| /* |
| ** 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" |
| |
| #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++) |
| L_tmp += (exc2[i] * exc2[i])<<1; |
| 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 -= *vo_p1++ * *vo_p2--; |
| L_tmp -= *vo_p1++ * *vo_p2--; |
| L_tmp -= *vo_p1++ * *vo_p2--; |
| 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 += *vo_p0 * *vo_p0; |
| vo_p0++; |
| L_tmp -= *vo_p1 * *vo_p1; |
| vo_p1++; |
| L_tmp += *vo_p0 * *vo_p0; |
| vo_p0++; |
| 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 = (gain_code * y2[L_SUBFR - 1])<<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++) |
| { |
| /* code in Q9, gain_pit in Q14 */ |
| L_tmp = (gain_code * code[i])<<1; |
| L_tmp = (L_tmp << 5); |
| L_tmp += (exc[i + i_subfr] * gain_pit)<<1; |
| 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 = (code2[i] * gain_code)<<1; |
| L_tmp = (L_tmp << 5); |
| L_tmp += (exc2[i] * gain_pit)<<1; |
| L_tmp = (L_tmp << 1); |
| exc2[i] = vo_round(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; |
| 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 || 0 > pInput->Length) |
| { |
| 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 |