Switch to using float range INT16_MIN:INT16_MAX (a94ea3b8) · Commits · IVAS Codec Public Collaboration / IVAS Codec

apps/renderer.c

+76 −29

Original line number	Diff line number	Diff line
		@@ -293,6 +293,17 @@ static void convert_backslash(
		static void remove_cr(
		char *str );

		static void convertInputBuffer( const int16_t *intBuffer,
		int32_t numIntSamplesPerChannel, /* Number of samples per channel in the int buffer */
		int32_t numFloatSamplesPerChannel, /* Per-channel length of the float buffer. If > numIntSamplesPerChannel, remaining samples will be set to 0. */
		int32_t numChannels,
		float *floatBuffer );

		static void convertOutputBuffer( const float *floatBuffer,
		int32_t numSamplesPerChannel,
		int32_t numChannels,
		int16_t *intBuffer );


		/* ============================================================================ */

		@@ -475,10 +486,8 @@ int32_t main( int32_t argc, char **argv )

		while ( 1 )
		{
		int32_t chnl, smpl;
		int32_t num_in_channels;
		num_in_channels = inBuffer.config.numChannels;
		i = 0;

		/* Read the input data */
		if ( ( error = AudioFileReader_read( audioReader, inpInt16Buffer, inBufferSize, &numSamplesRead ) ) != IVAS_ERR_OK )
		@@ -494,22 +503,7 @@ int32_t main( int32_t argc, char **argv )
		}

		/* Convert from int to float and from interleaved to packed */
		for ( smpl = 0; smpl < frameSize_smpls; ++smpl )
		{
		for ( chnl = 0; chnl < num_in_channels; ++chnl )
		{
		if ( i < numSamplesRead )
		{
		inFloatBuffer[chnl * frameSize_smpls + smpl] = (float) inpInt16Buffer[i] / INT16_MAX;
		}
		else
		{
		inFloatBuffer[chnl * frameSize_smpls + smpl] = 0.f;
		}

		++i;
		}
		}
		convertInputBuffer( inpInt16Buffer, numSamplesRead, frameSize_smpls, num_in_channels, inFloatBuffer );

		if ( masaReader != NULL )
		{
		@@ -537,18 +531,10 @@ int32_t main( int32_t argc, char **argv )

		int32_t num_out_channels;
		num_out_channels = outBuffer.config.numChannels;
		i = 0;

		/* Convert from float to int and from packed to interleaved */
		for ( smpl = 0; smpl < frameSize_smpls; ++smpl )
		{
		for ( chnl = 0; chnl < num_out_channels; ++chnl )
		{
		outInt16Buffer[i] = (int16_t) ( outFloatBuffer[chnl * frameSize_smpls + smpl] * INT16_MAX );

		++i;
		}
		}
		/* Convert from float to int and from packed to interleaved.
		* Values in outFloatBuffer are guaranteed to be within range INT16_MIN:INT16_MAX */
		convertOutputBuffer( outFloatBuffer, frameSize_smpls, num_out_channels, outInt16Buffer );

		/* TODO tmu : delay compensation not finalized yet */
		if ( delayNumSamples == -1 )
		@@ -2064,3 +2050,64 @@ static void remove_cr( char *str )

		return;
		}

		/--------------------------------------------------------------------------
		* convertInputBuffer()
		*
		* Convert input buffer from WAV/PCM file (int16_t, interleaved) to a format
		* accepted by the renderer (float, packed)
		--------------------------------------------------------------------------/
		static void convertInputBuffer( const int16_t *intBuffer,
		int32_t numIntSamplesPerChannel,
		int32_t numFloatSamplesPerChannel,
		int32_t numChannels,
		float *floatBuffer )
		{
		int32_t chnl, smpl, i;

		i = 0;


		for ( smpl = 0; smpl < numFloatSamplesPerChannel; ++smpl )
		{
		for ( chnl = 0; chnl < numChannels; ++chnl )
		{
		if ( i < numIntSamplesPerChannel )
		{
		floatBuffer[chnl * numFloatSamplesPerChannel + smpl] = (float) intBuffer[i];
		}
		else
		{
		floatBuffer[chnl * numFloatSamplesPerChannel + smpl] = 0.f;
		}

		++i;
		}
		}
		}

		/--------------------------------------------------------------------------
		* 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,
		int32_t numSamplesPerChannel,
		int32_t numChannels,
		int16_t *intBuffer )
		{
		int32_t chnl, smpl, i;

		i = 0;

		for ( smpl = 0; smpl < numSamplesPerChannel; ++smpl )
		{
		for ( chnl = 0; chnl < numChannels; ++chnl )
		{
		intBuffer[i] = (int16_t) roundf( floatBuffer[chnl * numSamplesPerChannel + smpl] );

		++i;
		}
		}
		}

lib_rend/lib_rend.c

+2 −2

Original line number	Diff line number	Diff line
		@@ -46,7 +46,7 @@
		#include <string.h>


		#define LIMITER_THRESHOLD 0.9988493699f /* -0.01 dBFS */
		#define LIMITER_THRESHOLD ( 0.9988493699f * INT16_MAX ) /* -0.01 dBFS */

		/* Due to API of some rendering methods, the renderer has to use the decoder struct.
		Only struct members relevant for rendering will be initialized, therefore typedef as "dummy" decoder struct */
		@@ -3136,7 +3136,7 @@ static int32_t ivas_limiter_renderer(
		}
		#endif

		output[i] = min( max( -1.0f, output[i] ), 1.0f );
		output[i] = min( max( INT16_MIN, output[i] ), INT16_MAX );
		}

		return numClipping;