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