Merge branch 'main' into 1386-stereo_downmix_evs_quality_improvements_pha (25b87f3a) · Commits · SA4 / Audio / IVAS BASOP

lib_com/cldfb.c

+54 −20

Original line number	Diff line number	Diff line
		@@ -1101,6 +1101,9 @@ void cldfbSynthesis_ivas_fx(
		Word32 *imagBuffer_fx, / i : imag values Qx*/
		Word32 timeOut_fx, / o : output time domain samples Qx - 1*/
		const Word16 samplesToProcess, /* i : number of processed samples */
		#ifdef OPT_SBA_AVOID_SPAR_RESCALE
		const Word16 shift, /* i : scale for state buffer */
		#endif /* OPT_SBA_AVOID_SPAR_RESCALE */
		HANDLE_CLDFB_FILTER_BANK h_cldfb /* i : filter bank state */
		)
		{
		@@ -1266,6 +1269,10 @@ void cldfbSynthesis_ivas_fx(
		}

		/* synthesis prototype filter */
		#ifdef OPT_SBA_AVOID_SPAR_RESCALE
		IF( 0 == shift )
		{
		#endif /* OPT_SBA_AVOID_SPAR_RESCALE */
		FOR( i = 0; i < L2; i++ )
		{
		accu0 = Madd_32_16( synthesisBuffer_fx[i], Mpy_32_16_1( new_samples_fx[L2 - 1 - i], p_filter[i] ), p_filter_sf ); // Qx - 1
		@@ -1285,6 +1292,33 @@ void cldfbSynthesis_ivas_fx(
		synthesisBuffer_fx[4 * L2 + i] = accu4;
		move32();
		}
		#ifdef OPT_SBA_AVOID_SPAR_RESCALE
		}
		ELSE
		{

		FOR( i = 0; i < L2; i++ )
		{
		Word32 prod = L_shl_sat( Mpy_32_16_1( new_samples_fx[L2 - 1 - i], p_filter_sf ), shift );
		accu0 = Madd_32_16( synthesisBuffer_fx[i], prod, p_filter[i] ); // Qx - 1
		accu1 = Madd_32_16( synthesisBuffer_fx[1 * L2 + i], prod, p_filter[( 1 * L2 + i )] ); // Qx - 1
		accu2 = Madd_32_16( synthesisBuffer_fx[2 * L2 + i], prod, p_filter[( 2 * L2 + i )] ); // Qx - 1
		accu3 = Madd_32_16( synthesisBuffer_fx[3 * L2 + i], prod, p_filter[( 3 * L2 + i )] ); // Qx - 1
		accu4 = Madd_32_16( synthesisBuffer_fx[4 * L2 + i], prod, p_filter[( 4 * L2 + i )] ); // Qx - 1

		synthesisBuffer_fx[i] = accu0;
		move32();
		synthesisBuffer_fx[1 * L2 + i] = accu1;
		move32();
		synthesisBuffer_fx[2 * L2 + i] = accu2;
		move32();
		synthesisBuffer_fx[3 * L2 + i] = accu3;
		move32();
		synthesisBuffer_fx[4 * L2 + i] = accu4;
		move32();
		}
		}
		#endif /* OPT_SBA_AVOID_SPAR_RESCALE */

		FOR( i = 0; i < M1; i++ )
		{

lib_com/cng_exc_fx.c

+2 −3

Original line number	Diff line number	Diff line
		@@ -295,10 +295,9 @@ IF( NE_16( Opt_AMR_WB, 1 ) )
		/* calculate the spectrum of random excitation signal */
		Copy( exc2, fft_io, L_frame );

		Word16 Q_new_inp, mem_decim_size; // TO be removed
		IF( EQ_16( L_frame, L_FRAME16k ) )
		{
		modify_Fs_fx( fft_io, L_FRAME16k, 16000, fft_io, 12800, exc_mem1, 0, &Q_new_inp, &mem_decim_size );
		modify_Fs_fx( fft_io, L_FRAME16k, 16000, fft_io, 12800, exc_mem1, 0 );
		}

		/* fft_rel(fft_io, L_FFT, LOG2_L_FFT); */
		@@ -411,7 +410,7 @@ IF( NE_16( Opt_AMR_WB, 1 ) )

		IF( EQ_16( L_frame, L_FRAME16k ) )
		{
		modify_Fs_fx( fft_io, L_FFT, 12800, fft_io, 16000, exc_mem, 0, &Q_new_inp, &mem_decim_size );
		modify_Fs_fx( fft_io, L_FFT, 12800, fft_io, 16000, exc_mem, 0 );
		}

		/* enr1 = dotp( fft_io, fft_io, L_frame ) / L_frame; */

lib_com/ivas_prot_fx.h

+18 −8

Original line number	Diff line number	Diff line
		@@ -3786,8 +3786,11 @@ ivas_error ivas_osba_dirac_td_binaural_jbm_fx(
		const UWord16 nSamplesAsked, /* i : number of CLDFB slots requested */
		UWord16 nSamplesRendered, / o : number of CLDFB slots rendered */
		UWord16 nSamplesAvailable, / o : number of CLDFB slots still to render */
		Word32 output_fx[], / o : rendered time signal */
		Word32 output_fx[] / o : rendered time signal */
		#ifndef OPT_SBA_AVOID_SPAR_RESCALE
		,
		Word16 out_len /Store the length of values in each channel/
		#endif /* OPT_SBA_AVOID_SPAR_RESCALE */
		);

		ivas_error ivas_osba_ism_metadata_dec_fx(
		@@ -5343,15 +5346,22 @@ ivas_error ivas_sba_dec_render_fx(
		const UWord16 nSamplesAsked, /* i : number of CLDFB slots requested */
		UWord16 nSamplesRendered, / o : number of CLDFB slots rendered */
		UWord16 nSamplesAvailableNext, / o : number of CLDFB slots still to render */
		Word32 output_fx[], / o : rendered time signal Q11*/
		Word32 output_fx[] / o : rendered time signal Q11*/
		#ifndef OPT_SBA_AVOID_SPAR_RESCALE
		,
		Word16 out_len /Store the length of values in each channel/
		#endif /* OPT_SBA_AVOID_SPAR_RESCALE */
		);

		void ivas_spar_dec_upmixer_sf_fx(
		Decoder_Struct st_ivas, / i/o: IVAS decoder handle */
		Word32 output_fx[], / o : output audio channels */
		const Word16 nchan_internal, /* i : number of internal channels */
		Word16 out_len );
		const Word16 nchan_internal /* i : number of internal channels */
		#ifndef OPT_SBA_AVOID_SPAR_RESCALE
		,
		Word16 out_len
		#endif /* OPT_SBA_AVOID_SPAR_RESCALE */
		);

		ivas_error ivas_spar_md_enc_open_fx(
		ivas_spar_md_enc_state_t *hMdEnc_in, / i/o: SPAR MD encoder handle */

lib_com/modif_fs_fx.c

+8 −26

Original line number	Diff line number	Diff line
		@@ -330,9 +330,7 @@ Word16 modify_Fs_fx( /* o : length of output Q0 */
		Word16 sigOut_fx[], /* o : decimated signal Q0 */
		const Word32 fout, /* i : frequency of output Q0 */
		Word16 mem_fx[], /* i/o: filter memory Q0 */
		const Word16 nblp, /* i : flag indicating if NB low-pass is applied */
		Word16 *Q_new_inp, // TO be removed
		Word16 *mem_decim_size // TO be removed
		const Word16 nblp /* i : flag indicating if NB low-pass is applied */
		)
		{
		Word16 i;
		@@ -364,17 +362,13 @@ Word16 modify_Fs_fx( /* o : length of output Q0 */
		/-------------------------------------------------------------------
		* Find the resampling configuration
		-------------------------------------------------------------------/
		*Q_new_inp = 0;
		move16();

		/* check if fin and fout are the same */
		IF( EQ_32( fin, fout ) )
		{
		/* just copy the signal_fx and quit */
		Copy( sigIn_fx, sigOut_fx, lg );
		*mem_decim_size = 0;
		*Q_new_inp = 0;
		move16();
		move16();

		return lg;
		}
		ELSE
		@@ -439,8 +433,6 @@ Word16 modify_Fs_fx( /* o : length of output Q0 */
		}

		mem_len = shl( filt_len, 1 );
		*mem_decim_size = mem_len;
		move16();
		signal_fx = signal_tab_fx + 2 * L_FILT_MAX + sub( L_FRAME48k, add( mem_len, lg ) );
		signal_ana_fx = signal_fx;
		mem_len_ana = mem_len;
		@@ -508,7 +500,7 @@ Word16 modify_Fs_fx( /* o : length of output Q0 */
		}
		/* interpolation */

		datastep = shr( div_s( shl( fac_den, 7 ), shl( fac_num, 10 ) ), 12 );
		datastep = shr( div_s( shl( fac_den, 8 ), shl( fac_num, 11 ) ), 12 );
		/* equivalent to 'datastep = fac_den % fac_num' */
		temp_n = i_mult2( datastep, fac_num ); /Q0/
		fracstep = sub( fac_den, temp_n );
		@@ -550,9 +542,6 @@ Word16 modify_Fs_fx( /* o : length of output Q0 */
		FOR( i = 0; i < lg_out; i++ )
		{
		sigOut_fx[i] = round_fx( L_shl( L_mult( sigOut_fx[i], num_den ), 1 ) ); /Q0/
		*Q_new_inp = -1;
		move16();
		move16();
		}
		}
		ELSE
		@@ -562,16 +551,17 @@ Word16 modify_Fs_fx( /* o : length of output Q0 */
		test();
		if ( GT_32( fin, 16000 ) && ( EQ_16( lg_out, L_FRAME ) \|\| EQ_16( lg_out, L_FRAME16k ) \|\| EQ_16( lg_out, 512 ) ) )
		{
		#ifdef BASOP_NOGLOB_DECLARE_LOCAL
		num_den = shl_o( num_den, 1, &Overflow );
		//*Q_new_inp = 2;
		#else
		num_den = shl( num_den, 1 );
		#endif
		}
		FOR( i = 0; i < lg_out; i++ )
		{
		sigOut_fx[i] = mult_r( sigOut_fx[i], num_den ); /Q0/
		move16();
		}
		*Q_new_inp = -1;
		move16();
		}
		}
		ELSE
		@@ -579,13 +569,10 @@ Word16 modify_Fs_fx( /* o : length of output Q0 */
		IF( EQ_16( fac_num, 8 ) )
		{
		num_den = 26214;
		move16();
		FOR( i = 0; i < lg_out; i++ )
		{
		sigOut_fx[i] = mult_r( sigOut_fx[i], num_den ); /Q-1/
		move16();
		*Q_new_inp = -2;
		move16();
		}
		}
		ELSE
		@@ -594,9 +581,6 @@ Word16 modify_Fs_fx( /* o : length of output Q0 */
		FOR( i = 0; i < lg_out; i++ )
		{
		sigOut_fx[i] = round_fx( L_mac( L_deposit_h( sigOut_fx[i] ), sigOut_fx[i], num_den ) ); /Q0/
		*Q_new_inp = -1;
		move16();
		move16();
		}
		}
		}
		@@ -607,8 +591,6 @@ Word16 modify_Fs_fx( /* o : length of output Q0 */
		{
		sigOut_fx[i] = mult_r( sigOut_fx[i], 16384 );
		move16(); /Q-1/
		*Q_new_inp = -2;
		move16();
		}
		}
		/* update the filter memory */

lib_com/options.h

+1 −0

Original line number	Diff line number	Diff line
		@@ -174,5 +174,6 @@
		#define FIX_ISSUE_1327 /* Ittiam: Fix for issue 1327: Glitch when stereo is switching from TD to FD*/
		#define NONBE_FIX_1402_WAVEADJUST /* VA: BASOP iisue 1402: fix waveform adjustment decoder PLC */
		#define FIX_ISSUE_1376 /* VA: Fix for issue 1376 (issue with GSC excitation) */
		#define OPT_SBA_AVOID_SPAR_RESCALE /* Optimization made to spar decoder and IGF */
		#define NONBE_FIX_1386_STEREO_DMX_EVS_PHA /* Orange: Fix for stereo DMX / PHA mode : Change the filter taps resolution (Q31->Q30), improve precision for the IR window, for the ILD & IPD smoothing in sub-bands, for the ISD counters and for ICCr. */
		#endif