Merge branch 'complete_path_sba_binaural_ren_fxd' into 'main' (a7a703c5) · Commits · SA4 / Audio / IVAS BASOP

apps/renderer.c

+227 −103

Original line number	Diff line number	Diff line
		@@ -62,7 +62,8 @@

		#define IVAS_MAX16B_FLT 32767.0f
		#define IVAS_MIN16B_FLT ( -32768.0f )

		#define IVAS_MAX16B_FX 32767
		#define IVAS_MIN16B_FX ( -32768 )
		#if !defined( WMOPS )
		static
		#endif
		@@ -404,12 +405,46 @@ static IVAS_AUDIO_CONFIG parseAudioConfig( const char *configString );
		static void convertInputBuffer( const int16_t intBuffer, const int16_t numIntSamplesPerChannel, const int16_t numFloatSamplesPerChannel, const int16_t numChannels, float floatBuffer );

		static void convertOutputBuffer( const float floatBuffer, const int16_t numSamplesPerChannel, const int16_t numChannels, int16_t intBuffer );
		#ifdef IVAS_FLOAT_FIXED
		static void convertOutputBuffer_fx( const Word32 Word32Buffer, const Word16 numSamplesPerChannel, const Word16 numChannels, Word16 intBuffer,Word16 q_factor );
		static void convertInputBuffer_fx( const Word16 intBuffer, const Word16 numIntSamplesPerChannel, const Word16 numFloatSamplesPerChannel, const Word16 numChannels, Word32 Int32Buffer,Word16 in_q_factor );

		#endif

		/------------------------------------------------------------------------------------------
		* Local functions
		------------------------------------------------------------------------------------------/
		#ifdef IVAS_FLOAT_FIXED
		// The function is defined inside library files too with different nsme but same functionality
		static Word16 find_guard_bits( Word32 n )
		{
		return n <= 1 ? 0 : n <= 2 ? 1
		: n <= 4 ? 2
		: n <= 8 ? 3
		: n <= 16 ? 4
		: n <= 32 ? 5
		: n <= 64 ? 6
		: n <= 128 ? 7
		: n <= 256 ? 8
		: n <= 512 ? 9
		: n <= 1024 ? 10
		: 11;
		}
		static IVAS_REND_ReadOnlyAudioBuffer getReadOnlySubBuffer(
		IVAS_REND_AudioBuffer buffer,
		const Word16 chBeginIdx,
		const Word16 numChannels )
		{
		IVAS_REND_ReadOnlyAudioBuffer subBuffer;

		subBuffer.config = buffer.config;
		subBuffer.config.numChannels = numChannels;
		subBuffer.data = buffer.data + subBuffer.config.numSamplesPerChannel * chBeginIdx;
		subBuffer.data_fx = buffer.data_fx + subBuffer.config.numSamplesPerChannel * chBeginIdx;

		return subBuffer;
		}
		#else
		static IVAS_REND_ReadOnlyAudioBuffer getReadOnlySubBuffer(
		IVAS_REND_AudioBuffer buffer,
		const int16_t chBeginIdx,
		@@ -423,15 +458,14 @@ static IVAS_REND_ReadOnlyAudioBuffer getReadOnlySubBuffer(

		return subBuffer;
		}

		#endif

		static int16_t getTotalNumInChannels(
		IVAS_REND_HANDLE hIvasRend,
		IVAS_REND_InputId mcIds[RENDERER_MAX_MC_INPUTS],
		IVAS_REND_InputId ismIds[RENDERER_MAX_ISM_INPUTS],
		IVAS_REND_InputId sbaIds[RENDERER_MAX_SBA_INPUTS],
		IVAS_REND_InputId masaIds[RENDERER_MAX_MASA_INPUTS]
		)
		IVAS_REND_InputId masaIds[RENDERER_MAX_MASA_INPUTS] )
		{
		int16_t totalNumInChannels = 0;
		int16_t i, numInputChannels;
		@@ -517,8 +551,7 @@ static void setupWithSingleFormatInput(
		CmdlnArgs args,
		char *audioFilePath,
		IsmPositionProvider *positionProvider,
		MasaFileReader **masaReaders
		)
		MasaFileReader **masaReaders )
		{
		/* With single-format input, inputFilePath is the path to input audio file. */
		strncpy( audioFilePath, args.inputFilePath, FILENAME_MAX - 1 );
		@@ -577,8 +610,6 @@ static float dBToLin(
		}




		/------------------------------------------------------------------------------------------
		* main()
		*
		@@ -611,6 +642,7 @@ int main(
		int16_t *outInt16Buffer;
		#ifdef IVAS_FLOAT_FIXED
		Word32 *outInt32Buffer;
		Word32 *inInt32Buffer;
		#endif
		float *outFloatBuffer;
		IVAS_REND_AudioBuffer inBuffer;
		@@ -855,7 +887,6 @@ int main(
		fprintf( stderr, "\nIVAS_DEC_FeedRenderConfig failed\n" );
		exit( -1 );
		}

		}

		if ( ( error = IVAS_REND_SetOrientationTrackingMode( hIvasRend, args.orientation_tracking ) ) != IVAS_ERR_OK )
		@@ -1105,6 +1136,7 @@ int main(
		outInt16Buffer = malloc( outBufferSize * sizeof( int16_t ) );
		#ifdef IVAS_FLOAT_FIXED
		outInt32Buffer = malloc( outBufferSize * sizeof( Word32 ) );
		inInt32Buffer = malloc( inBufferSize * sizeof( Word32 ) );
		#endif
		outFloatBuffer = malloc( outBufferSize * sizeof( float ) );

		@@ -1117,6 +1149,7 @@ int main(
		outBuffer.data = outFloatBuffer;
		#ifdef IVAS_FLOAT_FIXED
		outBuffer.data_fx = outInt32Buffer;
		inBuffer.data_fx = inInt32Buffer;
		#endif
		#ifdef WMOPS
		reset_stack();
		@@ -1134,7 +1167,11 @@ int main(
		}

		ObjectPositionBuffer mtdBuffer;

		#ifdef IVAS_FLOAT_FIXED
		outBuffer.pq_fact = &outBuffer.q_factor;
		Word16 subframe_len = (Word16) ( args.sampleRate / ( 200 ) ); // sample rate /FRAMES_PER_SEC * MAX_PARAM_SPATIAL_SUBFRAMES
		Word16 gd_bits = find_guard_bits( subframe_len );
		#endif
		while ( 1 )
		{
		int16_t num_in_channels;
		@@ -1156,7 +1193,10 @@ int main(

		/* Convert from int to float and from interleaved to packed */
		convertInputBuffer( inpInt16Buffer, numSamplesRead, inBuffer.config.numSamplesPerChannel, num_in_channels, inFloatBuffer );

		#ifdef IVAS_FLOAT_FIXED
		*outBuffer.pq_fact = 16 - ( gd_bits + 1 );
		convertInputBuffer_fx( inpInt16Buffer, numSamplesRead, inBuffer.config.numSamplesPerChannel, num_in_channels, inInt32Buffer, *outBuffer.pq_fact );
		#endif
		int16_t num_subframes, sf_idx;
		num_subframes = (int16_t) args.render_framesize;

		@@ -1327,12 +1367,13 @@ int main(
		exit( -1 );
		}
		IVAS_REND_ReadOnlyAudioBuffer tmpBuffer = getReadOnlySubBuffer( inBuffer, (int16_t) args.inConfig.ambisonicsBuses[i].inputChannelIndex, numChannels );

		if ( ( error = IVAS_REND_FeedInputAudio( hIvasRend, sbaIds[i], tmpBuffer ) ) != IVAS_ERR_OK )
		#ifdef IVAS_FLOAT_FIXED
		if ( ( error = IVAS_REND_FeedInputAudio_fx( hIvasRend, sbaIds[i], tmpBuffer ) ) != IVAS_ERR_OK )
		{
		fprintf( stderr, "Error: %s\n", ivas_error_to_string( error ) );
		exit( -1 );
		}
		#endif
		}

		for ( i = 0; i < args.inConfig.numMasaBuses; ++i )
		@@ -1370,8 +1411,6 @@ int main(
		}
		}



		if ( ( error = IVAS_REND_GetSamples( hIvasRend, outBuffer ) ) != IVAS_ERR_OK )
		{
		fprintf( stderr, "Error in getting samples\n" );
		@@ -1380,11 +1419,20 @@ int main(

		int16_t num_out_channels;
		num_out_channels = outBuffer.config.numChannels;

		#ifdef IVAS_FLOAT_FIXED
		/* Convert from float to int and from packed to interleaved.
		* Values in outFloatBuffer are guaranteed to be within range INT16_MIN:INT16_MAX */
		if ( *outBuffer.pq_fact == 0 )
		{
		#endif
		convertOutputBuffer( outFloatBuffer, outBuffer.config.numSamplesPerChannel, num_out_channels, outInt16Buffer );

		#ifdef IVAS_FLOAT_FIXED
		}
		else
		{
		convertOutputBuffer_fx( outInt32Buffer, outBuffer.config.numSamplesPerChannel, num_out_channels, outInt16Buffer, *outBuffer.pq_fact );
		}
		#endif
		if ( delayNumSamples == -1 )
		{
		if ( args.delayCompensationEnabled )
		@@ -1561,6 +1609,7 @@ int main(
		free( outFloatBuffer );
		#ifdef IVAS_FLOAT_FIXED
		free( outInt32Buffer );
		free( inInt32Buffer );
		#endif
		for ( i = 0; i < RENDERER_MAX_MASA_INPUTS; ++i )
		{
		@@ -3251,7 +3300,6 @@ static ivas_error parseLfePanMtxFile(
		return IVAS_ERR_OK;
		}


		/--------------------------------------------------------------------------
		* convertInputBuffer()
		*
		@@ -3264,8 +3312,7 @@ static void convertInputBuffer(
		const int16_t numIntSamplesPerChannel,
		const int16_t numFloatSamplesPerChannel,
		const int16_t numChannels,
		float *floatBuffer
		)
		float *floatBuffer )
		{
		int16_t chnl, smpl, i;

		@@ -3290,20 +3337,98 @@ static void convertInputBuffer(

		return;
		}
		#ifdef IVAS_FLOAT_FIXED

		/--------------------------------------------------------------------------
		* convertInputBuffer()
		*
		* Convert input buffer from WAV/PCM file (int16_t, interleaved) to a format
		* accepted by the renderer (float, packed)
		--------------------------------------------------------------------------/

		static void convertInputBuffer_fx(
		const Word16 *intBuffer,
		const Word16 numIntSamplesPerChannel,
		const Word16 numFloatSamplesPerChannel,
		const Word16 numChannels,
		Word32 *Word32Buffer,
		Word16 in_q_factor)
		{
		Word16 chnl, smpl, i;

		i = 0;

		FOR ( smpl = 0; smpl < numFloatSamplesPerChannel; ++smpl )
		{
		FOR ( chnl = 0; chnl < numChannels; ++chnl )
		{
		IF ( i < numIntSamplesPerChannel )
		{
		Word32Buffer[chnl * numFloatSamplesPerChannel + smpl] = L_shl( (Word32) intBuffer[i], in_q_factor );
		}
		ELSE
		{
		Word32Buffer[chnl * numFloatSamplesPerChannel + smpl] = 0;
		}

		++i;
		}
		}

		return;
		}
		/--------------------------------------------------------------------------
		* convertOutputBuffer()
		*
		* Convert output buffer from the renderer (Word32, packed) to a format ready
		* for writing to a WAV/PCM file (Word16, interleaved)
		--------------------------------------------------------------------------/
		static void convertOutputBuffer_fx(
		const Word32 *Word32Buffer,
		const Word16 numSamplesPerChannel,
		const Word16 numChannels,
		Word16 *intBuffer,
		Word16 out_q )
		{
		Word16 chnl, smpl, i;
		Word32 temp_fx;
		Word32 temp_fx1;
		i = 0;

		FOR ( smpl = 0; smpl < numSamplesPerChannel; ++smpl )
		{
		FOR ( chnl = 0; chnl < numChannels; ++chnl )
		{
		temp_fx = L_add( Word32Buffer[chnl * numSamplesPerChannel + smpl], lshl( 1, ( out_q - 1 ) ) );
		temp_fx1 = L_shr(temp_fx , out_q);
		IF (GT_32( temp_fx1, IVAS_MAX16B_FX ))
		{
		temp_fx1 = IVAS_MAX16B_FX;
		}
		ELSE IF ( LT_32(temp_fx1 , IVAS_MIN16B_FX ))
		{
		temp_fx1 = IVAS_MIN16B_FX;
		}
		intBuffer[i] = (Word16) temp_fx1;

		++i;
		}
		}

		return;
		}
		#endif
		/--------------------------------------------------------------------------
		* convertOutputBuffer()
		*
		* Convert output buffer from the renderer (float, packed) to a format ready
		* for writing to a WAV/PCM file (int16_t, interleaved)
		--------------------------------------------------------------------------/

		static void convertOutputBuffer(
		const float *floatBuffer,
		const int16_t numSamplesPerChannel,
		const int16_t numChannels,
		int16_t *intBuffer
		)
		int16_t *intBuffer )
		{
		int16_t chnl, smpl, i;
		float temp;
		@@ -3332,5 +3457,4 @@ static void convertOutputBuffer(

		return;
		}

		#undef WMC_TOOL_SKIP

lib_rend/ivas_limiter.c

+2 −2

Original line number	Diff line number	Diff line
		@@ -494,10 +494,10 @@ void limiter_process_fx(
		/For powf(200.0f,x)/
		fact = L_Extract_lc( mul, &exp_pow );
		result = extract_l( Pow2( 14, fact ) );
		div32 = divide3232( L_lshl( 1, ( 14 - exp_pow + 1 ) ), L_mult( result, sample_num ) );
		div32 = divide3232( L_shl( 1, ( 14 - exp_pow + 1 ) ), L_mult( result, sample_num ) );
		/For powf(0.01f,x)/
		fact_2 = shr(norm_l(div32 ),1) ;
		release_constant = pow_10( L_lshl( -div32, fact_2 ), &exp_pow_2 );
		release_constant = pow_10( L_shl( -div32, fact_2 ), &exp_pow_2 );
		/* Unoptimized code for reference */
		/* releaseTimeInSeconds = 0.005f * powf(200.f, releaseHeuristic); <-- Map heuristic value (0; 1) exponentially to range (0.005; 1)
		* release_constant = powf( 0.01f, 1.0f / ( releaseTimeInSeconds * sampling_rate ) );

lib_rend/lib_rend.c

+115 −38

Original line number	Diff line number	Diff line
		@@ -224,7 +224,33 @@ typedef struct
		MASA_EXT_REND_HANDLE hMasaExtRend;
		MASA_PREREND_HANDLE hMasaPrerend;
		} input_masa;
		#ifdef IVAS_FLOAT_FIXED
		struct IVAS_REND
		{
		Word32 sampleRateOut;
		IVAS_LIMITER_HANDLE hLimiter;

		input_ism inputsIsm[RENDERER_MAX_ISM_INPUTS];
		input_mc inputsMc[RENDERER_MAX_MC_INPUTS];
		input_sba inputsSba[RENDERER_MAX_SBA_INPUTS];
		input_masa inputsMasa[RENDERER_MAX_MASA_INPUTS];

		AUDIO_CONFIG inputConfig;
		AUDIO_CONFIG outputConfig;
		EFAP_WRAPPER efapOutWrapper;
		IVAS_LSSETUP_CUSTOM_STRUCT customLsOut;

		IVAS_REND_HeadRotData headRotData;

		EXTERNAL_ORIENTATION_HANDLE hExternalOrientationData;
		COMBINED_ORIENTATION_HANDLE hCombinedOrientationData;

		Word8 rendererConfigEnabled;
		RENDER_CONFIG_DATA hRendererConfig; / Renderer config pointer */

		Word16 num_subframes;
		};
		#else
		struct IVAS_REND
		{
		int32_t sampleRateOut;
		@@ -251,7 +277,7 @@ struct IVAS_REND

		int16_t num_subframes;
		};

		#endif

		/-------------------------------------------------------------------
		* Local function prototypes
		@@ -483,7 +509,7 @@ static Word32 limitRendererOutput_fx(
		FOR( i = 0; i < output_frame * num_channels; ++i )
		{

		output[i] = min( max( L_lshl( INT16_MIN, q_factor ), output[i] ), L_lshl( INT16_MAX, q_factor ) );
		output[i] = min( max( L_shl( INT16_MIN, q_factor ), output[i] ), L_shl( INT16_MAX, q_factor ) );
		}

		return numClipping;
		@@ -4029,7 +4055,75 @@ ivas_error IVAS_REND_GetDelay(
		return IVAS_ERR_OK;
		}

		#ifdef IVAS_FLOAT_FIXED
		/-------------------------------------------------------------------
		* IVAS_REND_FeedInputAudio()
		*
		*
		-------------------------------------------------------------------/

		ivas_error IVAS_REND_FeedInputAudio_fx(
		IVAS_REND_HANDLE hIvasRend, /* i/o: Renderer handle */
		const IVAS_REND_InputId inputId, /* i : ID of the input */
		const IVAS_REND_ReadOnlyAudioBuffer inputAudio /* i : buffer with input audio */
		)
		{
		ivas_error error;
		input_base *inputBase;
		int16_t numInputChannels;

		/* Validate function arguments */
		if ( hIvasRend == NULL \|\| inputAudio.data == NULL )
		{
		return IVAS_ERR_UNEXPECTED_NULL_POINTER;
		}

		if ( inputAudio.config.numSamplesPerChannel <= 0 \|\| MAX_BUFFER_LENGTH_PER_CHANNEL < inputAudio.config.numSamplesPerChannel )
		{
		return IVAS_ERROR( IVAS_ERR_INVALID_BUFFER_SIZE, "Buffer size outside of supported range" );
		}

		if ( inputAudio.config.numChannels <= 0 \|\| MAX_INPUT_CHANNELS < inputAudio.config.numChannels )
		{
		return IVAS_ERR_WRONG_NUM_CHANNELS;
		}

		if ( getAudioConfigType( hIvasRend->outputConfig ) == IVAS_REND_AUDIO_CONFIG_TYPE_BINAURAL &&
		inputAudio.config.numSamplesPerChannel * 1000 != ( BINAURAL_RENDERING_FRAME_SIZE_MS * hIvasRend->num_subframes ) * hIvasRend->sampleRateOut )
		{
		return IVAS_ERROR( IVAS_ERR_INVALID_BUFFER_SIZE, "Binaural rendering requires specific frame size" );
		}

		if ( ( error = getInputById( hIvasRend, inputId, (void **) &inputBase ) ) != IVAS_ERR_OK )
		{
		return error;
		}

		if ( ( error = getRendInputNumChannels( inputBase, &numInputChannels ) ) != IVAS_ERR_OK )
		{
		return error;
		}

		if ( ( hIvasRend->outputConfig == IVAS_AUDIO_CONFIG_MASA1 \|\| hIvasRend->outputConfig == IVAS_AUDIO_CONFIG_MASA2 ) && inputBase->inConfig == IVAS_AUDIO_CONFIG_OBA )
		{
		numInputChannels = (int16_t) hIvasRend->inputsIsm[0].total_num_objects;
		}

		if ( numInputChannels != inputAudio.config.numChannels )
		{
		return IVAS_ERR_WRONG_NUM_CHANNELS;
		}

		inputBase->inputBuffer.config = inputAudio.config;

		mvr2r( inputAudio.data, inputBase->inputBuffer.data, inputAudio.config.numSamplesPerChannel * inputAudio.config.numChannels );
		mvr2r_Word32( inputAudio.data_fx, inputBase->inputBuffer.data_fx, inputAudio.config.numSamplesPerChannel * inputAudio.config.numChannels );

		inputBase->numNewSamplesPerChannel = inputAudio.config.numSamplesPerChannel;

		return IVAS_ERR_OK;
		}
		#endif
		/-------------------------------------------------------------------
		* IVAS_REND_FeedInputAudio()
		*
		@@ -4955,7 +5049,6 @@ static ivas_error rotateFrameSba_fx(
		for ( l = 0; l < 3; l++ )
		{
		Rmat[i][l] = ( *hCombinedOrientationData )->Rmat_fx[subframe_idx][i][l]; // Q30

		}
		}
		else
		@@ -6884,10 +6977,9 @@ static ivas_error renderSbaToBinaural(
		}
		IF( hCrend->hReverb != NULL )
		{
		*outAudio.q_fact -= 2;
		*outAudio.pq_fact -= 2;
		}
		accumulate2dArrayToBuffer_fx( output_buffer_fx, &outAudio );

		}

		pop_wmops();
		@@ -7020,7 +7112,7 @@ static ivas_error renderSbaToBinauralRoom(
		{
		for ( Word16 chnl = 0; chnl < tmpRotBuffer.config.numChannels; ++chnl )
		{
		tmpRotBuffer.data_fx[chnl * tmpRotBuffer.config.numSamplesPerChannel + smpl] = (Word32) float_to_fix( tmpRotBuffer.data[chnl * tmpRotBuffer.config.numSamplesPerChannel + smpl], *outAudio.q_fact );
		tmpRotBuffer.data_fx[chnl * tmpRotBuffer.config.numSamplesPerChannel + smpl] = (Word32) float_to_fix( tmpRotBuffer.data[chnl * tmpRotBuffer.config.numSamplesPerChannel + smpl], *outAudio.pq_fact );
		}
		}

		@@ -7035,7 +7127,7 @@ static ivas_error renderSbaToBinauralRoom(
		{
		for ( Word16 chnl = 0; chnl < tmpRotBuffer.config.numChannels; ++chnl )
		{
		tmpRotBuffer.data[chnl * tmpRotBuffer.config.numSamplesPerChannel + smpl] = fix_to_float( tmpRotBuffer.data_fx[chnl * tmpRotBuffer.config.numSamplesPerChannel + smpl], *outAudio.q_fact );
		tmpRotBuffer.data[chnl * tmpRotBuffer.config.numSamplesPerChannel + smpl] = fix_to_float( tmpRotBuffer.data_fx[chnl * tmpRotBuffer.config.numSamplesPerChannel + smpl], *outAudio.pq_fact );
		}
		}
		}
		@@ -7081,7 +7173,7 @@ static ivas_error renderSbaToBinauralRoom(
		{
		for ( j = 0; j < L_FRAME48k; j++ )
		{
		output_fx[i][j] = (Word32) float_to_fix( p_tmpCrendBuffer[i][j], *outAudio.q_fact );
		output_fx[i][j] = (Word32) float_to_fix( p_tmpCrendBuffer[i][j], *outAudio.pq_fact );
		}
		}
		/* call CREND */
		@@ -7093,7 +7185,7 @@ static ivas_error renderSbaToBinauralRoom(
		}
		if ( hCrend->hReverb != NULL )
		{
		*outAudio.q_fact -= 2;
		*outAudio.pq_fact -= 2;
		}
		for ( i = 0; i < nchan_out; i++ )
		{
		@@ -7101,7 +7193,7 @@ static ivas_error renderSbaToBinauralRoom(
		for ( j = 0; j < L_FRAME48k; j++ )
		{

		p_tmpCrendBuffer[i][j] = fix_to_float( output_fx[i][j], *outAudio.q_fact );
		p_tmpCrendBuffer[i][j] = fix_to_float( output_fx[i][j], *outAudio.pq_fact );
		}
		}
		accumulate2dArrayToBuffer( tmpCrendBuffer, &outAudio );
		@@ -7252,7 +7344,7 @@ static ivas_error renderInputSba(

		/* Apply input gain to new audio */
		v_multc_fixed( inAudio.data_fx, sbaInput->base.gain_fx, inAudio.data_fx, inAudio.config.numSamplesPerChannel * inAudio.config.numChannels );
		*outAudio.q_fact -= 1; // to compensate for the qfactor reduction in gain multiplication.
		*outAudio.pq_fact -= 1; // to compensate for the qfactor reduction in gain multiplication.

		/* set combined orientation subframe info to start info */
		ivas_combined_orientation_set_to_start_index( *( sbaInput->base.ctx.pCombinedOrientationData ) );
		@@ -7847,22 +7939,6 @@ static ivas_error getSamplesInternal(
		{
		ivas_error error;
		int16_t numOutChannels;
		Word16 subframe_len, gd_bits,j;
		if ( hIvasRend->inputsSba[0].base.inConfig != IVAS_AUDIO_CONFIG_INVALID )
		{
		subframe_len = (Word16) ( hIvasRend->inputsSba->base.ctx.pOutSampleRate / ( FRAMES_PER_SEC MAX_PARAM_SPATIAL_SUBFRAMES ) );
		gd_bits = find_guarded_bits_fx( subframe_len );
		outAudio.q_fact = &outAudio.q_factor;
		*outAudio.q_fact = 16;
		*outAudio.q_fact -= gd_bits + 1;
		for ( Word16 smpl = 0; smpl < hIvasRend->inputsSba->base.inputBuffer.config.numSamplesPerChannel; ++smpl )
		{
		for ( Word16 chnl = 0; chnl < hIvasRend->inputsSba->base.inputBuffer.config.numChannels; ++chnl )
		{
		hIvasRend->inputsSba->base.inputBuffer.data_fx[chnl * hIvasRend->inputsSba->base.inputBuffer.config.numSamplesPerChannel + smpl] = (Word32) float_to_fix( hIvasRend->inputsSba->base.inputBuffer.data[chnl * hIvasRend->inputsSba->base.inputBuffer.config.numSamplesPerChannel + smpl], *outAudio.q_fact );
		}
		}
		}
		/* Validate function arguments */
		if ( hIvasRend == NULL \|\| outAudio.data == NULL )
		{
		@@ -7952,13 +8028,13 @@ static ivas_error getSamplesInternal(
		{

		#ifndef DISABLE_LIMITER
		Word32 limiter_thresold = L_lshl( IVAS_LIMITER_THRESHOLD, *outAudio.q_fact );
		limitRendererOutput_fx( hIvasRend->hLimiter, outAudio.data_fx, outAudio.config.numSamplesPerChannel, limiter_thresold, *outAudio.q_fact );
		for ( j = 0; j < outAudio.config.numChannels * outAudio.config.numSamplesPerChannel; j++ )
		Word32 limiter_thresold = L_lshl( IVAS_LIMITER_THRESHOLD, *outAudio.pq_fact );
		limitRendererOutput_fx( hIvasRend->hLimiter, outAudio.data_fx, outAudio.config.numSamplesPerChannel, limiter_thresold, *outAudio.pq_fact );
		/* for ( j = 0; j < outAudio.config.numChannels * outAudio.config.numSamplesPerChannel; j++ )
		{
		outAudio.data[j] = fix_to_float( outAudio.data_fx[j], *outAudio.q_fact );
		}

		*/
		#endif
		}
		else
		@@ -7966,6 +8042,7 @@ static ivas_error getSamplesInternal(
		#ifndef DISABLE_LIMITER
		limitRendererOutput( hIvasRend->hLimiter, outAudio.data, outAudio.config.numSamplesPerChannel, IVAS_LIMITER_THRESHOLD );
		#endif
		*outAudio.pq_fact = 0;
		}

		/* update global cominbed orientation start index */

lib_rend/lib_rend.h

+17 −3

Original line number	Diff line number	Diff line
		@@ -77,20 +77,27 @@ typedef struct
		{
		IVAS_REND_AudioBufferConfig config;
		Word16 q_factor;
		Word16 *q_fact;
		Word16 *pq_fact;
		float *data;
		Word32 *data_fx;
		} IVAS_REND_AudioBuffer;

		#endif


		#ifdef IVAS_FLOAT_FIXED
		typedef struct
		{
		IVAS_REND_AudioBufferConfig config;
		const float *data;
		const Word32 *data_fx;
		} IVAS_REND_ReadOnlyAudioBuffer;

		#else
		typedef struct
		{
		IVAS_REND_AudioBufferConfig config;
		const float *data;
		} IVAS_REND_ReadOnlyAudioBuffer;
		#endif
		typedef struct IVAS_REND *IVAS_REND_HANDLE;
		typedef struct IVAS_REND const *IVAS_REND_CONST_HANDLE;

		@@ -201,7 +208,14 @@ ivas_error IVAS_REND_FeedInputAudio(
		const IVAS_REND_InputId inputId, /* i : ID of the input */
		const IVAS_REND_ReadOnlyAudioBuffer inputAudio /* i : buffer with input audio */
		);
		#ifdef IVAS_FLOAT_FIXED

		ivas_error IVAS_REND_FeedInputAudio_fx(
		IVAS_REND_HANDLE hIvasRend, /* i/o: Renderer handle */
		const IVAS_REND_InputId inputId, /* i : ID of the input */
		const IVAS_REND_ReadOnlyAudioBuffer inputAudio /* i : buffer with input audio */
		);
		#endif
		ivas_error IVAS_REND_FeedInputObjectMetadata(
		IVAS_REND_HANDLE hIvasRend, /* i/o: Renderer handle */
		const IVAS_REND_InputId inputId, /* i : ID of the input */