add code from float mr 2540 - To be ported (662244ce) · Commits · SA4 / Audio / IVAS BASOP

lib_com/options.h

+1 −0

Original line number	Diff line number	Diff line
		@@ -115,6 +115,7 @@
		#define FIX_FLOAT_1533_BLEND_SUBFR2 /* FhG: float issue 1533: correct blending in blend_subfr2() */

		#define FIX_2436_CLDFBANAHANDLE_ADRESS /FhG: cldfb handle pointer were handed over in faulty manner/
		//#define FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR /FhG: basop issue 2436 (related to basop 2283): fix garbage output for >1 object OMASA with extrend as ISAR prerenderer /

		/* ##################### End NON-BE switches ########################### */

lib_rend/lib_rend_fx.c

+132 −1

Original line number	Diff line number	Diff line
		@@ -564,6 +564,67 @@ static void accumulate2dArrayToBuffer_fx(

		return;
		}


		#ifdef FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR
		/-------------------------------------------------------------------
		* audio_buffer_td_to_cldfb()
		*
		* Performs CLDFB analysis on contents of td_buffer and mixes the result into cldfb_buffer.
		*
		* This function does not clear the destination buffer before writing.
		*
		* The number of valid CLDFB handles in cldfbAna must be a least equal to the number of channels
		* in the audio buffers.
		-------------------------------------------------------------------/
		static void audio_buffer_td_to_cldfb(
		IVAS_REND_AudioBuffer td_buffer,
		IVAS_REND_AudioBuffer cldfb_buffer,
		int32_t fs,
		HANDLE_CLDFB_FILTER_BANK *cldfbAna )
		{
		int16_t ch, slot_idx;
		float *td_read_ptr;
		float *cldfb_write_ptr;
		int16_t num_bands, num_slots;

		assert( !td_buffer.config.is_cldfb );
		assert( cldfb_buffer.config.is_cldfb );

		assert( td_buffer.config.numChannels == cldfb_buffer.config.numChannels );
		assert( td_buffer.config.numSamplesPerChannel * 2 == cldfb_buffer.config.numSamplesPerChannel );

		td_read_ptr = td_buffer.data;
		cldfb_write_ptr = cldfb_buffer.data;
		num_bands = (int16_t) ( ( CLDFB_NO_CHANNELS_MAX * fs ) / 48000 );
		assert( td_buffer.config.numSamplesPerChannel % num_bands == 0 );
		num_slots = td_buffer.config.numSamplesPerChannel / num_bands;

		for ( ch = 0; ch < cldfb_buffer.config.numChannels; ++ch )
		{
		for ( slot_idx = 0; slot_idx < num_slots; ++slot_idx )
		{
		float re[CLDFB_NO_CHANNELS_MAX];
		float im[CLDFB_NO_CHANNELS_MAX];

		cldfbAnalysis_ts( td_read_ptr,
		re,
		im,
		num_bands,
		cldfbAna[ch] );

		td_read_ptr += num_bands;

		v_add( re, cldfb_write_ptr, cldfb_write_ptr, num_bands );
		cldfb_write_ptr += num_bands;
		v_add( im, cldfb_write_ptr, cldfb_write_ptr, num_bands );
		cldfb_write_ptr += num_bands;
		}
		}
		}
		#endif


		/-------------------------------------------------------------------
		* limitRendererOutput()
		*
		@@ -7104,16 +7165,25 @@ static ivas_error renderIsmToSplitBinaural(
		const SPLIT_REND_WRAPPER *pSplitRendWrapper;
		IVAS_QUATERNION originalHeadRot[MAX_PARAM_SPATIAL_SUBFRAMES];
		IVAS_QUATERNION localHeadRot[MAX_PARAM_SPATIAL_SUBFRAMES];
		Word16 i, ch, slot_idx, num_bands;
		Word16 i;
		#ifndef FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR
		Word16 ch, slot_idx, num_bands;
		Word32 tmpBinaural[MAX_HEAD_ROT_POSES * BINAURAL_CHANNELS][L_FRAME48k];
		Word32 tmpBinaural_CldfbRe[MAX_HEAD_ROT_POSES * BINAURAL_CHANNELS][CLDFB_NO_COL_MAX][CLDFB_NO_CHANNELS_MAX];
		Word32 tmpBinaural_CldfbIm[MAX_HEAD_ROT_POSES * BINAURAL_CHANNELS][CLDFB_NO_COL_MAX][CLDFB_NO_CHANNELS_MAX];
		#endif
		Word16 output_frame = ismInput->base.inputBuffer.config.numSamplesPerChannel;
		COMBINED_ORIENTATION_HANDLE pCombinedOrientationData;
		Word16 ism_md_subframe_update_ext, exp;

		push_wmops( "renderIsmToSplitBinaural" );

		#ifdef FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR
		assert( !outAudio.config.is_cldfb &&
		"ISM renderering only supports TD output. If CLDFB output was requested, "
		"we convert to CLDFB higher up the stack after rendering all ISMs" );
		#endif

		pSplitRendWrapper = ismInput->base.ctx.pSplitRendWrapper;
		pMultiBinPoseData = &pSplitRendWrapper->multiBinPoseData;

		@@ -7195,6 +7265,7 @@ static ivas_error renderIsmToSplitBinaural(
		}
		}

		#ifndef FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR
		IF( NE_16( outAudio.config.is_cldfb, 0 ) )
		{
		/* Perform CLDFB analysis on rendered audio, since the output buffer is CLDFB domain */
		@@ -7246,12 +7317,15 @@ static ivas_error renderIsmToSplitBinaural(
		}
		ELSE
		{
		#endif /FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR/

		/* Copy rendered audio to tmp storage buffer. Copying directly to output would
		* overwrite original audio, which is still needed for rendering next head pose. */
		Copy32( tmpProcessing[0], tmpBinaural[i_mult( 2, pos_idx )], output_frame );
		Copy32( tmpProcessing[1], tmpBinaural[add( i_mult( 2, pos_idx ), 1 )], output_frame );
		#ifndef FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR
		}
		#endif

		/* Overwrite processing buffer with original input audio again */
		copyBufferTo2dArray_fx( ismInput->base.inputBuffer, tmpProcessing );
		@@ -7263,6 +7337,7 @@ static ivas_error renderIsmToSplitBinaural(
		Copy_Quat_fx( &originalHeadRot[i], &pCombinedOrientationData->Quaternions[i] );
		}

		#ifndef FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR
		if ( outAudio.config.is_cldfb )
		{
		#ifdef FIX_2283_ACCU_CLDFB
		@@ -7273,8 +7348,11 @@ static ivas_error renderIsmToSplitBinaural(
		}
		else
		{
		#endif /FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR/
		accumulate2dArrayToBuffer_fx( tmpBinaural, &outAudio );
		#ifndef FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR
		}
		#endif
		pop_wmops();

		/* Encoding to split rendering bitstream done at a higher level */
		@@ -7427,6 +7505,44 @@ static ivas_error renderActiveInputsIsm(
		ivas_error error;
		Word16 input_q = outAudio.q_factor;
		move16();

		#ifdef FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR
		int16_t num_active_inputs;
		IVAS_REND_AudioBuffer work_buffer;
		float tmp_td_binaural[MAX_HEAD_ROT_POSES * BINAURAL_CHANNELS * L_FRAME48k];

		// Early return in case there are no active ISM inputs
		num_active_inputs = 0;
		for ( i = 0, pCurrentInput = hIvasRend->inputsIsm; i < RENDERER_MAX_ISM_INPUTS; ++i, ++pCurrentInput )
		{
		if ( pCurrentInput->base.inConfig != IVAS_AUDIO_CONFIG_INVALID )
		{
		++num_active_inputs;
		}
		}
		if ( num_active_inputs == 0 )
		{
		return IVAS_ERR_OK;
		}

		// By default (TD output), use outAudio as work_buffer. This means we render individual ISMs
		// directly to outAudio, since the `data` member of both these structs is pointing to the same memory.
		// Initializing this way also fixes MSVC's false-positive warning about work_buffer being used
		// without initialization in code below, as well as some WMC tool instrumentation problems.
		work_buffer = outAudio;

		// ISM rendering only supports TD output. If CLDFB output was requested, we first render
		// to TD in a temporary buffer and then convert to CLDFB. We do this by swapping out the
		// underlying memory of work_buffer and modifying its config to TD.
		if ( outAudio.config.is_cldfb )
		{
		work_buffer.config.numSamplesPerChannel /= 2;
		work_buffer.config.is_cldfb = false;
		work_buffer.data = tmp_td_binaural;
		set_zero( tmp_td_binaural, MAX_HEAD_ROT_POSES * BINAURAL_CHANNELS * L_FRAME48k );
		}
		#endif

		FOR( ( i = 0, pCurrentInput = hIvasRend->inputsIsm ); i < RENDERER_MAX_ISM_INPUTS; ( ++i, ++pCurrentInput ) )
		{
		IF( EQ_32( pCurrentInput->base.inConfig, IVAS_AUDIO_CONFIG_INVALID ) )
		@@ -7437,10 +7553,25 @@ static ivas_error renderActiveInputsIsm(

		*outAudio.pq_fact = input_q;
		move16();
		#ifdef FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR
		IF( NE_32( ( error = renderInputIsm( pCurrentInput, hIvasRend->outputConfig, work_buffer ) ), IVAS_ERR_OK ) )
		#else
		IF( NE_32( ( error = renderInputIsm( pCurrentInput, hIvasRend->outputConfig, outAudio ) ), IVAS_ERR_OK ) )
		#endif
		{
		return error;
		}

		#ifdef FIX_BASOP_2436_REUSED_CLDFB_IN_OMASA_SR
		if ( outAudio.config.is_cldfb )
		{
		audio_buffer_td_to_cldfb( work_buffer,
		outAudio,
		hIvasRend->sampleRateOut,
		hIvasRend->splitRendWrapper->hCldfbHandles->cldfbAna );
		}
		#endif

		FOR( Word16 j = 0; j < outAudio.config.numSamplesPerChannel * outAudio.config.numChannels; ++j )
		{
		outAudio.data_fx[j] = L_shl( outAudio.data_fx[j], sub( sub( input_q, 1 ), ( outAudio.pq_fact ) ) ); / Q(input_q - 1) */