Loading lib_com/enhancer_fx.c +240 −1 Original line number Diff line number Diff line Loading @@ -520,7 +520,7 @@ void enhancer_ivas_fx( { /* tmp = 0.150 * (1.0 + voice_fac) */ /* 0.30=voiced, 0=unvoiced */ tmp = mac_r( 0x10000000L, voice_fac, 4915 ); /*Q15 */ tmp = mac_r( 0x13333333L, voice_fac, 4915 ); /*Q15 */ } ELSE { Loading Loading @@ -581,6 +581,245 @@ void enhancer_ivas_fx( } } void enhancer_ivas_fx2( const Word32 core_brate, /* i : decoder bitrate */ const Word16 Opt_AMR_WB, /* i : flag indicating AMR-WB IO mode */ const Word16 coder_type, /* i : coder type */ const Word16 i_subfr, /* i : subframe number */ const Word16 L_frame, /* i : frame size */ const Word16 voice_fac, /* i : subframe voicing estimation Q15 */ const Word16 stab_fac, /* i : LP filter stablility measure Q15 */ Word32 norm_gain_code, /* i : normalised innovative cb. gain Q16 */ const Word16 gain_inov, /* i : gain of the unscaled innovation Q12 */ Word32 *gc_threshold, /* i/o: gain code threshold Q16 */ Word16 *code, /* i/o: innovation Q12 */ Word16 *exc2, /* i/o: adapt. excitation/total exc. Q_exc*/ const Word16 gain_pit, /* i : quantized pitch gain Q14 */ struct dispMem_fx *dm_fx, /* i/o: phase dispersion algorithm memory */ const Word16 Q_exc /* i : Q of the excitation */ ) { Word16 tmp, fac, *pt_exc2; Word16 i; Word32 L_tmp; Word16 gain_code_hi; Word16 pit_sharp, tmp16; Word16 excp[L_SUBFR], sc; pit_sharp = gain_pit; move16(); /* to remove gcc warning */ pt_exc2 = exc2 + i_subfr; /*------------------------------------------------------------* * Phase dispersion to enhance noise at low bit rate *------------------------------------------------------------*/ i = 2; move16(); /* no dispersion */ IF( Opt_AMR_WB ) { IF( LE_32( core_brate, ACELP_6k60 ) ) { i = 0; move16(); /* high dispersion */ } ELSE if ( LE_32( core_brate, ACELP_8k85 ) ) { i = 1; move16(); /* low dispersion */ } } ELSE IF( NE_16( coder_type, UNVOICED ) ) { test(); test(); test(); test(); IF( LE_32( core_brate, ACELP_7k20 ) ) { i = 0; move16(); /* high dispersion */ } ELSE if ( ( EQ_16( coder_type, GENERIC ) || EQ_16( coder_type, TRANSITION ) || EQ_16( coder_type, AUDIO ) || coder_type == INACTIVE ) && LE_32( core_brate, ACELP_9k60 ) ) { i = 1; move16(); /* low dispersion */ } } phase_dispersion_fx( norm_gain_code, gain_pit, code, i, dm_fx ); /*------------------------------------------------------------ * 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 decreases by 3 dB noise energy variation. *-----------------------------------------------------------*/ /* tmp = 0.5f * (1.0f - voice_fac) */ #ifdef BASOP_NOGLOB tmp = msu_r_sat( 0x40000000, voice_fac, 16384 ); /*Q15 */ /* 1=unvoiced, 0=voiced */ #else tmp = msu_r( 0x40000000, voice_fac, 16384 ); /*Q15 */ /* 1=unvoiced, 0=voiced */ #endif /* fac = stab_fac * tmp */ fac = mult( stab_fac, tmp ); /*Q15*/ IF( LT_32( norm_gain_code, *gc_threshold ) ) { L_tmp = Madd_32_16( norm_gain_code, norm_gain_code, 6226 ); /*Q16 */ L_tmp = L_min( L_tmp, *gc_threshold ); /*Q16 */ } ELSE { L_tmp = Mult_32_16( norm_gain_code, 27536 ); /*Q16 */ L_tmp = L_max( L_tmp, *gc_threshold ); /*Q16 */ } *gc_threshold = L_tmp; move32(); /*Q16 */ /* gain_code = (fac * tmp) + (1.0 - fac) * gain_code ==> fac * (tmp - gain_code) + gain_code */ L_tmp = L_sub( L_tmp, norm_gain_code ); /*Q16 */ norm_gain_code = Madd_32_16( norm_gain_code, L_tmp, fac ); /*Q16 */ /* gain_code *= gain_inov - Inverse the normalization */ L_tmp = Mult_32_16( norm_gain_code, gain_inov ); /*Q13*/ /* gain_inov in Q12 */ sc = 6; move16(); gain_code_hi = round_fx( L_shl( L_tmp, add( Q_exc, 3 ) ) ); /* in Q_exc */ /*------------------------------------------------------------* * pitch enhancer * * - Enhance excitation on voiced. (HP filtering of code) * On voiced signal, filtering of code by a smooth fir HP * filter to decrease energy of code at low frequency. *------------------------------------------------------------*/ test(); IF( !Opt_AMR_WB && EQ_16( coder_type, UNVOICED ) ) { /* Copy(code, exc2, L_SUBFR) */ FOR( i = 0; i < L_SUBFR; i++ ) { pt_exc2[i] = round_fx( L_shl( L_mult( gain_code_hi, code[i] ), sc ) ); /*Q0 */ /* code in Q12 (Q9 for encoder) */ move16(); } } ELSE { test(); test(); IF( Opt_AMR_WB && ( EQ_32( core_brate, ACELP_8k85 ) || EQ_32( core_brate, ACELP_6k60 ) ) ) { #ifdef BASOP_NOGLOB pit_sharp = shl_sat( gain_pit, 1 ); /* saturation can occur here Q14 -> Q15 */ #else pit_sharp = shl( gain_pit, 1 ); /* saturation can occur here Q14 -> Q15 */ #endif /* saturation takes care of "if (pit_sharp > 1.0) { pit_sharp=1.0; }" */ IF( GT_16( pit_sharp, 16384 ) ) { tmp16 = mult( pit_sharp, 8192 ); FOR( i = 0; i < L_SUBFR; i++ ) { /* excp[i] = pt_exc2[i] * pit_sharp * 0.25 */ excp[i] = mult_r( pt_exc2[i], tmp16 ); move16(); } } } IF( EQ_16( L_frame, L_FRAME16k ) ) { /* tmp = 0.150 * (1.0 + voice_fac) */ /* 0.30=voiced, 0=unvoiced */ tmp = mac_r( 0x13333333L, voice_fac, 4915 ); /*Q15 */ } ELSE { /* tmp = 0.125 * (1.0 + voice_fac) */ /* 0.25=voiced, 0=unvoiced */ tmp = mac_r( 0x10000000L, voice_fac, 4096 ); /*Q15 */ } /*----------------------------------------------------------------- * Do a simple noncasual "sharpening": effectively an FIR * filter with coefs [-tmp 1.0 -tmp] where tmp=0...0.25. * This is applied to code and add_fxed to exc2 *-----------------------------------------------------------------*/ /* pt_exc2[0] += code[0] - tmp * code[1] */ L_tmp = L_deposit_h( code[0] ); /* if Enc :Q9 * Q15 -> Q25 */ L_tmp = L_msu( L_tmp, code[1], tmp ); /* Q12 * Q15 -> Q28 */ #ifdef BASOP_NOGLOB L_tmp = L_shl_sat( L_mult( gain_code_hi, extract_h( L_tmp ) ), sc ); pt_exc2[0] = msu_r_sat( L_tmp, -32768, pt_exc2[0] ); move16(); #else L_tmp = L_shl( L_mult( gain_code_hi, extract_h( L_tmp ) ), sc ); pt_exc2[0] = msu_r( L_tmp, -32768, pt_exc2[0] ); move16(); #endif move16(); /* in Q_exc */ FOR( i = 1; i < L_SUBFR - 1; i++ ) { /* pt_exc2[i] += code[i] - tmp * code[i-1] - tmp * code[i+1] */ L_tmp = L_msu( -32768, code[i], -32768 ); L_tmp = L_msu( L_tmp, code[i + 1], tmp ); #ifdef BASOP_NOGLOB tmp16 = msu_r_sat( L_tmp, code[i - 1], tmp ); L_tmp = L_shl_sat( L_mult( gain_code_hi, tmp16 ), sc ); pt_exc2[i] = msu_r_sat( L_tmp, -32768, pt_exc2[i] ); move16(); #else tmp16 = msu_r( L_tmp, code[i - 1], tmp ); L_tmp = L_shl( L_mult( gain_code_hi, tmp16 ), sc ); pt_exc2[i] = msu_r( L_tmp, -32768, pt_exc2[i] ); #endif move16(); /* in Q_exc */ } /* pt_exc2[L_SUBFR-1] += code[L_SUBFR-1] - tmp * code[L_SUBFR-2] */ L_tmp = L_deposit_h( code[L_SUBFR - 1] ); /*Q28 */ L_tmp = L_msu( L_tmp, code[L_SUBFR - 2], tmp ); /*Q28 */ L_tmp = L_shl( L_mult( gain_code_hi, extract_h( L_tmp ) ), sc ); #ifdef BASOP_NOGLOB pt_exc2[L_SUBFR - 1] = msu_r_sat( L_tmp, -32768, pt_exc2[L_SUBFR - 1] ); move16(); #else pt_exc2[L_SUBFR - 1] = msu_r( L_tmp, -32768, pt_exc2[L_SUBFR - 1] ); move16(); #endif move16(); /* in Q_exc */ test(); test(); IF( Opt_AMR_WB && ( EQ_32( core_brate, ACELP_8k85 ) || EQ_32( core_brate, ACELP_6k60 ) ) ) { IF( GT_16( pit_sharp, 16384 ) ) { FOR( i = 0; i < L_SUBFR; i++ ) { /* excp[i] += pt_exc2[i] */ #ifdef BASOP_NOGLOB excp[i] = add_sat( excp[i], pt_exc2[i] ); #else excp[i] = add( excp[i], pt_exc2[i] ); #endif move16(); } agc2_fx( pt_exc2, excp, L_SUBFR ); Copy( excp, pt_exc2, L_SUBFR ); } } } } /*---------------------------------------------------------* * Enhancement of the excitation signal before synthesis *---------------------------------------------------------*/ Loading lib_com/ivas_prot_fx.h +14 −0 Original line number Diff line number Diff line Loading @@ -1708,6 +1708,19 @@ void synchro_synthesis_fx( // ivas_dirac_output_synthesis_cov void ivas_dirac_dec_output_synthesis_cov_param_mc_collect_slot_fx( #ifdef FIX_835_PARAMMC_BUFFER_VALUES Word32 *RealBuffer_fx, /* i : input channel filter bank samples (real part) */ Word16 RealBuffer_e, /* i : exponent input channel filter bank samples (real part)*/ Word32 *ImagBuffer_fx, /* i : input channel filter bank samples (imaginary part */ Word16 ImagBuffer_e, /* i : exponent input channel filter bank samples (real part)*/ Word32 cx_fx[PARAM_MC_MAX_TRANSPORT_CHANS * PARAM_MC_MAX_TRANSPORT_CHANS], /* o : accumulated input covariance (real part) */ Word16 *cx_e, /* i : exponent for accumulated input covariance (real part) */ Word32 cx_imag_fx[PARAM_MC_MAX_TRANSPORT_CHANS * PARAM_MC_MAX_TRANSPORT_CHANS], /* o : accumulated input covariance (imaginary part) */ Word16 *cx_imag_e, /* i : exponent accumulated input covariance (imag part) */ PARAM_MC_DEC_HANDLE hParamMC, /* i : handle to Parametric MC state */ const Word16 param_band, /* i : parameter band */ const Word16 nchan_in /* i : number of input channels */ #else Word32 *RealBuffer_fx, /* i : input channel filter bank samples (real part) */ Word16 RealBuffer_e, /* i : exponent input channel filter bank samples (real part)*/ Word32 *ImagBuffer_fx, /* i : input channel filter bank samples (imaginary part */ Loading @@ -1718,6 +1731,7 @@ void ivas_dirac_dec_output_synthesis_cov_param_mc_collect_slot_fx( Word16 *cx_imag_e, /* i : exponent accumulated input covariance (imag part) */ PARAM_MC_DEC_HANDLE hParamMC, /* i : handle to Parametric MC state */ const Word16 nchan_in /* i : number of input channels */ #endif ); void configureFdCngDec_ivas_fx( Loading lib_com/ivas_spar_com.c +1 −1 Original line number Diff line number Diff line Loading @@ -3531,7 +3531,7 @@ void ivas_get_spar_md_from_dirac_fx( /*SPAR from DirAC*/ set32_fx( response_avg_fx, 0, MAX_OUTPUT_CHANNELS ); IF( GE_16( n_ts, 1 ) ) IF( GT_16( n_ts, 1 ) ) { ivas_dirac_dec_get_response_fx( extract_l( L_shr( azi_dirac_fx[band][i_ts], Q22 ) ), extract_l( L_shr( ele_dirac_fx[band][i_ts], Q22 ) ), response_avg_fx, order, Q30 ); } Loading lib_com/options.h +2 −0 Original line number Diff line number Diff line Loading @@ -169,6 +169,8 @@ #define FIX_854_HILBERT_SCALING /* VA: reduce lost of precision due to unnecessary scaling, reduce a lot the 2 kHz tone */ #define FIX_856_EXTRACT_L /* VA: Fix undesirable wrap-around */ #define FIX_835_PARAMMC_BUFFER_VALUES /* FhG: issue 835: wide range of buffer values for cx in ParamMC */ /* ################## End DEVELOPMENT switches ######################### */ /* clang-format on */ Loading lib_com/prot_fx.h +33 −0 Original line number Diff line number Diff line Loading @@ -5193,6 +5193,24 @@ void enhancer_ivas_fx( const Word16 Q_exc /* i : Q of the excitation */ ); void enhancer_ivas_fx2( const Word32 core_brate, /* i : decoder bitrate */ const Word16 Opt_AMR_WB, /* i : flag indicating AMR-WB IO mode */ const Word16 coder_type, /* i : coder type */ const Word16 i_subfr, /* i : subframe number */ const Word16 L_frame, /* i : frame size */ const Word16 voice_fac, /* i : subframe voicing estimation Q15 */ const Word16 stab_fac, /* i : LP filter stablility measure Q15 */ Word32 norm_gain_code, /* i : normalised innovative cb. gain Q16 */ const Word16 gain_inov, /* i : gain of the unscaled innovation Q12 */ Word32 *gc_threshold, /* i/o: gain code threshold Q16 */ Word16 *code, /* i/o: innovation Q12 */ Word16 *exc2, /* i/o: adapt. excitation/total exc. Q_exc*/ const Word16 gain_pit, /* i : quantized pitch gain Q14 */ struct dispMem_fx *dm_fx, /* i/o: phase dispersion algorithm memory */ const Word16 Q_exc /* i : Q of the excitation */ ); Word16 E_UTIL_enhancer( Word16 voice_fac, /* i : subframe voicing estimation Q15 */ Word16 stab_fac, /* i : LP filter stability measure Q15 */ Loading Loading @@ -6416,6 +6434,21 @@ void gain_dec_lbr_fx( const Word16 L_subfr /* i : subfr lenght */ ); void gain_dec_lbr_ivas_fx( Decoder_State *st_fx, /* i/o: decoder state structure */ const Word16 coder_type, /* i : coding type */ const Word16 i_subfr, /* i : subframe index */ const Word16 *code_fx, /* i : algebraic excitation Q9 */ Word16 *gain_pit_fx, /* o : quantized pitch gain Q14*/ Word32 *gain_code_fx, /* o : quantized codebook gain Q16*/ Word16 *gain_inov_fx, /* o : gain of the innovation (used for normalization) Q12*/ Word32 *norm_gain_code_fx, /* o : norm. gain of the codebook excitation Q16*/ Word32 gc_mem[], /* i/o: gain_code from previous subframes */ Word16 gp_mem[] /* i/o: gain_pitch from previous subframes */ , const Word16 L_subfr /* i : subfr lenght */ ); void lp_gain_updt_fx( const Word16 i_subfr, /* i : subframe number Q0 */ const Word16 gain_pit, /* i : Decoded gain pitch Q14 */ Loading Loading
lib_com/enhancer_fx.c +240 −1 Original line number Diff line number Diff line Loading @@ -520,7 +520,7 @@ void enhancer_ivas_fx( { /* tmp = 0.150 * (1.0 + voice_fac) */ /* 0.30=voiced, 0=unvoiced */ tmp = mac_r( 0x10000000L, voice_fac, 4915 ); /*Q15 */ tmp = mac_r( 0x13333333L, voice_fac, 4915 ); /*Q15 */ } ELSE { Loading Loading @@ -581,6 +581,245 @@ void enhancer_ivas_fx( } } void enhancer_ivas_fx2( const Word32 core_brate, /* i : decoder bitrate */ const Word16 Opt_AMR_WB, /* i : flag indicating AMR-WB IO mode */ const Word16 coder_type, /* i : coder type */ const Word16 i_subfr, /* i : subframe number */ const Word16 L_frame, /* i : frame size */ const Word16 voice_fac, /* i : subframe voicing estimation Q15 */ const Word16 stab_fac, /* i : LP filter stablility measure Q15 */ Word32 norm_gain_code, /* i : normalised innovative cb. gain Q16 */ const Word16 gain_inov, /* i : gain of the unscaled innovation Q12 */ Word32 *gc_threshold, /* i/o: gain code threshold Q16 */ Word16 *code, /* i/o: innovation Q12 */ Word16 *exc2, /* i/o: adapt. excitation/total exc. Q_exc*/ const Word16 gain_pit, /* i : quantized pitch gain Q14 */ struct dispMem_fx *dm_fx, /* i/o: phase dispersion algorithm memory */ const Word16 Q_exc /* i : Q of the excitation */ ) { Word16 tmp, fac, *pt_exc2; Word16 i; Word32 L_tmp; Word16 gain_code_hi; Word16 pit_sharp, tmp16; Word16 excp[L_SUBFR], sc; pit_sharp = gain_pit; move16(); /* to remove gcc warning */ pt_exc2 = exc2 + i_subfr; /*------------------------------------------------------------* * Phase dispersion to enhance noise at low bit rate *------------------------------------------------------------*/ i = 2; move16(); /* no dispersion */ IF( Opt_AMR_WB ) { IF( LE_32( core_brate, ACELP_6k60 ) ) { i = 0; move16(); /* high dispersion */ } ELSE if ( LE_32( core_brate, ACELP_8k85 ) ) { i = 1; move16(); /* low dispersion */ } } ELSE IF( NE_16( coder_type, UNVOICED ) ) { test(); test(); test(); test(); IF( LE_32( core_brate, ACELP_7k20 ) ) { i = 0; move16(); /* high dispersion */ } ELSE if ( ( EQ_16( coder_type, GENERIC ) || EQ_16( coder_type, TRANSITION ) || EQ_16( coder_type, AUDIO ) || coder_type == INACTIVE ) && LE_32( core_brate, ACELP_9k60 ) ) { i = 1; move16(); /* low dispersion */ } } phase_dispersion_fx( norm_gain_code, gain_pit, code, i, dm_fx ); /*------------------------------------------------------------ * 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 decreases by 3 dB noise energy variation. *-----------------------------------------------------------*/ /* tmp = 0.5f * (1.0f - voice_fac) */ #ifdef BASOP_NOGLOB tmp = msu_r_sat( 0x40000000, voice_fac, 16384 ); /*Q15 */ /* 1=unvoiced, 0=voiced */ #else tmp = msu_r( 0x40000000, voice_fac, 16384 ); /*Q15 */ /* 1=unvoiced, 0=voiced */ #endif /* fac = stab_fac * tmp */ fac = mult( stab_fac, tmp ); /*Q15*/ IF( LT_32( norm_gain_code, *gc_threshold ) ) { L_tmp = Madd_32_16( norm_gain_code, norm_gain_code, 6226 ); /*Q16 */ L_tmp = L_min( L_tmp, *gc_threshold ); /*Q16 */ } ELSE { L_tmp = Mult_32_16( norm_gain_code, 27536 ); /*Q16 */ L_tmp = L_max( L_tmp, *gc_threshold ); /*Q16 */ } *gc_threshold = L_tmp; move32(); /*Q16 */ /* gain_code = (fac * tmp) + (1.0 - fac) * gain_code ==> fac * (tmp - gain_code) + gain_code */ L_tmp = L_sub( L_tmp, norm_gain_code ); /*Q16 */ norm_gain_code = Madd_32_16( norm_gain_code, L_tmp, fac ); /*Q16 */ /* gain_code *= gain_inov - Inverse the normalization */ L_tmp = Mult_32_16( norm_gain_code, gain_inov ); /*Q13*/ /* gain_inov in Q12 */ sc = 6; move16(); gain_code_hi = round_fx( L_shl( L_tmp, add( Q_exc, 3 ) ) ); /* in Q_exc */ /*------------------------------------------------------------* * pitch enhancer * * - Enhance excitation on voiced. (HP filtering of code) * On voiced signal, filtering of code by a smooth fir HP * filter to decrease energy of code at low frequency. *------------------------------------------------------------*/ test(); IF( !Opt_AMR_WB && EQ_16( coder_type, UNVOICED ) ) { /* Copy(code, exc2, L_SUBFR) */ FOR( i = 0; i < L_SUBFR; i++ ) { pt_exc2[i] = round_fx( L_shl( L_mult( gain_code_hi, code[i] ), sc ) ); /*Q0 */ /* code in Q12 (Q9 for encoder) */ move16(); } } ELSE { test(); test(); IF( Opt_AMR_WB && ( EQ_32( core_brate, ACELP_8k85 ) || EQ_32( core_brate, ACELP_6k60 ) ) ) { #ifdef BASOP_NOGLOB pit_sharp = shl_sat( gain_pit, 1 ); /* saturation can occur here Q14 -> Q15 */ #else pit_sharp = shl( gain_pit, 1 ); /* saturation can occur here Q14 -> Q15 */ #endif /* saturation takes care of "if (pit_sharp > 1.0) { pit_sharp=1.0; }" */ IF( GT_16( pit_sharp, 16384 ) ) { tmp16 = mult( pit_sharp, 8192 ); FOR( i = 0; i < L_SUBFR; i++ ) { /* excp[i] = pt_exc2[i] * pit_sharp * 0.25 */ excp[i] = mult_r( pt_exc2[i], tmp16 ); move16(); } } } IF( EQ_16( L_frame, L_FRAME16k ) ) { /* tmp = 0.150 * (1.0 + voice_fac) */ /* 0.30=voiced, 0=unvoiced */ tmp = mac_r( 0x13333333L, voice_fac, 4915 ); /*Q15 */ } ELSE { /* tmp = 0.125 * (1.0 + voice_fac) */ /* 0.25=voiced, 0=unvoiced */ tmp = mac_r( 0x10000000L, voice_fac, 4096 ); /*Q15 */ } /*----------------------------------------------------------------- * Do a simple noncasual "sharpening": effectively an FIR * filter with coefs [-tmp 1.0 -tmp] where tmp=0...0.25. * This is applied to code and add_fxed to exc2 *-----------------------------------------------------------------*/ /* pt_exc2[0] += code[0] - tmp * code[1] */ L_tmp = L_deposit_h( code[0] ); /* if Enc :Q9 * Q15 -> Q25 */ L_tmp = L_msu( L_tmp, code[1], tmp ); /* Q12 * Q15 -> Q28 */ #ifdef BASOP_NOGLOB L_tmp = L_shl_sat( L_mult( gain_code_hi, extract_h( L_tmp ) ), sc ); pt_exc2[0] = msu_r_sat( L_tmp, -32768, pt_exc2[0] ); move16(); #else L_tmp = L_shl( L_mult( gain_code_hi, extract_h( L_tmp ) ), sc ); pt_exc2[0] = msu_r( L_tmp, -32768, pt_exc2[0] ); move16(); #endif move16(); /* in Q_exc */ FOR( i = 1; i < L_SUBFR - 1; i++ ) { /* pt_exc2[i] += code[i] - tmp * code[i-1] - tmp * code[i+1] */ L_tmp = L_msu( -32768, code[i], -32768 ); L_tmp = L_msu( L_tmp, code[i + 1], tmp ); #ifdef BASOP_NOGLOB tmp16 = msu_r_sat( L_tmp, code[i - 1], tmp ); L_tmp = L_shl_sat( L_mult( gain_code_hi, tmp16 ), sc ); pt_exc2[i] = msu_r_sat( L_tmp, -32768, pt_exc2[i] ); move16(); #else tmp16 = msu_r( L_tmp, code[i - 1], tmp ); L_tmp = L_shl( L_mult( gain_code_hi, tmp16 ), sc ); pt_exc2[i] = msu_r( L_tmp, -32768, pt_exc2[i] ); #endif move16(); /* in Q_exc */ } /* pt_exc2[L_SUBFR-1] += code[L_SUBFR-1] - tmp * code[L_SUBFR-2] */ L_tmp = L_deposit_h( code[L_SUBFR - 1] ); /*Q28 */ L_tmp = L_msu( L_tmp, code[L_SUBFR - 2], tmp ); /*Q28 */ L_tmp = L_shl( L_mult( gain_code_hi, extract_h( L_tmp ) ), sc ); #ifdef BASOP_NOGLOB pt_exc2[L_SUBFR - 1] = msu_r_sat( L_tmp, -32768, pt_exc2[L_SUBFR - 1] ); move16(); #else pt_exc2[L_SUBFR - 1] = msu_r( L_tmp, -32768, pt_exc2[L_SUBFR - 1] ); move16(); #endif move16(); /* in Q_exc */ test(); test(); IF( Opt_AMR_WB && ( EQ_32( core_brate, ACELP_8k85 ) || EQ_32( core_brate, ACELP_6k60 ) ) ) { IF( GT_16( pit_sharp, 16384 ) ) { FOR( i = 0; i < L_SUBFR; i++ ) { /* excp[i] += pt_exc2[i] */ #ifdef BASOP_NOGLOB excp[i] = add_sat( excp[i], pt_exc2[i] ); #else excp[i] = add( excp[i], pt_exc2[i] ); #endif move16(); } agc2_fx( pt_exc2, excp, L_SUBFR ); Copy( excp, pt_exc2, L_SUBFR ); } } } } /*---------------------------------------------------------* * Enhancement of the excitation signal before synthesis *---------------------------------------------------------*/ Loading
lib_com/ivas_prot_fx.h +14 −0 Original line number Diff line number Diff line Loading @@ -1708,6 +1708,19 @@ void synchro_synthesis_fx( // ivas_dirac_output_synthesis_cov void ivas_dirac_dec_output_synthesis_cov_param_mc_collect_slot_fx( #ifdef FIX_835_PARAMMC_BUFFER_VALUES Word32 *RealBuffer_fx, /* i : input channel filter bank samples (real part) */ Word16 RealBuffer_e, /* i : exponent input channel filter bank samples (real part)*/ Word32 *ImagBuffer_fx, /* i : input channel filter bank samples (imaginary part */ Word16 ImagBuffer_e, /* i : exponent input channel filter bank samples (real part)*/ Word32 cx_fx[PARAM_MC_MAX_TRANSPORT_CHANS * PARAM_MC_MAX_TRANSPORT_CHANS], /* o : accumulated input covariance (real part) */ Word16 *cx_e, /* i : exponent for accumulated input covariance (real part) */ Word32 cx_imag_fx[PARAM_MC_MAX_TRANSPORT_CHANS * PARAM_MC_MAX_TRANSPORT_CHANS], /* o : accumulated input covariance (imaginary part) */ Word16 *cx_imag_e, /* i : exponent accumulated input covariance (imag part) */ PARAM_MC_DEC_HANDLE hParamMC, /* i : handle to Parametric MC state */ const Word16 param_band, /* i : parameter band */ const Word16 nchan_in /* i : number of input channels */ #else Word32 *RealBuffer_fx, /* i : input channel filter bank samples (real part) */ Word16 RealBuffer_e, /* i : exponent input channel filter bank samples (real part)*/ Word32 *ImagBuffer_fx, /* i : input channel filter bank samples (imaginary part */ Loading @@ -1718,6 +1731,7 @@ void ivas_dirac_dec_output_synthesis_cov_param_mc_collect_slot_fx( Word16 *cx_imag_e, /* i : exponent accumulated input covariance (imag part) */ PARAM_MC_DEC_HANDLE hParamMC, /* i : handle to Parametric MC state */ const Word16 nchan_in /* i : number of input channels */ #endif ); void configureFdCngDec_ivas_fx( Loading
lib_com/ivas_spar_com.c +1 −1 Original line number Diff line number Diff line Loading @@ -3531,7 +3531,7 @@ void ivas_get_spar_md_from_dirac_fx( /*SPAR from DirAC*/ set32_fx( response_avg_fx, 0, MAX_OUTPUT_CHANNELS ); IF( GE_16( n_ts, 1 ) ) IF( GT_16( n_ts, 1 ) ) { ivas_dirac_dec_get_response_fx( extract_l( L_shr( azi_dirac_fx[band][i_ts], Q22 ) ), extract_l( L_shr( ele_dirac_fx[band][i_ts], Q22 ) ), response_avg_fx, order, Q30 ); } Loading
lib_com/options.h +2 −0 Original line number Diff line number Diff line Loading @@ -169,6 +169,8 @@ #define FIX_854_HILBERT_SCALING /* VA: reduce lost of precision due to unnecessary scaling, reduce a lot the 2 kHz tone */ #define FIX_856_EXTRACT_L /* VA: Fix undesirable wrap-around */ #define FIX_835_PARAMMC_BUFFER_VALUES /* FhG: issue 835: wide range of buffer values for cx in ParamMC */ /* ################## End DEVELOPMENT switches ######################### */ /* clang-format on */ Loading
lib_com/prot_fx.h +33 −0 Original line number Diff line number Diff line Loading @@ -5193,6 +5193,24 @@ void enhancer_ivas_fx( const Word16 Q_exc /* i : Q of the excitation */ ); void enhancer_ivas_fx2( const Word32 core_brate, /* i : decoder bitrate */ const Word16 Opt_AMR_WB, /* i : flag indicating AMR-WB IO mode */ const Word16 coder_type, /* i : coder type */ const Word16 i_subfr, /* i : subframe number */ const Word16 L_frame, /* i : frame size */ const Word16 voice_fac, /* i : subframe voicing estimation Q15 */ const Word16 stab_fac, /* i : LP filter stablility measure Q15 */ Word32 norm_gain_code, /* i : normalised innovative cb. gain Q16 */ const Word16 gain_inov, /* i : gain of the unscaled innovation Q12 */ Word32 *gc_threshold, /* i/o: gain code threshold Q16 */ Word16 *code, /* i/o: innovation Q12 */ Word16 *exc2, /* i/o: adapt. excitation/total exc. Q_exc*/ const Word16 gain_pit, /* i : quantized pitch gain Q14 */ struct dispMem_fx *dm_fx, /* i/o: phase dispersion algorithm memory */ const Word16 Q_exc /* i : Q of the excitation */ ); Word16 E_UTIL_enhancer( Word16 voice_fac, /* i : subframe voicing estimation Q15 */ Word16 stab_fac, /* i : LP filter stability measure Q15 */ Loading Loading @@ -6416,6 +6434,21 @@ void gain_dec_lbr_fx( const Word16 L_subfr /* i : subfr lenght */ ); void gain_dec_lbr_ivas_fx( Decoder_State *st_fx, /* i/o: decoder state structure */ const Word16 coder_type, /* i : coding type */ const Word16 i_subfr, /* i : subframe index */ const Word16 *code_fx, /* i : algebraic excitation Q9 */ Word16 *gain_pit_fx, /* o : quantized pitch gain Q14*/ Word32 *gain_code_fx, /* o : quantized codebook gain Q16*/ Word16 *gain_inov_fx, /* o : gain of the innovation (used for normalization) Q12*/ Word32 *norm_gain_code_fx, /* o : norm. gain of the codebook excitation Q16*/ Word32 gc_mem[], /* i/o: gain_code from previous subframes */ Word16 gp_mem[] /* i/o: gain_pitch from previous subframes */ , const Word16 L_subfr /* i : subfr lenght */ ); void lp_gain_updt_fx( const Word16 i_subfr, /* i : subframe number Q0 */ const Word16 gain_pit, /* i : Decoded gain pitch Q14 */ Loading