319 lines
7.4 KiB
C++
319 lines
7.4 KiB
C++
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// Mixf.cpp
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// Copyright (c) Microsoft Corporation 1996-1999
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// Filtered Mix Engine
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#include "simple.h"
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#include <mmsystem.h>
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#include "synth.h"
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//>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
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#pragma message ("Programer note: property hack")
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//#define DEBUG_DUMP_FILE
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#pragma warning(disable : 4101 4102 4146)
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//>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
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#ifdef DEBUG_DUMP_FILE
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DWORD dmp_bufsize = 4000000;
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DWORD dmp_samplesrecorded;
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DWORD dmp_buffercount;
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short dmp_soundbuffer[4000000];
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#endif
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//>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
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DWORD CDigitalAudio::Mix16Filtered(
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short **ppBuffers,
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DWORD *pdwChannels,
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DWORD dwBufferCount,
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DWORD dwLength,
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DWORD dwDeltaPeriod,
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VFRACT vfDeltaLVolume,
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VFRACT vfDeltaRVolume,
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PFRACT pfDeltaPitch,
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PFRACT pfSampleLength,
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PFRACT pfLoopLength,
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COEFFDELTA cfdK,
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COEFFDELTA cfdB1,
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COEFFDELTA cfdB2)
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{
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DWORD dwI;
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DWORD dwIndex;
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DWORD dwPosition;
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long lA;
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long lM;
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DWORD dwIncDelta = dwDeltaPeriod;
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VFRACT dwFract;
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short * pcWave = m_pnWave;
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PFRACT pfSamplePos = m_pfLastSample;
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VFRACT vfLVolume = m_vfLastLVolume;
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VFRACT vfRVolume = m_vfLastRVolume;
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PFRACT pfPitch = m_pfLastPitch;
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PFRACT pfPFract = pfPitch << 8;
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VFRACT vfLVFract = vfLVolume << 8;
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VFRACT vfRVFract = vfRVolume << 8;
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COEFF cfK = m_cfLastK;
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COEFF cfB1 = m_cfLastB1;
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COEFF cfB2 = m_cfLastB2;
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for (dwI = 0; dwI < dwLength;)
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{
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if (pfSamplePos >= pfSampleLength)
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{
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if (pfLoopLength)
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{
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pfSamplePos -= pfLoopLength;
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}
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else
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break;
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}
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dwIncDelta--;
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if (!dwIncDelta)
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{
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dwIncDelta = dwDeltaPeriod;
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pfPFract += pfDeltaPitch;
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pfPitch = pfPFract >> 8;
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vfLVFract += vfDeltaLVolume;
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vfLVolume = vfLVFract >> 8;
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vfRVFract += vfDeltaRVolume;
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vfRVolume = vfRVFract >> 8;
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cfK += cfdK;
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cfB1 += cfdB1;
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cfB2 += cfdB2;
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}
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dwPosition = pfSamplePos >> 12;
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dwFract = pfSamplePos & 0xFFF;
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pfSamplePos += pfPitch;
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// Interpolate
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lA = (long)pcWave[dwPosition];
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lM = (((pcWave[dwPosition+1] - lA) * dwFract) >> 12) + lA;
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//
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// Filter
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//
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// z = k*s - b1*z1 - b2*b2
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// >>>> We store the negative of b1 in the table, so we flip the sign again by
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// >>>> adding here
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// >>>> Lookinto simply using a float here, it may just be faster, save a div
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//
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lM = MulDiv(lM, cfK, (1 << 30))
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+ MulDiv(m_lPrevSample, cfB1, (1 << 30))
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- MulDiv(m_lPrevPrevSample, cfB2, (1 << 30));
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//
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//
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//
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m_lPrevPrevSample = m_lPrevSample;
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m_lPrevSample = lM;
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//
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//
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//
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lA = lM;
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lA *= vfLVolume;
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lA >>= 13; // Signal bumps up to 15 bits.
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lM *= vfRVolume;
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lM >>= 13;
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dwIndex = 0;
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while ( dwIndex < dwBufferCount )
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{
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short *pBuffer = &ppBuffers[dwIndex][dwI];
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if ( pdwChannels[dwIndex] & WAVELINK_CHANNEL_LEFT )
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{
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// Keep this around so we can use it to generate new assembly code (see below...)
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*pBuffer += (short) lA;
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_asm{jno no_oflowl}
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*pBuffer = 0x7fff;
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_asm{js no_oflowl}
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*pBuffer = (short) 0x8000;
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}
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no_oflowl:
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if ( pdwChannels[dwIndex] & WAVELINK_CHANNEL_RIGHT )
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{
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// Keep this around so we can use it to generate new assembly code (see below...)
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*pBuffer += (short) lM;
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_asm{jno no_oflowr}
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*pBuffer = 0x7fff;
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_asm{js no_oflowr}
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*pBuffer = (short) 0x8000;
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}
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no_oflowr:
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dwIndex++;
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}
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#ifdef DEBUG_DUMP_FILE
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dmp_soundbuffer[dmp_samplesrecorded] = pBuffer[dwI];
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if (dmp_samplesrecorded < dmp_bufsize)
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dmp_samplesrecorded++ ;
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#endif
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dwI++;
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}
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m_vfLastLVolume = vfLVolume;
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m_vfLastRVolume = vfRVolume;
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m_pfLastPitch = pfPitch;
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m_pfLastSample = pfSamplePos;
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m_cfLastK = cfK;
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m_cfLastB1 = cfB1;
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m_cfLastB2 = cfB2;
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return (dwI);
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}
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DWORD CDigitalAudio::Mix16FilteredInterleaved(
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short **ppBuffers,
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DWORD *pdwChannels,
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DWORD dwBufferCount,
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DWORD dwLength,
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DWORD dwDeltaPeriod,
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VFRACT vfDeltaLVolume,
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VFRACT vfDeltaRVolume,
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PFRACT pfDeltaPitch,
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PFRACT pfSampleLength,
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PFRACT pfLoopLength,
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COEFFDELTA cfdK,
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COEFFDELTA cfdB1,
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COEFFDELTA cfdB2)
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{
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DWORD dwI;
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DWORD dwIndex;
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DWORD dwPosition;
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long lA;
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long lM;
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DWORD dwIncDelta = dwDeltaPeriod;
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VFRACT dwFract;
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short * pcWave = m_pnWave;
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PFRACT pfSamplePos = m_pfLastSample;
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VFRACT vfLVolume = m_vfLastLVolume;
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VFRACT vfRVolume = m_vfLastRVolume;
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PFRACT pfPitch = m_pfLastPitch;
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PFRACT pfPFract = pfPitch << 8;
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VFRACT vfLVFract = vfLVolume << 8;
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VFRACT vfRVFract = vfRVolume << 8;
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COEFF cfK = m_cfLastK;
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COEFF cfB1 = m_cfLastB1;
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COEFF cfB2 = m_cfLastB2;
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dwLength <<= 1;
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for (dwI = 0; dwI < dwLength;)
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{
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if (pfSamplePos >= pfSampleLength)
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{
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if (pfLoopLength)
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{
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pfSamplePos -= pfLoopLength;
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}
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else
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break;
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}
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dwIncDelta--;
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if (!dwIncDelta)
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{
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dwIncDelta = dwDeltaPeriod;
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pfPFract += pfDeltaPitch;
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pfPitch = pfPFract >> 8;
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vfLVFract += vfDeltaLVolume;
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vfLVolume = vfLVFract >> 8;
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vfRVFract += vfDeltaRVolume;
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vfRVolume = vfRVFract >> 8;
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cfK += cfdK;
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cfB1 += cfdB1;
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cfB2 += cfdB2;
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}
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dwPosition = pfSamplePos >> 12;
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dwFract = pfSamplePos & 0xFFF;
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pfSamplePos += pfPitch;
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// Interpolate
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lA = (long)pcWave[dwPosition];
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lM = (((pcWave[dwPosition+1] - lA) * dwFract) >> 12) + lA;
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//
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// Filter
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//
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// z = k*s - b1*z1 - b2*b2
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// >>>> We store the negative of b1 in the table, so we flip the sign again by
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// >>>> adding here
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// >>>> Lookinto simply using a float here, it may just be faster, save a div
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//
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lM = MulDiv(lM, cfK, (1 << 30))
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+ MulDiv(m_lPrevSample, cfB1, (1 << 30))
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- MulDiv(m_lPrevPrevSample, cfB2, (1 << 30));
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//
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//
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//
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m_lPrevPrevSample = m_lPrevSample;
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m_lPrevSample = lM;
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//
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//
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//
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lA = lM;
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lA *= vfLVolume;
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lA >>= 13; // Signal bumps up to 15 bits.
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lM *= vfRVolume;
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lM >>= 13;
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dwIndex = 0;
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while ( dwIndex < dwBufferCount )
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{
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short *pBuffer = &ppBuffers[dwIndex][dwI];
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if ( pdwChannels[dwIndex] & WAVELINK_CHANNEL_LEFT )
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{
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// Keep this around so we can use it to generate new assembly code (see below...)
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*pBuffer += (short) lA;
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_asm{jno no_oflowl}
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*pBuffer = 0x7fff;
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_asm{js no_oflowl}
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*pBuffer = (short) 0x8000;
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}
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no_oflowl:
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if ( pdwChannels[dwIndex] & WAVELINK_CHANNEL_RIGHT )
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{
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// Keep this around so we can use it to generate new assembly code (see below...)
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pBuffer++;
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*pBuffer += (short) lM;
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_asm{jno no_oflowr}
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*pBuffer = 0x7fff;
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_asm{js no_oflowr}
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*pBuffer = (short) 0x8000;
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}
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no_oflowr:
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dwIndex++;
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}
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#ifdef DEBUG_DUMP_FILE
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dmp_soundbuffer[dmp_samplesrecorded] = pBuffer[dwI];
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if (dmp_samplesrecorded < dmp_bufsize)
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dmp_samplesrecorded++ ;
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#endif
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dwI += 2;
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}
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m_vfLastLVolume = vfLVolume;
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m_vfLastRVolume = vfRVolume;
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m_pfLastPitch = pfPitch;
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m_pfLastSample = pfSamplePos;
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m_cfLastK = cfK;
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m_cfLastB1 = cfB1;
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m_cfLastB2 = cfB2;
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return (dwI >> 1);
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}
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