diff options
Diffstat (limited to 'media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c')
| -rw-r--r-- | media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c | 3354 |
1 files changed, 1679 insertions, 1675 deletions
diff --git a/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c b/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c index df7b9b3..4cafb01 100644 --- a/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c +++ b/media/libstagefright/codecs/amrwbenc/src/voAMRWBEnc.c @@ -19,8 +19,8 @@ * * * Description: Performs the main encoder routine * * Fixed-point C simulation of AMR WB ACELP coding * -* algorithm with 20 msspeech frames for * -* wideband speech signals. * +* algorithm with 20 msspeech frames for * +* wideband speech signals. * * * ************************************************************************/ @@ -51,95 +51,95 @@ 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 + 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 + 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 + 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 */ - ); + 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; + 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; } /*-----------------------------------------------------------------* @@ -149,1176 +149,1180 @@ void Reset_encoder(void *st, Word16 reset_all) * * *-----------------------------------------------------------------*/ 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 */ - ) + 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); + /* 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); + Scale_sig_opt(buf, L_FRAME, 1 - Q_new); #else - Scale_sig(buf, L_FRAME, 1 - Q_new); + 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); + 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); + Residu_opt(Ap, &speech[i_subfr], &wsp[i_subfr], L_SUBFR); #else - Residu(Ap, &speech[i_subfr], &wsp[i_subfr], L_SUBFR); + 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 */ + 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); + Scale_sig_opt(wsp, L_FRAME / OPL_DECIM, shift); #else - Scale_sig(wsp, L_FRAME / OPL_DECIM, shift); + 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 */ + /* 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); + Residu_opt(&A[3 * (M + 1)], speech, exc, L_FRAME); #else - Residu(&A[3 * (M + 1)], speech, exc, L_FRAME); + Residu(&A[3 * (M + 1)], speech, exc, L_FRAME); #endif - for (i = 0; i < L_FRAME; i++) - { - exc2[i] = shr(exc[i], Q_new); - } + 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) - { + 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); + Residu_opt(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR); #else - Residu(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR); + 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]); - } + 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); + Residu_opt(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR); #else - Residu(p_Aq, &speech[i_subfr], &exc[i_subfr], L_SUBFR); + 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); + 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); + Residu_opt(Ap, error + M, xn, L_SUBFR); #else - Residu(Ap, error + M, xn, L_SUBFR); + 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); + 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); + Residu_opt(p_Aq,code + M, cn, L_SUBFR / 2); #else - Residu(p_Aq,code + M, cn, L_SUBFR / 2); + 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() * - *---------------------------------------------------------------*/ + /* 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 */ + 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 */ + 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 */ + /*----------------------------------------------------------------------* + * 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); + pred_lt4_asm(&exc[i_subfr], T0, T0_frac, L_SUBFR + 1); #else - Pred_lt4(&exc[i_subfr], T0, T0_frac, L_SUBFR + 1); + Pred_lt4(&exc[i_subfr], T0, T0_frac, L_SUBFR + 1); #endif - if (*ser_size > NBBITS_9k) - { + if (*ser_size > NBBITS_9k) + { #ifdef ASM_OPT /* asm optimization branch */ - Convolve_asm(&exc[i_subfr], h1, y1, L_SUBFR); + Convolve_asm(&exc[i_subfr], h1, y1, L_SUBFR); #else - Convolve(&exc[i_subfr], h1, y1, L_SUBFR); + 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; - } + 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); + Convolve_asm(code, h1, y2, L_SUBFR); #else - Convolve(code, h1, y2, L_SUBFR); + 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); + 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 */ + 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 */ + 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); + /*-----------------------------------------------------------------* + * - 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); + Scale_sig_opt(exc2, L_SUBFR, shift); #else - Scale_sig(exc2, L_SUBFR, shift); + 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; + 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++) + { + Word32 tmp; + /* code in Q9, gain_pit in Q14 */ + L_tmp = (gain_code * code[i])<<1; + L_tmp = (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 = (code2[i] * gain_code)<<1; + L_tmp = (L_tmp << 5); + L_tmp += (exc2[i] * gain_pit)<<1; + L_tmp = (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; } /*-----------------------------------------------------* @@ -1329,225 +1333,225 @@ void coder( *-----------------------------------------------------*/ 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 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); + 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); + 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); + 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); + 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)); + 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)); + 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 */ + 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)); + 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)); + 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 */ + 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)); + tmp = extract_h(Dot_product12_asm(HF, HF, L_SUBFR16k, &exp)); #else - tmp = extract_h(Dot_product12(HF, HF, L_SUBFR16k, &exp)); + 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 */ + 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)); + 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)); + 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); + 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); } /************************************************* @@ -1558,33 +1562,33 @@ static Word16 synthesis( 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; + 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; } /*************************************************************************** @@ -1594,94 +1598,94 @@ int AMR_Enc_Encode(HAMRENC hCodec) ***************************************************************************/ 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 */ - ) + VO_AUDIO_CODINGTYPE vType, /* i: Codec Type ID */ + VO_CODEC_INIT_USERDATA * pUserData /* i: init Parameters */ + ) { - Coder_State *st; - FrameStream *stream; + Coder_State *st; + FrameStream *stream; #ifdef USE_DEAULT_MEM - VO_MEM_OPERATOR voMemoprator; + VO_MEM_OPERATOR voMemoprator; #endif - VO_MEM_OPERATOR *pMemOP; + VO_MEM_OPERATOR *pMemOP; UNUSED(vType); - int interMem = 0; + int interMem = 0; - if(pUserData == NULL || pUserData->memflag != VO_IMF_USERMEMOPERATOR || pUserData->memData == NULL ) - { + 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; + 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; + *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; + } + 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; } /********************************************************************************** @@ -1691,32 +1695,32 @@ VO_U32 VO_API voAMRWB_Init(VO_HANDLE * phCodec, /* o: the audi ***********************************************************************************/ 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 */ - ) + 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; + Coder_State *gData; + FrameStream *stream; - if(NULL == hCodec) - { - return VO_ERR_INVALID_ARG; - } + if(NULL == hCodec) + { + return VO_ERR_INVALID_ARG; + } - gData = (Coder_State *)hCodec; - stream = gData->stream; + gData = (Coder_State *)hCodec; + stream = gData->stream; - if(NULL == pInput || NULL == pInput->Buffer) - { - return VO_ERR_INVALID_ARG; - } + 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; + 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; + return VO_ERR_NONE; } /************************************************************************************** @@ -1726,52 +1730,52 @@ VO_U32 VO_API voAMRWB_SetInputData( ***************************************************************************************/ 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*/ - ) + 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; + 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; } /************************************************************************* @@ -1782,50 +1786,50 @@ VO_U32 VO_API voAMRWB_GetOutputData( 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 */ - ) + 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; + 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; + 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; } /************************************************************************** @@ -1835,52 +1839,52 @@ VO_U32 VO_API voAMRWB_SetParam( ***************************************************************************/ 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 */ - ) + 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; - } + 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; + 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; + 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; } /*********************************************************************************** @@ -1890,32 +1894,32 @@ VO_U32 VO_API voAMRWB_GetParam( *************************************************************************************/ 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; + 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; } /******************************************************************************** @@ -1925,19 +1929,19 @@ VO_U32 VO_API voAMRWB_Uninit(VO_HANDLE hCodec /* i/o: Codec handle poi ********************************************************************************/ VO_S32 VO_API voGetAMRWBEncAPI( - VO_AUDIO_CODECAPI * pEncHandle /* i/o: Codec handle pointer */ - ) + 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; + 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 |