFR, -3);
Q_new = Q_new - 3;
ener = extract_h(Dot_product12(exc, exc, L_SUBFR, &exp_ener));
#endif
exp_ener = exp_ener - (Q_new + Q_new);
/* set energy of white noise to energy of excitation */
#ifdef ASM_OPT /* asm optimization branch */
tmp = extract_h(Dot_product12_asm(HF, HF, L_SUBFR16k, &exp));
#else
tmp = extract_h(Dot_product12(HF, HF, L_SUBFR16k, &exp));
#endif
if(tmp > ener)
{
tmp = (tmp >> 1); /* Be sure tmp < ener */
exp = (exp + 1);
}
L_tmp = L_deposit_h(div_s(tmp, ener)); /* result is normalized */
exp = (exp - exp_ener);
Isqrt_n(&L_tmp, &exp);
L_tmp = L_shl(L_tmp, (exp + 1)); /* L_tmp x 2, L_tmp in Q31 */
tmp = extract_h(L_tmp); /* tmp = 2 x sqrt(ener_exc/ener_hf) */
for (i = 0; i < L_SUBFR16k; i++)
{
HF[i] = vo_mult(HF[i], tmp);
}
/* find tilt of synthesis speech (tilt: 1=voiced, -1=unvoiced) */
HP400_12k8(synth, L_SUBFR, st->mem_hp400);
L_tmp = 1L;
for (i = 0; i < L_SUBFR; i++)
L_tmp += (synth[i] * synth[i])<<1;
exp = norm_l(L_tmp);
ener = extract_h(L_tmp << exp); /* ener = r[0] */
L_tmp = 1L;
for (i = 1; i < L_SUBFR; i++)
L_tmp +=(synth[i] * synth[i - 1])<<1;
tmp = extract_h(L_tmp << exp); /* tmp = r[1] */
if (tmp > 0)
{
fac = div_s(tmp, ener);
} else
{
fac = 0;
}
/* modify energy of white noise according to synthesis tilt */
gain1 = 32767 - fac;
gain2 = vo_mult(gain1, 20480);
gain2 = shl(gain2, 1);
if (st->vad_hist > 0)
{
weight1 = 0;
weight2 = 32767;
} else
{
weight1 = 32767;
weight2 = 0;
}
tmp = vo_mult(weight1, gain1);
tmp = add1(tmp, vo_mult(weight2, gain2));
if (tmp != 0)
{
tmp = (tmp + 1);
}
HP_est_gain = tmp;
if(HP_est_gain < 3277)
{
HP_est_gain = 3277; /* 0.1 in Q15 */
}
/* synthesis of noise: 4.8kHz..5.6kHz --> 6kHz..7kHz */
Weight_a(Aq, Ap, 19661, M); /* fac=0.6 */
#ifdef ASM_OPT /* asm optimization branch */
Syn_filt_asm(Ap, HF, HF, st->mem_syn_hf);
/* noise High Pass filtering (1ms of delay) */
Filt_6k_7k_asm(HF, L_SUBFR16k, st->mem_hf);
/* filtering of the original signal */
Filt_6k_7k_asm(HF_SP, L_SUBFR16k, st->mem_hf2);
/* check the gain difference */
Scale_sig_opt(HF_SP, L_SUBFR16k, -1);
ener = extract_h(Dot_product12_asm(HF_SP, HF_SP, L_SUBFR16k, &exp_ener));
/* set energy of white noise to energy of excitation */
tmp = extract_h(Dot_product12_asm(HF, HF, L_SUBFR16k, &exp));
#else
Syn_filt(Ap, HF, HF, L_SUBFR16k, st->mem_syn_hf, 1);
/* noise High Pass filtering (1ms of delay) */
Filt_6k_7k(HF, L_SUBFR16k, st->mem_hf);
/* filtering of the original signal */
Filt_6k_7k(HF_SP, L_SUBFR16k, st->mem_hf2);
/* check the gain difference */
Scale_sig(HF_SP, L_SUBFR16k, -1);
ener = extract_h(Dot_product12(HF_SP, HF_SP, L_SUBFR16k, &exp_ener));
/* set energy of white noise to energy of excitation */
tmp = extract_h(Dot_product12(HF, HF, L_SUBFR16k, &exp));
#endif
if (tmp > ener)
{
tmp = (tmp >> 1); /* Be sure tmp < ener */
exp = (exp + 1);
}
L_tmp = L_deposit_h(div_s(tmp, ener)); /* result is normalized */
exp = vo_sub(exp, exp_ener);
Isqrt_n(&L_tmp, &exp);
L_tmp = L_shl(L_tmp, exp); /* L_tmp, L_tmp in Q31 */
HP_calc_gain = extract_h(L_tmp); /* tmp = sqrt(ener_input/ener_hf) */
/* st->gain_alpha *= st->dtx_encSt->dtxHangoverCount/7 */
L_tmp = (vo_L_mult(st->dtx_encSt->dtxHangoverCount, 4681) << 15);
st->gain_alpha = vo_mult(st->gain_alpha, extract_h(L_tmp));
if(st->dtx_encSt->dtxHangoverCount > 6)
st->gain_alpha = 32767;
HP_est_gain = HP_est_gain >> 1; /* From Q15 to Q14 */
HP_corr_gain = add1(vo_mult(HP_calc_gain, st->gain_alpha), vo_mult((32767 - st->gain_alpha), HP_est_gain));
/* Quantise the correction gain */
dist_min = 32767;
for (i = 0; i < 16; i++)
{
dist = vo_mult((HP_corr_gain - HP_gain[i]), (HP_corr_gain - HP_gain[i]));
if (dist_min > dist)
{
dist_min = dist;
HP_gain_ind = i;
}
}
HP_corr_gain = HP_gain[HP_gain_ind];
/* return the quantised gain index when using the highest mode, otherwise zero */
return (HP_gain_ind);
}
/*************************************************
*
* Breif: Codec main function
*
**************************************************/
int AMR_Enc_Encode(HAMRENC hCodec)
{
Word32 i;
Coder_State *gData = (Coder_State*)hCodec;
Word16 *signal;
Word16 packed_size = 0;
Word16 prms[NB_BITS_MAX];
Word16 coding_mode = 0, nb_bits, allow_dtx, mode, reset_flag;
mode = gData->mode;
coding_mode = gData->mode;
nb_bits = nb_of_bits[mode];
signal = (Word16 *)gData->inputStream;
allow_dtx = gData->allow_dtx;
/* check for homing frame */
reset_flag = encoder_homing_frame_test(signal);
for (i = 0; i < L_FRAME16k; i++) /* Delete the 2 LSBs (14-bit input) */
{
*(signal + i) = (Word16) (*(signal + i) & 0xfffC);
}
coder(&coding_mode, signal, prms, &nb_bits, gData, allow_dtx);
packed_size = PackBits(prms, coding_mode, mode, gData);
if (reset_flag != 0)
{
Reset_encoder(gData, 1);
}
return packed_size;
}
/***************************************************************************
*
*Brief: Codec API function --- Initialize the codec and return a codec handle
*
***************************************************************************/
VO_U32 VO_API voAMRWB_Init(VO_HANDLE * phCodec, /* o: the audio codec handle */
VO_AUDIO_CODINGTYPE vType, /* i: Codec Type ID */
VO_CODEC_INIT_USERDATA * pUserData /* i: init Parameters */
)
{
Coder_State *st;
FrameStream *stream;
#ifdef USE_DEAULT_MEM
VO_MEM_OPERATOR voMemoprator;
#endif
VO_MEM_OPERATOR *pMemOP;
UNUSED(vType);
int interMem = 0;
if(pUserData == NULL || pUserData->memflag != VO_IMF_USERMEMOPERATOR || pUserData->memData == NULL )
{
#ifdef USE_DEAULT_MEM
voMemoprator.Alloc = cmnMemAlloc;
voMemoprator.Copy = cmnMemCopy;
voMemoprator.Free = cmnMemFree;
voMemoprator.Set = cmnMemSet;
voMemoprator.Check = cmnMemCheck;
interMem = 1;
pMemOP = &voMemoprator;
#else
*phCodec = NULL;
return VO_ERR_INVALID_ARG;
#endif
}
else
{
pMemOP = (VO_MEM_OPERATOR *)pUserData->memData;
}
/*-------------------------------------------------------------------------*
* Memory allocation for coder state. *
*-------------------------------------------------------------------------*/
if ((st = (Coder_State *)mem_malloc(pMemOP, sizeof(Coder_State), 32, VO_INDEX_ENC_AMRWB)) == NULL)
{
return VO_ERR_OUTOF_MEMORY;
}
st->vadSt = NULL;
st->dtx_encSt = NULL;
st->sid_update_counter = 3;
st->sid_handover_debt = 0;
st->prev_ft = TX_SPEECH;
st->inputStream = NULL;
st->inputSize = 0;
/* Default setting */
st->mode = VOAMRWB_MD2385; /* bit rate 23.85kbps */
st->frameType = VOAMRWB_RFC3267; /* frame type: RFC3267 */
st->allow_dtx = 0; /* disable DTX mode */
st->outputStream = NULL;
st->outputSize = 0;
st->stream = (FrameStream *)mem_malloc(pMemOP, sizeof(FrameStream), 32, VO_INDEX_ENC_AMRWB);
if(st->stream == NULL)
return VO_ERR_OUTOF_MEMORY;
st->stream->frame_ptr = (unsigned char *)mem_malloc(pMemOP, Frame_Maxsize, 32, VO_INDEX_ENC_AMRWB);
if(st->stream->frame_ptr == NULL)
return VO_ERR_OUTOF_MEMORY;
stream = st->stream;
voAWB_InitFrameBuffer(stream);
wb_vad_init(&(st->vadSt), pMemOP);
dtx_enc_init(&(st->dtx_encSt), isf_init, pMemOP);
Reset_encoder((void *) st, 1);
if(interMem)
{
st->voMemoprator.Alloc = cmnMemAlloc;
st->voMemoprator.Copy = cmnMemCopy;
st->voMemoprator.Free = cmnMemFree;
st->voMemoprator.Set = cmnMemSet;
st->voMemoprator.Check = cmnMemCheck;
pMemOP = &st->voMemoprator;
}
st->pvoMemop = pMemOP;
*phCodec = (void *) st;
return VO_ERR_NONE;
}
/**********************************************************************************
*
* Brief: Codec API function: Input PCM data
*
***********************************************************************************/
VO_U32 VO_API voAMRWB_SetInputData(
VO_HANDLE hCodec, /* i/o: The codec handle which was created by Init function */
VO_CODECBUFFER * pInput /* i: The input buffer parameter */
)
{
Coder_State *gData;
FrameStream *stream;
if(NULL == hCodec)
{
return VO_ERR_INVALID_ARG;
}
gData = (Coder_State *)hCodec;
stream = gData->stream;
if(NULL == pInput || NULL == pInput->Buffer)
{
return VO_ERR_INVALID_ARG;
}
stream->set_ptr = pInput->Buffer;
stream->set_len = pInput->Length;
stream->frame_ptr = stream->frame_ptr_bk;
stream->used_len = 0;
return VO_ERR_NONE;
}
/**************************************************************************************
*
* Brief: Codec API function: Get the compression audio data frame by frame
*
***************************************************************************************/
VO_U32 VO_API voAMRWB_GetOutputData(
VO_HANDLE hCodec, /* i: The Codec Handle which was created by Init function*/
VO_CODECBUFFER * pOutput, /* o: The output audio data */
VO_AUDIO_OUTPUTINFO * pAudioFormat /* o: The encoder module filled audio format and used the input size*/
)
{
Coder_State* gData = (Coder_State*)hCodec;
VO_MEM_OPERATOR *pMemOP;
FrameStream *stream = (FrameStream *)gData->stream;
pMemOP = (VO_MEM_OPERATOR *)gData->pvoMemop;
if(stream->framebuffer_len < Frame_MaxByte) /* check the work buffer len */
{
stream->frame_storelen = stream->framebuffer_len;
if(stream->frame_storelen)
{
pMemOP->Copy(VO_INDEX_ENC_AMRWB, stream->frame_ptr_bk , stream->frame_ptr , stream->frame_storelen);
}
if(stream->set_len > 0)
{
voAWB_UpdateFrameBuffer(stream, pMemOP);
}
if(stream->framebuffer_len < Frame_MaxByte)
{
if(pAudioFormat)
pAudioFormat->InputUsed = stream->used_len;
return VO_ERR_INPUT_BUFFER_SMALL;
}
}
gData->inputStream = stream->frame_ptr;
gData->outputStream = (unsigned short*)pOutput->Buffer;
gData->outputSize = AMR_Enc_Encode(gData); /* encoder main function */
pOutput->Length = gData->outputSize; /* get the output buffer length */
stream->frame_ptr += 640; /* update the work buffer ptr */
stream->framebuffer_len -= 640;
if(pAudioFormat) /* return output audio information */
{
pAudioFormat->Format.Channels = 1;
pAudioFormat->Format.SampleRate = 8000;
pAudioFormat->Format.SampleBits = 16;
pAudioFormat->InputUsed = stream->used_len;
}
return VO_ERR_NONE;
}
/*************************************************************************
*
* Brief: Codec API function---set the data by specified parameter ID
*
*************************************************************************/
VO_U32 VO_API voAMRWB_SetParam(
VO_HANDLE hCodec, /* i/o: The Codec Handle which was created by Init function */
VO_S32 uParamID, /* i: The param ID */
VO_PTR pData /* i: The param value depend on the ID */
)
{
Coder_State* gData = (Coder_State*)hCodec;
FrameStream *stream = (FrameStream *)(gData->stream);
int *lValue = (int*)pData;
switch(uParamID)
{
/* setting AMR-WB frame type*/
case VO_PID_AMRWB_FRAMETYPE:
if(*lValue < VOAMRWB_DEFAULT || *lValue > VOAMRWB_RFC3267)
return VO_ERR_WRONG_PARAM_ID;
gData->frameType = *lValue;
break;
/* setting AMR-WB bit rate */
case VO_PID_AMRWB_MODE:
{
if(*lValue < VOAMRWB_MD66 || *lValue > VOAMRWB_MD2385)
return VO_ERR_WRONG_PARAM_ID;
gData->mode = *lValue;
}
break;
/* enable or disable DTX mode */
case VO_PID_AMRWB_DTX:
gData->allow_dtx = (Word16)(*lValue);
break;
case VO_PID_COMMON_HEADDATA:
break;
/* flush the work buffer */
case VO_PID_COMMON_FLUSH:
stream->set_ptr = NULL;
stream->frame_storelen = 0;
stream->framebuffer_len = 0;
stream->set_len = 0;
break;
default:
return VO_ERR_WRONG_PARAM_ID;
}
return VO_ERR_NONE;
}
/**************************************************************************
*
*Brief: Codec API function---Get the data by specified parameter ID
*
***************************************************************************/
VO_U32 VO_API voAMRWB_GetParam(
VO_HANDLE hCodec, /* i: The Codec Handle which was created by Init function */
VO_S32 uParamID, /* i: The param ID */
VO_PTR pData /* o: The param value depend on the ID */
)
{
int temp;
Coder_State* gData = (Coder_State*)hCodec;
if (gData==NULL)
return VO_ERR_INVALID_ARG;
switch(uParamID)
{
/* output audio format */
case VO_PID_AMRWB_FORMAT:
{
VO_AUDIO_FORMAT* fmt = (VO_AUDIO_FORMAT*)pData;
fmt->Channels = 1;
fmt->SampleRate = 16000;
fmt->SampleBits = 16;
break;
}
/* output audio channel number */
case VO_PID_AMRWB_CHANNELS:
temp = 1;
pData = (void *)(&temp);
break;
/* output audio sample rate */
case VO_PID_AMRWB_SAMPLERATE:
temp = 16000;
pData = (void *)(&temp);
break;
/* output audio frame type */
case VO_PID_AMRWB_FRAMETYPE:
temp = gData->frameType;
pData = (void *)(&temp);
break;
/* output audio bit rate */
case VO_PID_AMRWB_MODE:
temp = gData->mode;
pData = (void *)(&temp);
break;
default:
return VO_ERR_WRONG_PARAM_ID;
}
return VO_ERR_NONE;
}
/***********************************************************************************
*
* Brief: Codec API function---Release the codec after all encoder operations are done
*
*************************************************************************************/
VO_U32 VO_API voAMRWB_Uninit(VO_HANDLE hCodec /* i/o: Codec handle pointer */
)
{
Coder_State* gData = (Coder_State*)hCodec;
VO_MEM_OPERATOR *pMemOP;
pMemOP = gData->pvoMemop;
if(hCodec)
{
if(gData->stream)
{
if(gData->stream->frame_ptr_bk)
{
mem_free(pMemOP, gData->stream->frame_ptr_bk, VO_INDEX_ENC_AMRWB);
gData->stream->frame_ptr_bk = NULL;
}
mem_free(pMemOP, gData->stream, VO_INDEX_ENC_AMRWB);
gData->stream = NULL;
}
wb_vad_exit(&(((Coder_State *) gData)->vadSt), pMemOP);
dtx_enc_exit(&(((Coder_State *) gData)->dtx_encSt), pMemOP);
mem_free(pMemOP, hCodec, VO_INDEX_ENC_AMRWB);
hCodec = NULL;
}
return VO_ERR_NONE;
}
/********************************************************************************
*
* Brief: voGetAMRWBEncAPI gets the API handle of the codec
*
********************************************************************************/
VO_S32 VO_API voGetAMRWBEncAPI(
VO_AUDIO_CODECAPI * pEncHandle /* i/o: Codec handle pointer */
)
{
if(NULL == pEncHandle)
return VO_ERR_INVALID_ARG;
pEncHandle->Init = voAMRWB_Init;
pEncHandle->SetInputData = voAMRWB_SetInputData;
pEncHandle->GetOutputData = voAMRWB_GetOutputData;
pEncHandle->SetParam = voAMRWB_SetParam;
pEncHandle->GetParam = voAMRWB_GetParam;
pEncHandle->Uninit = voAMRWB_Uninit;
return VO_ERR_NONE;
}
#ifdef __cplusplus
}
#endif