1HEADmaster

1 files changed, 469 insertions, 0 deletions

diff --git a/engine/audio/private/snd_wave_mixer_adpcm.cpp b/engine/audio/private/snd_wave_mixer_adpcm.cpp
new file mode 100644
index 0000000..ffa71f6
--- /dev/null
+++ b/engine/audio/private/snd_wave_mixer_adpcm.cpp
@@ -0,0 +1,469 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+//=====================================================================================//
+
+#include "audio_pch.h"
+// memdbgon must be the last include file in a .cpp file!!!
+#include "tier0/memdbgon.h"
+
+// max size of ADPCM block in bytes
+#define MAX_BLOCK_SIZE	4096
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Mixer for ADPCM encoded audio
+//-----------------------------------------------------------------------------
+class CAudioMixerWaveADPCM : public CAudioMixerWave
+{
+public:
+	CAudioMixerWaveADPCM( IWaveData *data );
+	~CAudioMixerWaveADPCM( void );
+	
+	virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress );
+	virtual int	 GetOutputData( void **pData, int sampleCount, char copyBuf[AUDIOSOURCE_COPYBUF_SIZE] );
+
+	// need to override this to fixup blocks
+	void SetSampleStart( int newPosition );
+	virtual int GetMixSampleSize() { return CalcSampleSize( 16, NumChannels() ); }
+
+private:
+	bool					DecodeBlock( void );
+	int						NumChannels( void );
+	void					DecompressBlockMono( short *pOut, const char *pIn, int count );
+	void					DecompressBlockStereo( short *pOut, const char *pIn, int count );
+
+	const ADPCMWAVEFORMAT	*m_pFormat;
+	const ADPCMCOEFSET		*m_pCoefficients;
+
+	short					*m_pSamples;
+	int						m_sampleCount;
+	int						m_samplePosition;
+
+	int						m_blockSize;
+	int						m_offset;
+
+	int						m_totalBytes;
+};
+
+
+CAudioMixerWaveADPCM::CAudioMixerWaveADPCM( IWaveData *data ) : CAudioMixerWave( data ) 
+{
+	m_pSamples = NULL;
+	m_sampleCount = 0;
+	m_samplePosition = 0;
+	m_offset = 0;
+
+	CAudioSourceWave &source = reinterpret_cast<CAudioSourceWave &>(m_pData->Source());
+
+#ifdef _DEBUG
+	CAudioSource *pSource = NULL;
+	pSource = &m_pData->Source();
+	Assert( dynamic_cast<CAudioSourceWave *>(pSource) != NULL );
+#endif
+
+	m_pFormat = (const ADPCMWAVEFORMAT *)source.GetHeader();
+	if ( m_pFormat )
+	{
+		m_pCoefficients = (ADPCMCOEFSET *)((char *)m_pFormat + sizeof(WAVEFORMATEX) + 4);
+
+		// create the decode buffer
+		m_pSamples = new short[m_pFormat->wSamplesPerBlock * m_pFormat->wfx.nChannels];
+
+		// number of bytes for samples
+		m_blockSize = ((m_pFormat->wSamplesPerBlock - 2) * m_pFormat->wfx.nChannels ) / 2;
+		// size of channel header
+		m_blockSize += 7 * m_pFormat->wfx.nChannels;
+		Assert( m_blockSize < MAX_BLOCK_SIZE );
+
+		m_totalBytes = source.DataSize();
+	}
+}
+
+
+CAudioMixerWaveADPCM::~CAudioMixerWaveADPCM( void )
+{
+	delete[] m_pSamples;
+}
+
+
+int	CAudioMixerWaveADPCM::NumChannels( void )
+{
+	if ( m_pFormat )
+	{
+		return m_pFormat->wfx.nChannels;
+	}
+	return 0;
+}
+
+void CAudioMixerWaveADPCM::Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress )
+{
+	if ( NumChannels() == 1 )
+		pDevice->Mix16Mono( pChannel, (short *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress );
+	else
+		pDevice->Mix16Stereo( pChannel, (short *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress );
+}
+
+
+static int error_sign_lut[] =		  { 0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1 };
+static int error_coefficients_lut[] = { 230, 230, 230, 230, 307, 409, 512, 614,
+										768, 614, 512, 409, 307, 230, 230, 230 };
+
+//-----------------------------------------------------------------------------
+// Purpose: ADPCM decompress a single block of 1-channel audio
+// Input  : *pOut - output buffer 16-bit
+//			*pIn - input block
+//			count - number of samples to decode (to support partial blocks)
+//-----------------------------------------------------------------------------
+void CAudioMixerWaveADPCM::DecompressBlockMono( short *pOut, const char *pIn, int count )
+{
+	int pred = *pIn++;
+	int co1 = m_pCoefficients[pred].iCoef1;
+	int co2 = m_pCoefficients[pred].iCoef2;
+
+	// read initial delta
+	int delta = *((short *)pIn);
+	pIn += 2;
+
+	// read initial samples for prediction
+	int samp1 = *((short *)pIn);
+	pIn += 2;
+
+	int samp2 = *((short *)pIn);
+	pIn += 2;
+
+	// write out the initial samples (stored in reverse order)
+	*pOut++ = (short)samp2;
+	*pOut++ = (short)samp1;
+
+	// subtract the 2 samples in the header
+	count -= 2;
+
+	// this is a toggle to read nibbles, first nibble is high
+	int high = 1;
+
+	int error, sample=0;
+
+	// now process the block
+	while ( count )
+	{
+		// read the error nibble from the input stream
+		if ( high )
+		{
+			sample = (unsigned char) (*pIn++);
+			// high nibble
+			error = sample >> 4;
+			// cache low nibble for next read
+			sample = sample & 0xf;
+			// Next read is from cache, not stream
+			high = 0;
+		}
+		else
+		{
+			// stored in previous read (low nibble)
+			error = sample;
+			// next read is from stream
+			high = 1;
+		}
+		// convert to signed with LUT
+		int errorSign = error_sign_lut[error];
+
+		// interpolate the new sample
+		int predSample = (samp1 * co1) + (samp2 * co2);
+		// coefficients are fixed point 8-bit, so shift back to 16-bit integer
+		predSample >>= 8;
+
+		// Add in current error estimate
+		predSample += (errorSign * delta);
+
+		// Correct error estimate
+		delta = (delta * error_coefficients_lut[error]) >> 8;
+		// Clamp error estimate
+		if ( delta < 16 )
+			delta = 16;
+
+		// clamp
+		if ( predSample > 32767L )
+			predSample = 32767L;
+		else if ( predSample < -32768L )
+			predSample = -32768L;
+		
+		// output
+		*pOut++ = (short)predSample;
+		// move samples over
+		samp2 = samp1;
+		samp1 = predSample;
+
+		count--;
+	}
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Decode a single block of stereo ADPCM audio
+// Input  : *pOut - 16-bit output buffer
+//			*pIn - ADPCM encoded block data
+//			count - number of sample pairs to decode
+//-----------------------------------------------------------------------------
+void CAudioMixerWaveADPCM::DecompressBlockStereo( short *pOut, const char *pIn, int count )
+{
+	int pred[2], co1[2], co2[2];
+	int i;
+
+	for ( i = 0; i < 2; i++ )
+	{
+		pred[i] = *pIn++;
+		co1[i] = m_pCoefficients[pred[i]].iCoef1;
+		co2[i] = m_pCoefficients[pred[i]].iCoef2;
+	}
+
+	int delta[2], samp1[2], samp2[2];
+
+	for ( i = 0; i < 2; i++, pIn += 2 )
+	{
+		// read initial delta
+		delta[i] = *((short *)pIn);
+	}
+
+	// read initial samples for prediction
+	for ( i = 0; i < 2; i++, pIn += 2 )
+	{
+		samp1[i] = *((short *)pIn);
+	}
+	for ( i = 0; i < 2; i++, pIn += 2 )
+	{
+		samp2[i] = *((short *)pIn);
+	}
+
+	// write out the initial samples (stored in reverse order)
+	*pOut++ = (short)samp2[0];	// left
+	*pOut++ = (short)samp2[1];	// right
+	*pOut++ = (short)samp1[0];	// left
+	*pOut++ = (short)samp1[1];	// right
+
+	// subtract the 2 samples in the header
+	count -= 2;
+
+	// this is a toggle to read nibbles, first nibble is high
+	int high = 1;
+
+	int error, sample=0;
+
+	// now process the block
+	while ( count )
+	{
+		for ( i = 0; i < 2; i++ )
+		{
+			// read the error nibble from the input stream
+			if ( high )
+			{
+				sample = (unsigned char) (*pIn++);
+				// high nibble
+				error = sample >> 4;
+				// cache low nibble for next read
+				sample = sample & 0xf;
+				// Next read is from cache, not stream
+				high = 0;
+			}
+			else
+			{
+				// stored in previous read (low nibble)
+				error = sample;
+				// next read is from stream
+				high = 1;
+			}
+			// convert to signed with LUT
+			int errorSign = error_sign_lut[error];
+
+			// interpolate the new sample
+			int predSample = (samp1[i] * co1[i]) + (samp2[i] * co2[i]);
+			// coefficients are fixed point 8-bit, so shift back to 16-bit integer
+			predSample >>= 8;
+
+			// Add in current error estimate
+			predSample += (errorSign * delta[i]);
+
+			// Correct error estimate
+			delta[i] = (delta[i] * error_coefficients_lut[error]) >> 8;
+			// Clamp error estimate
+			if ( delta[i] < 16 )
+				delta[i] = 16;
+
+			// clamp
+			if ( predSample > 32767L )
+				predSample = 32767L;
+			else if ( predSample < -32768L )
+				predSample = -32768L;
+			
+			// output
+			*pOut++ = (short)predSample;
+			// move samples over
+			samp2[i] = samp1[i];
+			samp1[i] = predSample;
+		}
+		count--;
+	}
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Read data from the source and pass it to the appropriate decompress
+//			routine.
+// Output : Returns true if data was decoded, false if none.
+//-----------------------------------------------------------------------------
+bool CAudioMixerWaveADPCM::DecodeBlock( void )
+{
+	char	tmpBlock[MAX_BLOCK_SIZE];
+	char	*pData;
+	int		blockSize;
+	int		firstSample;
+
+	// fixup position with possible loop
+	CAudioSourceWave &source = reinterpret_cast<CAudioSourceWave &>(m_pData->Source());
+	m_offset = source.ConvertLoopedPosition( m_offset );
+
+	if ( m_offset >= m_totalBytes )
+	{
+		// no more data
+		return false;
+	}
+
+	// can only decode in block sized chunks
+	firstSample = m_offset % m_blockSize;
+	m_offset    = m_offset - firstSample;
+
+	// adpcm must calculate and request correct block size for proper decoding
+	// last block size may be truncated
+	blockSize = m_totalBytes - m_offset;
+	if ( blockSize > m_blockSize )
+	{
+		blockSize = m_blockSize;
+	}
+
+	// get requested data
+	int available = m_pData->ReadSourceData( (void **)(&pData), m_offset, blockSize, NULL );
+	if ( available < blockSize )
+	{
+		// pump to get all of requested data
+		int total = 0;
+		while ( available && total < blockSize )
+		{
+			memcpy( tmpBlock + total, pData, available );
+			total += available;
+			available = m_pData->ReadSourceData( (void **)(&pData), m_offset + total, blockSize - total, NULL );
+		}
+		pData     = tmpBlock;
+		available = total;
+	}
+
+	if ( !available )
+	{
+		// no more data
+		return false;
+	}
+
+	// advance the file pointer
+	m_offset += available;
+
+	int channelCount = NumChannels();
+
+	// this is sample pairs for stereo, samples for mono
+	m_sampleCount = m_pFormat->wSamplesPerBlock;
+
+	// short block?, fixup sample count (2 samples per byte, divided by number of channels per sample set)
+	m_sampleCount -= ((m_blockSize - available) * 2) / channelCount;
+
+	// new block, start at the first sample
+	m_samplePosition = firstSample;
+
+	// no need to subclass for different channel counts...
+	if ( channelCount == 1 )
+	{
+		DecompressBlockMono( m_pSamples, pData, m_sampleCount );
+	}
+	else
+	{
+		DecompressBlockStereo( m_pSamples, pData, m_sampleCount );
+	}
+	return true;
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Read existing buffer or decompress a new block when necessary
+// Input  : **pData - output data pointer
+//			sampleCount - number of samples (or pairs)
+// Output : int - available samples (zero to stop decoding)
+//-----------------------------------------------------------------------------
+int CAudioMixerWaveADPCM::GetOutputData( void **pData, int sampleCount, char copyBuf[AUDIOSOURCE_COPYBUF_SIZE] )
+{
+	if ( m_samplePosition >= m_sampleCount )
+	{
+		if ( !DecodeBlock() )
+			return 0;
+	}
+
+	if ( m_pSamples && m_samplePosition < m_sampleCount )
+	{
+		*pData = (void *)(m_pSamples + m_samplePosition * NumChannels());
+		int available = m_sampleCount - m_samplePosition;
+		if ( available > sampleCount )
+			available = sampleCount;
+
+		m_samplePosition += available;
+
+		// update count of max samples loaded in CAudioMixerWave
+		CAudioMixerWave::m_sample_max_loaded += available;
+	
+		// update index of last sample loaded
+		CAudioMixerWave::m_sample_loaded_index += available;
+
+		return available;
+	}
+
+	return 0;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Seek to a new position in the file
+//			NOTE: In most cases, only call this once, and call it before playing
+//			any data.
+// Input  : newPosition - new position in the sample clocks of this sample
+//-----------------------------------------------------------------------------
+void CAudioMixerWaveADPCM::SetSampleStart( int newPosition )
+{
+	// cascade to base wave to update sample counter
+	CAudioMixerWave::SetSampleStart( newPosition );
+	
+	// which block is the desired starting sample in?
+	int blockStart = newPosition / m_pFormat->wSamplesPerBlock;
+	// how far into the block is the sample
+	int blockOffset = newPosition % m_pFormat->wSamplesPerBlock;
+
+	// set the file position
+	m_offset = blockStart * m_blockSize;
+	
+	// NOTE: Must decode a block here to properly position the sample Index
+	// THIS MEANS YOU DON'T WANT TO CALL THIS ROUTINE OFTEN FOR ADPCM SOUNDS
+	DecodeBlock();
+	
+	// limit to the samples decoded
+	if ( blockOffset < m_sampleCount )
+		blockOffset = m_sampleCount;
+	
+	// set the new current position
+	m_samplePosition = blockOffset;
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Abstract factory function for ADPCM mixers
+// Input  : *data - wave data access object
+//			channels - 
+// Output : CAudioMixer
+//-----------------------------------------------------------------------------
+CAudioMixer *CreateADPCMMixer( IWaveData *data )
+{
+	return new CAudioMixerWaveADPCM( data );
+}