dec_main.c
上传用户:zhongxx05
上传日期:2007-06-06
资源大小:33641k
文件大小:27k
- /*
- *===================================================================
- * 3GPP AMR Wideband Floating-point Speech Codec
- *===================================================================
- */
- #include "hlxclib/stdlib.h"
- #include "hlxclib/math.h"
- #include "hlxclib/memory.h"
- #include "hlxclib/string.h"
- #include "typedef.h"
- #include "dec_main.h"
- #include "dec_dtx.h"
- #include "dec_acelp.h"
- #include "dec_gain.h"
- #include "dec_lpc.h"
- #include "dec_util.h"
- #define MAX_16 (Word16)0x7fff
- #define MIN_16 (Word16)0x8000
- #define L_FRAME 256 /* Frame size */
- #define NB_SUBFR 4 /* Number of subframe per frame */
- #define L_SUBFR 64 /* Subframe size */
- #define MODE_7k 0 /* modes */
- #define MODE_9k 1
- #define MODE_12k 2
- #define MODE_14k 3
- #define MODE_16k 4
- #define MODE_18k 5
- #define MODE_20k 6
- #define MODE_23k 7
- #define MODE_24k 8
- #define RX_SPEECH_PROBABLY_DEGRADED 1 /* rx types */
- #define RX_SPEECH_LOST 2
- #define RX_SPEECH_BAD 3
- #define RX_NO_DATA 7
- #define Q_MAX 8 /* scaling max for signal */
- #define PIT_SHARP 27853 /* pitch sharpening factor = 0.85 Q15 */
- #define PIT_MIN 34 /* Minimum pitch lag with resolution 1/4 */
- #define PIT_FR2 128 /* Minimum pitch lag with resolution 1/2 */
- #define PIT_FR1_9b 160 /* Minimum pitch lag with resolution 1 */
- #define PIT_FR1_8b 92 /* Minimum pitch lag with resolution 1 */
- extern const Word16 D_ROM_isp[];
- extern const Word16 D_ROM_isf[];
- extern const Word16 D_ROM_interpol_frac[];
- #ifdef WIN32
- #pragma warning( disable : 4310)
- #endif
- /*
- * Decoder_reset
- *
- * Parameters:
- * st I/O: pointer to state structure
- * reset_all I: perform full reset
- *
- * Function:
- * Initialisation of variables for the decoder section.
- *
- *
- * Returns:
- * void
- */
- void D_MAIN_reset(void *st, Word16 reset_all)
- {
- Word32 i;
- Decoder_State *dec_state;
- dec_state = (Decoder_State*)st;
- memset(dec_state->mem_exc, 0, (PIT_MAX + L_INTERPOL) * sizeof(Word16));
- memset(dec_state->mem_isf_q, 0, M * sizeof(Word16));
- dec_state->mem_T0_frac = 0; /* old pitch value = 64.0 */
- dec_state->mem_T0 = 64;
- dec_state->mem_first_frame = 1;
- dec_state->mem_gc_thres = 0;
- dec_state->mem_tilt_code = 0;
- memset(dec_state->mem_ph_disp, 0, 8 * sizeof(Word16));
- /* scaling memories for excitation */
- dec_state->mem_q = Q_MAX;
- dec_state->mem_subfr_q[3] = Q_MAX;
- dec_state->mem_subfr_q[2] = Q_MAX;
- dec_state->mem_subfr_q[1] = Q_MAX;
- dec_state->mem_subfr_q[0] = Q_MAX;
- if(reset_all != 0)
- {
- /* routines initialization */
- D_GAIN_init(dec_state->mem_gain);
- memset(dec_state->mem_oversamp, 0, (2 * 12) * sizeof(Word16));
- memset(dec_state->mem_sig_out, 0, 6 * sizeof(Word16));
- memset(dec_state->mem_hf, 0, (31 - 1) * sizeof(Word16));
- memset(dec_state->mem_hf3, 0, (31 - 1) * sizeof(Word16));
- memset(dec_state->mem_hp400, 0, 6 * sizeof(Word16));
- D_GAIN_lag_concealment_init(dec_state->mem_lag);
- /* isp initialization */
- memcpy(dec_state->mem_isp, D_ROM_isp, M * sizeof(Word16));
- memcpy(dec_state->mem_isf, D_ROM_isf, M * sizeof(Word16));
- for(i = 0; i < L_MEANBUF; i++)
- {
- memcpy(&dec_state->mem_isf_buf[i * M], D_ROM_isf, M * sizeof(Word16));
- }
- /* variable initialization */
- dec_state->mem_deemph = 0;
- dec_state->mem_seed = 21845; /* init random with 21845 */
- dec_state->mem_seed2 = 21845;
- dec_state->mem_seed3 = 21845;
- dec_state->mem_state = 0;
- dec_state->mem_bfi = 0;
- /* Static vectors to zero */
- memset(dec_state->mem_syn_hf, 0, M16k * sizeof(Word16));
- memset(dec_state->mem_syn_hi, 0, M * sizeof(Word16));
- memset(dec_state->mem_syn_lo, 0, M * sizeof(Word16));
- D_DTX_reset(dec_state->dtx_decSt, D_ROM_isf);
- dec_state->mem_vad_hist = 0;
- }
- return;
- }
- /*
- * Decoder_init
- *
- * Parameters:
- * spd_state O: pointer to state structure
- *
- * Function:
- * Initialization of variables for the decoder section.
- * Memory allocation.
- *
- * Returns:
- * return zero if succesful
- */
- Word32 D_MAIN_init(void **spd_state)
- {
- /* Decoder states */
- Decoder_State *st;
- *spd_state = NULL;
- /*
- * Memory allocation for coder state.
- */
- if((st = (Decoder_State*)malloc(sizeof(Decoder_State))) == NULL)
- {
- return(-1);
- }
- st->dtx_decSt = NULL;
- D_DTX_init(&st->dtx_decSt, D_ROM_isf);
- D_MAIN_reset((void *)st, 1);
- *spd_state = (void *)st;
- return(0);
- }
- /*
- * Decoder_close
- *
- * Parameters:
- * spd_state I: pointer to state structure
- *
- * Function:
- * Free coder memory.
- *
- * Returns:
- * void
- */
- void D_MAIN_close(void **spd_state)
- {
- D_DTX_exit(&(((Decoder_State *)(*spd_state))->dtx_decSt));
- free(*spd_state);
- return;
- }
- /*
- * Decoder_exe
- *
- * Parameters:
- * mode I: used mode
- * prms I: parameter vector
- * synth_out O: synthesis speech
- * spe_state B: state structure
- * frame_type I: received frame type
- *
- * Function:
- * Main decoder routine.
- *
- * Returns:
- * 0 if successful
- */
- Word32 D_MAIN_decode(Word16 mode, Word16 prms[], Word16 synth16k[],
- void *spd_state, UWord8 frame_type)
- {
- Word32 code2[L_SUBFR]; /* algebraic codevector */
- Word32 L_tmp, L_tmp2, L_gain_code, L_stab_fac;
- Word32 i, j, i_subfr, pit_flag;
- Word32 T0, T0_frac, T0_max, select, T0_min = 0;
- Word16 exc2[L_FRAME]; /* excitation vector */
- Word16 Aq[NB_SUBFR * (M + 1)]; /* A(z) quantized for the 4 subframes */
- Word16 code[L_SUBFR]; /* algebraic codevector */
- Word16 excp[L_SUBFR]; /* excitation vector */
- Word16 HfIsf[M16k];
- Word16 ispnew[M]; /* immittance spectral pairs at 4nd sfr*/
- Word16 isf[M]; /* ISF (frequency domain) at 4nd sfr */
- Word16 isf_tmp[M]; /* ISF tmp */
- Word16 ind[8]; /* quantization indices */
- Word16 index, fac, voice_fac, max, Q_new = 0;
- Word16 gain_pit, gain_code, gain_code_lo, tmp;
- Word16 corr_gain = 0;
- UWord16 pit_sharp = 0;
- Word16 *exc; /* Excitation vector */
- Word16 *p_Aq; /* ptr to A(z) for the 4 subframes */
- Word16 *p_isf; /* prt to isf */
- Decoder_State *st; /* Decoder states */
- UWord8 newDTXState, bfi, unusable_frame;
- UWord8 vad_flag;
- st = (Decoder_State*)spd_state;
- /* find the new DTX state SPEECH OR DTX */
- newDTXState = D_DTX_rx_handler(st->dtx_decSt, frame_type);
- if(newDTXState != SPEECH)
- {
- D_DTX_exe(st->dtx_decSt, exc2, newDTXState, isf, &prms);
- }
- /* SPEECH action state machine */
- if((frame_type == RX_SPEECH_BAD) | (frame_type == RX_NO_DATA) |
- (frame_type == RX_SPEECH_PROBABLY_DEGRADED))
- {
- /* bfi for all index, bits are not usable */
- bfi = 1;
- unusable_frame = 0;
- }
- else if(frame_type == RX_SPEECH_LOST)
- {
- /* bfi only for lsf, gains and pitch period */
- bfi = 1;
- unusable_frame = 1;
- }
- else
- {
- bfi = 0;
- unusable_frame = 0;
- }
- if(bfi != 0)
- {
- st->mem_state = (UWord8)(st->mem_state + 1);
- if(st->mem_state > 6)
- {
- st->mem_state = 6;
- }
- }
- else
- {
- st->mem_state = (UWord8)(st->mem_state >> 1);
- }
- /*
- * If this frame is the first speech frame after CNI period,
- * set the BFH state machine to an appropriate state depending
- * on whether there was DTX muting before start of speech or not
- * If there was DTX muting, the first speech frame is muted.
- * If there was no DTX muting, the first speech frame is not
- * muted. The BFH state machine starts from state 5, however, to
- * keep the audible noise resulting from a SID frame which is
- * erroneously interpreted as a good speech frame as small as
- * possible (the decoder output in this case is quickly muted)
- */
- if(st->dtx_decSt->mem_dtx_global_state == DTX)
- {
- st->mem_state = 5;
- st->mem_bfi = 0;
- }
- else if(st->dtx_decSt->mem_dtx_global_state == D_DTX_MUTE)
- {
- st->mem_state = 5;
- st->mem_bfi = 1;
- }
- if(newDTXState == SPEECH)
- {
- vad_flag = (UWord8)(*prms++);
- if(bfi == 0)
- {
- if(vad_flag == 0)
- {
- st->mem_vad_hist = (Word16)(st->mem_vad_hist + 1);
- if(st->mem_vad_hist > 32767)
- {
- st->mem_vad_hist = 32767;
- }
- }
- else
- {
- st->mem_vad_hist = 0;
- }
- }
- }
- /*
- * DTX-CNG
- */
- if(newDTXState != SPEECH) /* CNG mode */
- {
- /*
- * increase slightly energy of noise below 200 Hz
- * Convert ISFs to the cosine domain
- */
- D_LPC_isf_isp_conversion(isf, ispnew, M);
- D_LPC_isp_a_conversion(ispnew, Aq, M);
- memcpy(isf_tmp, st->mem_isf, M * sizeof(Word16));
- for(i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
- {
- j = (i_subfr >> 6);
- for(i = 0; i < M; i++)
- {
- L_tmp = (isf_tmp[i] * (32767 - D_ROM_interpol_frac[j])) << 1;
- L_tmp = L_tmp + ((isf[i] * D_ROM_interpol_frac[j]) << 1);
- HfIsf[i] = (Word16)((L_tmp + 0x8000) >> 16);
- }
- D_UTIL_dec_synthesis(Aq, &exc2[i_subfr], 0, &synth16k[i_subfr * 5 /4],
- (Word16) 1, HfIsf, mode, newDTXState, bfi, st);
- }
- /* reset speech coder memories */
- D_MAIN_reset(st, 0);
- memcpy(st->mem_isf, isf, M * sizeof(Word16));
- st->mem_bfi = bfi;
- st->dtx_decSt->mem_dtx_global_state = (UWord8)newDTXState;
- return(0);
- }
- /*
- * ACELP
- */
- exc = st->mem_exc + PIT_MAX + L_INTERPOL;
- /* Decode the ISFs */
- if(mode <= MODE_7k)
- {
- ind[0] = *prms++;
- ind[1] = *prms++;
- ind[2] = *prms++;
- ind[3] = *prms++;
- ind[4] = *prms++;
- D_LPC_isf_2s3s_decode(ind, isf, st->mem_isf_q, st->mem_isf,
- st->mem_isf_buf, bfi);
- }
- else
- {
- ind[0] = *prms++;
- ind[1] = *prms++;
- ind[2] = *prms++;
- ind[3] = *prms++;
- ind[4] = *prms++;
- ind[5] = *prms++;
- ind[6] = *prms++;
- D_LPC_isf_2s5s_decode(ind, isf, st->mem_isf_q, st->mem_isf,
- st->mem_isf_buf, bfi);
- }
- /* Convert ISFs to the cosine domain */
- D_LPC_isf_isp_conversion(isf, ispnew, M);
- if(st->mem_first_frame != 0)
- {
- st->mem_first_frame = 0;
- memcpy(st->mem_isp, ispnew, M * sizeof(Word16));
- }
- /* Find the interpolated ISPs and convert to a[] for all subframes */
- D_LPC_int_isp_find(st->mem_isp, ispnew, D_ROM_interpol_frac, Aq);
- /* update isp memory for the next frame */
- memcpy(st->mem_isp, ispnew, M * sizeof(Word16));
- /* Check stability on isf : distance between old isf and current isf */
- L_tmp = 0;
- p_isf = st->mem_isf;
- for(i = 0; i < M - 1; i++)
- {
- tmp = (Word16)((isf[i] - p_isf[i]));
- L_tmp = L_tmp + (tmp * tmp);
- }
- if(L_tmp < 3276928)
- {
- L_tmp = L_tmp >> 7;
- L_tmp = (L_tmp * 26214) >> 15; /* tmp = L_tmp*0.8/256 */
- L_tmp = 20480 - L_tmp; /* 1.25 - tmp */
- L_stab_fac = L_tmp << 1; /* Q14 -> Q15 with saturation */
- if(L_stab_fac > 0x7FFF)
- {
- L_stab_fac = 0x7FFF;
- }
- }
- else
- {
- L_stab_fac = 0x0;
- }
- memcpy(isf_tmp, st->mem_isf, M * sizeof(Word16));
- memcpy(st->mem_isf, isf, M * sizeof(Word16));
- /*
- * Loop for every subframe in the analysis frame
- *
- * The subframe size is L_SUBFR and the loop is repeated L_FRAME/L_SUBFR
- * times
- * - decode the pitch delay and filter mode
- * - decode algebraic code
- * - decode pitch and codebook gains
- * - find voicing factor and tilt of code for next subframe
- * - find the excitation and compute synthesis speech
- */
- p_Aq = Aq; /* pointer to interpolated LPC parameters */
- for(i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR)
- {
- pit_flag = i_subfr;
- if((i_subfr == (2 * L_SUBFR)) & (mode > MODE_7k))
- {
- pit_flag = 0;
- }
- /*
- * - Decode pitch lag
- * Lag indeces received also in case of BFI,
- * so that the parameter pointer stays in sync.
- */
- if(pit_flag == 0)
- {
- if(mode <= MODE_9k)
- {
- index = *prms++;
- if(index < ((PIT_FR1_8b - PIT_MIN) * 2))
- {
- T0 = (PIT_MIN + (index >> 1));
- T0_frac = (index - ((T0 - PIT_MIN) << 1));
- T0_frac = (T0_frac << 1);
- }
- else
- {
- T0 = index + (PIT_FR1_8b - ((PIT_FR1_8b - PIT_MIN) * 2));
- T0_frac = 0;
- }
- }
- else
- {
- index = *prms++;
- if(index < ((PIT_FR2 - PIT_MIN) * 4))
- {
- T0 = PIT_MIN + (index >> 2);
- T0_frac = index - ((T0 - PIT_MIN) << 2);
- }
- else if(index <
- ((((PIT_FR2 - PIT_MIN) * 4) + ((PIT_FR1_9b - PIT_FR2) * 2))))
- {
- index = (Word16)((index - ((PIT_FR2 - PIT_MIN) * 4)));
- T0 = PIT_FR2 + (index >> 1);
- T0_frac = index - ((T0 - PIT_FR2) << 1);
- T0_frac = T0_frac << 1;
- }
- else
- {
- T0 = index + (PIT_FR1_9b - ((PIT_FR2 - PIT_MIN) * 4) -
- ((PIT_FR1_9b - PIT_FR2) * 2));
- T0_frac = 0;
- }
- }
- /* 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 */
- if(mode <= MODE_9k)
- {
- index = *prms++;
- T0 = T0_min + (index >> 1);
- T0_frac = index - ((T0 - T0_min) << 1);
- T0_frac = T0_frac << 1;
- }
- else
- {
- index = *prms++;
- T0 = T0_min + (index >> 2);
- T0_frac = index - ((T0 - T0_min) << 2);
- }
- }
- /* check BFI after pitch lag decoding */
- if(bfi != 0) /* if frame erasure */
- {
- D_GAIN_lag_concealment(&(st->mem_gain[17]), st->mem_lag, &T0,
- &(st->mem_T0), &(st->mem_seed3), unusable_frame);
- T0_frac = 0;
- }
- /*
- * Find the pitch gain, the interpolation filter
- * and the adaptive codebook vector.
- */
- D_GAIN_adaptive_codebook_excitation(&exc[i_subfr], T0, T0_frac);
- if(unusable_frame)
- {
- select = 1;
- }
- else
- {
- if(mode <= MODE_9k)
- {
- select = 0;
- }
- else
- {
- select = *prms++;
- }
- }
- if(select == 0)
- {
- /* find pitch excitation with lp filter */
- for(i = 0; i < L_SUBFR; i++)
- {
- L_tmp = 2949 * exc[i - 1 + i_subfr];
- L_tmp = L_tmp + (10486 * exc[i + i_subfr]);
- L_tmp = L_tmp + (2949 * exc[i + 1 + i_subfr]);
- code[i] = (Word16)((L_tmp + 0x2000) >> 14);
- }
- memcpy(&exc[i_subfr], code, L_SUBFR * sizeof(Word16));
- }
- /*
- * Decode innovative codebook.
- * Add the fixed-gain pitch contribution to code[].
- */
- if(unusable_frame != 0)
- {
- /* the innovative code doesn't need to be scaled (see Q_gain2) */
- for(i = 0; i < L_SUBFR; i++)
- {
- code[i] = (Word16)(D_UTIL_random(&(st->mem_seed)) >> 3);
- }
- }
- else if(mode <= MODE_7k)
- {
- ind[0] = *prms++;
- D_ACELP_decode_2t(ind[0], code);
- }
- else if(mode <= MODE_9k)
- {
- memcpy(ind, prms, 4 * sizeof(Word16));
- prms += 4;
- D_ACELP_decode_4t(ind, 20, code);
- }
- else if(mode <= MODE_12k)
- {
- memcpy(ind, prms, 4 * sizeof(Word16));
- prms += 4;
- D_ACELP_decode_4t(ind, 36, code);
- }
- else if(mode <= MODE_14k)
- {
- memcpy(ind, prms, 4 * sizeof(Word16));
- prms += 4;
- D_ACELP_decode_4t(ind, 44, code);
- }
- else if(mode <= MODE_16k)
- {
- memcpy(ind, prms, 4 * sizeof(Word16));
- prms += 4;
- D_ACELP_decode_4t(ind, 52, code);
- }
- else if(mode <= MODE_18k)
- {
- memcpy(ind, prms, 8 * sizeof(Word16));
- prms += 8;
- D_ACELP_decode_4t(ind, 64, code);
- }
- else if(mode <= MODE_20k)
- {
- memcpy(ind, prms, 8 * sizeof(Word16));
- prms += 8;
- D_ACELP_decode_4t(ind, 72, code);
- }
- else
- {
- memcpy(ind, prms, 8 * sizeof(Word16));
- prms += 8;
- D_ACELP_decode_4t(ind, 88, code);
- }
- tmp = 0;
- D_UTIL_preemph(code, st->mem_tilt_code, L_SUBFR, &tmp);
- L_tmp = T0;
- if(T0_frac > 2)
- {
- L_tmp = L_tmp + 1;
- }
- D_GAIN_pitch_sharpening(code, L_tmp, PIT_SHARP);
- /*
- * Decode codebooks gains.
- */
- index = *prms++; /* codebook gain index */
- if(mode <= MODE_9k)
- {
- D_GAIN_decode(index, 6, code, &gain_pit, &L_gain_code, bfi,
- st->mem_bfi, st->mem_state, unusable_frame, st->mem_vad_hist,
- st->mem_gain);
- }
- else
- {
- D_GAIN_decode(index, 7, code, &gain_pit, &L_gain_code, bfi,
- st->mem_bfi, st->mem_state, unusable_frame, st->mem_vad_hist,
- st->mem_gain);
- }
- /* find best scaling to perform on excitation (Q_new) */
- tmp = st->mem_subfr_q[0];
- for(i = 1; i < 4; i++)
- {
- if(st->mem_subfr_q[i] < tmp)
- {
- tmp = st->mem_subfr_q[i];
- }
- }
- /* limit scaling (Q_new) to Q_MAX */
- if(tmp > Q_MAX)
- {
- tmp = Q_MAX;
- }
- Q_new = 0;
- L_tmp = L_gain_code; /* L_gain_code in Q16 */
- while((L_tmp < 0x08000000L) && (Q_new < tmp))
- {
- L_tmp = (L_tmp << 1);
- Q_new = (Word16)((Q_new + 1));
- }
- if(L_tmp < 0x7FFF7FFF)
- {
- gain_code = (Word16)((L_tmp + 0x8000) >> 16);
- /* scaled gain_code with Qnew */
- }
- else
- {
- gain_code = 32767;
- }
- if(Q_new > st->mem_q)
- {
- D_UTIL_signal_up_scale(exc + i_subfr - (PIT_MAX + L_INTERPOL),
- PIT_MAX + L_INTERPOL + L_SUBFR, (Word16)(Q_new - st->mem_q));
- }
- else
- {
- D_UTIL_signal_down_scale(exc + i_subfr - (PIT_MAX + L_INTERPOL),
- PIT_MAX + L_INTERPOL + L_SUBFR, (Word16)(st->mem_q - Q_new));
- }
- st->mem_q = Q_new;
- /*
- * Update parameters for the next subframe.
- * - tilt of code: 0.0 (unvoiced) to 0.5 (voiced)
- */
- if(bfi == 0)
- {
- /* LTP-Lag history update */
- for(i = 4; i > 0; i--)
- {
- st->mem_lag[i] = st->mem_lag[i - 1];
- }
- st->mem_lag[0] = (Word16)T0;
- st->mem_T0 = (Word16)T0;
- st->mem_T0_frac = 0; /* Remove fraction in case of BFI */
- }
- /* find voice factor in Q15 (1=voiced, -1=unvoiced) */
- memcpy(exc2, &exc[i_subfr], L_SUBFR * sizeof(Word16));
- D_UTIL_signal_down_scale(exc2, L_SUBFR, 3);
- /* post processing of excitation elements */
- if(mode <= MODE_9k)
- {
- pit_sharp = (Word16)(gain_pit << 1);
- if(pit_sharp > 16384)
- {
- if(pit_sharp > 32767)
- {
- pit_sharp = 32767;
- }
- for(i = 0; i < L_SUBFR; i++)
- {
- L_tmp = (exc2[i] * pit_sharp) >> 15;
- L_tmp = L_tmp * gain_pit;
- excp[i] = (Word16)((L_tmp + 0x8000) >> 16);
- }
- }
- }
- voice_fac = D_GAIN_find_voice_factor(exc2, -3, gain_pit, code, gain_code,
- L_SUBFR);
- /* tilt of code for next subframe: 0.5=voiced, 0=unvoiced */
- st->mem_tilt_code = (Word16)((voice_fac >> 2) + 8192);
- /*
- * Find the total excitation.
- * Find synthesis speech corresponding to exc[].
- * Find maximum value of excitation for next scaling
- */
- memcpy(exc2, &exc[i_subfr], L_SUBFR * sizeof(Word16));
- max = 1;
- for(i = 0; i < L_SUBFR; i++)
- {
- L_tmp = (code[i] * gain_code) << 5;
- L_tmp = L_tmp + (exc[i + i_subfr] * gain_pit);
- L_tmp = (L_tmp + 0x2000) >> 14;
- if((L_tmp > MIN_16) & (L_tmp < 32768))
- {
- exc[i + i_subfr] = (Word16)L_tmp;
- tmp = (Word16)(abs(L_tmp));
- if(tmp > max)
- {
- max = tmp;
- }
- }
- else if(L_tmp > MAX_16)
- {
- exc[i + i_subfr] = MAX_16;
- max = MAX_16;
- }
- else
- {
- exc[i + i_subfr] = MIN_16;
- max = MAX_16;
- }
- }
- /* tmp = scaling possible according to max value of excitation */
- tmp = (Word16)((D_UTIL_norm_s(max) + Q_new) - 1);
- st->mem_subfr_q[3] = st->mem_subfr_q[2];
- st->mem_subfr_q[2] = st->mem_subfr_q[1];
- st->mem_subfr_q[1] = st->mem_subfr_q[0];
- st->mem_subfr_q[0] = tmp;
- /*
- * phase dispersion to enhance noise in low bit rate
- */
- /* L_gain_code in Q16 */
- D_UTIL_l_extract(L_gain_code, &gain_code, &gain_code_lo);
- if(mode <= MODE_7k)
- {
- j = 0; /* high dispersion for rate <= 7.5 kbit/s */
- }
- else if(mode <= MODE_9k)
- {
- j = 1; /* low dispersion for rate <= 9.6 kbit/s */
- }
- else
- {
- j = 2; /* no dispersion for rate > 9.6 kbit/s */
- }
- D_ACELP_phase_dispersion(gain_code, gain_pit, code, (Word16)j,
- st->mem_ph_disp);
- /*
- * 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.
- */
- L_tmp = 16384 - (voice_fac >> 1); /* 1=unvoiced, 0=voiced */
- fac = (Word16)((L_stab_fac * L_tmp) >> 15);
- L_tmp = L_gain_code;
- if(L_tmp < st->mem_gc_thres)
- {
- L_tmp = (L_tmp + D_UTIL_mpy_32_16(gain_code, gain_code_lo, 6226));
- if(L_tmp > st->mem_gc_thres)
- {
- L_tmp = st->mem_gc_thres;
- }
- }
- else
- {
- L_tmp = D_UTIL_mpy_32_16(gain_code, gain_code_lo, 27536);
- if(L_tmp < st->mem_gc_thres)
- {
- L_tmp = st->mem_gc_thres;
- }
- }
- st->mem_gc_thres = L_tmp;
- L_gain_code =
- D_UTIL_mpy_32_16(gain_code, gain_code_lo, (Word16)(32767 - fac));
- D_UTIL_l_extract(L_tmp, &gain_code, &gain_code_lo);
- L_gain_code =
- L_gain_code + D_UTIL_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.
- */
- L_tmp2 = (voice_fac >> 3) + 4096; /* 0.25=voiced, 0=unvoiced */
- L_tmp = (code[0] << 15) - (code[1] * L_tmp2);
- code2[0] = (L_tmp + 0x4000) >> 15;
- for(i = 1; i < L_SUBFR - 1; i++)
- {
- L_tmp = code[i] << 15;
- L_tmp = L_tmp - (code[i + 1] * L_tmp2);
- L_tmp = L_tmp - (code[i - 1] * L_tmp2);
- code2[i] = (L_tmp + 0x4000) >> 15;
- }
- L_tmp = code[L_SUBFR - 1] << 15;
- L_tmp = L_tmp - (code[L_SUBFR - 2] * L_tmp2);
- code2[L_SUBFR - 1] = (L_tmp + 0x4000) >> 15;
- /* build excitation */
- gain_code = (Word16)(((L_gain_code << Q_new) + 0x8000) >> 16);
- for(i = 0; i < L_SUBFR; i++)
- {
- L_tmp = (code2[i] * gain_code) << 5;
- L_tmp = L_tmp + (exc2[i] * gain_pit);
- L_tmp = (L_tmp + 0x2000) >> 14;
- exc2[i] = D_UTIL_saturate(L_tmp);
- }
- if(mode <= MODE_9k)
- {
- if(pit_sharp > 16384)
- {
- for(i = 0; i < L_SUBFR; i++)
- {
- L_tmp = (excp[i] + exc2[i]);
- excp[i] = D_UTIL_saturate(L_tmp);
- }
- D_GAIN_adaptive_control(exc2, excp, L_SUBFR);
- memcpy(exc2, excp, L_SUBFR * sizeof(Word16));
- }
- }
- if(mode <= MODE_7k)
- {
- j = (i_subfr >> 6);
- for(i = 0; i < M; i++)
- {
- L_tmp = isf_tmp[i] * (32767 - D_ROM_interpol_frac[j]);
- L_tmp = L_tmp + (isf[i] * D_ROM_interpol_frac[j]);
- HfIsf[i] = (Word16)((L_tmp + 0x4000) >> 15);
- }
- }
- else
- {
- memset(st->mem_syn_hf, 0, (M16k - M) * sizeof(Word16));
- }
- if(mode >= MODE_24k)
- {
- corr_gain = *prms++;
- D_UTIL_dec_synthesis(p_Aq, exc2, Q_new, &synth16k[i_subfr * 5 / 4],
- corr_gain, HfIsf, mode, newDTXState, bfi, st);
- }
- else
- {
- D_UTIL_dec_synthesis(p_Aq, exc2, Q_new, &synth16k[i_subfr * 5 / 4], 0,
- HfIsf, mode, newDTXState, bfi, st);
- }
- p_Aq += (M + 1); /* interpolated LPC parameters for next subframe */
- }
- /*
- * Update signal for next frame
- * -> save past of exc[]
- * -> save pitch parameters.
- */
- memmove(st->mem_exc, &st->mem_exc[L_FRAME], (PIT_MAX + L_INTERPOL) * sizeof(Word16));
- D_UTIL_signal_down_scale(exc, L_FRAME, Q_new);
- D_DTX_activity_update(st->dtx_decSt, isf, exc);
- st->dtx_decSt->mem_dtx_global_state = (UWord8)newDTXState;
- st->mem_bfi = bfi;
- return(0);
- }