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//========= Copyright Valve Corporation, All rights reserved. ============//
//
// Purpose:
//
// $NoKeywords: $
//
//=============================================================================//
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#pragma warning( disable: 4201 )
#include <mmsystem.h>
#pragma warning( default: 4201 )
#include <mmreg.h>
#include "snd_wave_source.h"
#include "snd_wave_mixer_adpcm.h"
#include "snd_wave_mixer_private.h"
#include "soundsystem.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( CWaveData *data );
~CAudioMixerWaveADPCM( void );
virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress, bool forward = true );
virtual int GetOutputData( void **pData, int samplePosition, int sampleCount, bool forward = true );
virtual bool SetSamplePosition( int position, bool scrubbing = false );
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 );
void SetCurrentBlock( int block );
int GetCurrentBlock( void ) const;
int GetBlockNumberForSample( int samplePosition );
bool IsSampleInCurrentBlock( int samplePosition );
int GetFirstSampleForBlock( int blocknum ) const;
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_currentBlock;
};
CAudioMixerWaveADPCM::CAudioMixerWaveADPCM( CWaveData *data ) : CAudioMixerWave( data )
{
m_currentBlock = -1;
m_pSamples = NULL;
m_sampleCount = 0;
m_samplePosition = 0;
m_offset = 0;
m_pFormat = (const ADPCMWAVEFORMAT *)m_pData->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);
}
}
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, bool forward /*= true*/ )
{
if ( NumChannels() == 1 )
pDevice->Mix16Mono( pChannel, (short *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress, forward );
else
pDevice->Mix16Stereo( pChannel, (short *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress, forward );
}
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 = 0, 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--;
}
}
bool CAudioMixerWaveADPCM::DecodeBlock( void )
{
char tmpBlock[MAX_BLOCK_SIZE];
char *pData;
int available = m_pData->ReadSourceData( (void **) (&pData), m_offset, m_blockSize );
if ( available < m_blockSize )
{
int total = 0;
while ( available && total < m_blockSize )
{
memcpy( tmpBlock + total, pData, available );
total += available;
available = m_pData->ReadSourceData( (void **) (&pData), m_offset + total, m_blockSize - total );
}
pData = tmpBlock;
available = total;
}
Assert( m_blockSize > 0 );
// Current block number is based on starting offset
int blockNumber = m_offset / m_blockSize;
SetCurrentBlock( blockNumber );
if ( !available )
{
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 = 0;
// 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:
// Input : block -
//-----------------------------------------------------------------------------
void CAudioMixerWaveADPCM::SetCurrentBlock( int block )
{
m_currentBlock = block;
}
//-----------------------------------------------------------------------------
// Purpose:
// Output : int
//-----------------------------------------------------------------------------
int CAudioMixerWaveADPCM::GetCurrentBlock( void ) const
{
return m_currentBlock;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : samplePosition -
// Output : int
//-----------------------------------------------------------------------------
int CAudioMixerWaveADPCM::GetBlockNumberForSample( int samplePosition )
{
int blockNum = samplePosition / m_pFormat->wSamplesPerBlock;
return blockNum;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : samplePosition -
// Output : Returns true on success, false on failure.
//-----------------------------------------------------------------------------
bool CAudioMixerWaveADPCM::IsSampleInCurrentBlock( int samplePosition )
{
int currentBlock = GetCurrentBlock();
int startSample = currentBlock * m_pFormat->wSamplesPerBlock;
int endSample = startSample + m_pFormat->wSamplesPerBlock - 1;
if ( samplePosition >= startSample &&
samplePosition <= endSample )
{
return true;
}
return false;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : blocknum -
// Output : int
//-----------------------------------------------------------------------------
int CAudioMixerWaveADPCM::GetFirstSampleForBlock( int blocknum ) const
{
return m_pFormat->wSamplesPerBlock * blocknum;
}
//-----------------------------------------------------------------------------
// 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 samplePosition, int sampleCount, bool forward /*= true*/ )
{
int requestedBlock = GetBlockNumberForSample( samplePosition );
if ( requestedBlock != GetCurrentBlock() )
{
// Ran out of data!!!
if ( !SetSamplePosition( samplePosition ) )
return 0;
}
Assert( requestedBlock == GetCurrentBlock() );
if ( m_samplePosition >= m_sampleCount )
{
if ( !DecodeBlock() )
return 0;
}
if ( 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;
return available;
}
return 0;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : position -
//-----------------------------------------------------------------------------
bool CAudioMixerWaveADPCM::SetSamplePosition( int position, bool scrubbing )
{
position = max( 0, position );
CAudioMixerWave::SetSamplePosition( position, scrubbing );
int requestedBlock = GetBlockNumberForSample( position );
int firstSample = GetFirstSampleForBlock( requestedBlock );
if ( firstSample >= m_pData->Source().SampleCount() )
{
// Read past end of file!!!
return false;
}
int currentSample = ( position - firstSample );
if ( requestedBlock != GetCurrentBlock() )
{
// Rewind file to beginning of block
m_offset = requestedBlock * m_blockSize;
if ( !DecodeBlock() )
{
return false;
}
}
m_samplePosition = currentSample;
return true;
}
//-----------------------------------------------------------------------------
// Purpose: Abstract factory function for ADPCM mixers
// Input : *data - wave data access object
// channels -
// Output : CAudioMixer
//-----------------------------------------------------------------------------
CAudioMixer *CreateADPCMMixer( CWaveData *data )
{
return new CAudioMixerWaveADPCM( data );
}