/*
* Fastmix.cpp
* -----------
* Purpose: Mixer core for rendering samples, mixing plugins, etc...
* Notes : If this is Fastmix.cpp, where is Slowmix.cpp? :)
* Authors: Olivier Lapicque
* OpenMPT Devs
* The OpenMPT source code is released under the BSD license. Read LICENSE for more details.
*/
// FIXME:
// - Playing samples backwards should reverse interpolation LUTs for interpolation modes
// with more than two taps since they're not symmetric. We might need separate LUTs
// because otherwise we will add tons of branches.
// - Loop wraparound works pretty well in general, but not at the start of bidi samples.
// - The loop lookahead stuff might still fail for samples with backward loops.
#include "stdafx.h"
#include "Sndfile.h"
#include "MixerLoops.h"
#include "MixFuncTable.h"
#include "plugins/PlugInterface.h"
#include <cfloat> // For FLT_EPSILON
#include <algorithm>
OPENMPT_NAMESPACE_BEGIN
/////////////////////////////////////////////////////////////////////////
struct MixLoopState
{
const int8 * samplePointer = nullptr;
const int8 * lookaheadPointer = nullptr;
SmpLength lookaheadStart = 0;
uint32 maxSamples = 0;
const uint8 ITPingPongDiff;
const bool precisePingPongLoops;
MixLoopState(const CSoundFile &sndFile, const ModChannel &chn)
: ITPingPongDiff{sndFile.m_playBehaviour[kITPingPongMode] ? uint8(1) : uint8(0)}
, precisePingPongLoops{!sndFile.m_playBehaviour[kImprecisePingPongLoops]}
{
if(chn.pCurrentSample == nullptr)
return;
UpdateLookaheadPointers(chn);
// For platforms that have no fast 64-bit division, precompute this constant
// as it won't change during the invocation of CreateStereoMix.
SamplePosition increment = chn.increment;
if(increment.IsNegative())
increment.Negate();
maxSamples = 16384u / (increment.GetUInt() + 1u);
if(maxSamples < 2)
maxSamples = 2;
}
// Calculate offset of loop wrap-around buffer for this sample.
void UpdateLookaheadPointers(const ModChannel &chn)
{
samplePointer = static_cast<const int8 *>(chn.pCurrentSample);
lookaheadPointer = nullptr;
if(!samplePointer)
return;
if(chn.nLoopEnd < InterpolationLookaheadBufferSize)
lookaheadStart = chn.nLoopStart;
else
lookaheadStart = std::max(chn.nLoopStart, chn.nLoopEnd - InterpolationLookaheadBufferSize);
// We only need to apply the loop wrap-around logic if the sample is actually looping and if interpolation is applied.
// If there is no interpolation happening, there is no lookahead happening the sample read-out is exact.
if(chn.dwFlags[CHN_LOOP] && chn.resamplingMode != SRCMODE_NEAREST)
{
const bool inSustainLoop = chn.InSustainLoop() && chn.nLoopStart == chn.pModSample->nSustainStart && chn.nLoopEnd == chn.pModSample->nSustainEnd;
// Do not enable wraparound magic if we're previewing a custom loop!
if(inSustainLoop || chn.nLoopEnd == chn.pModSample->nLoopEnd)
{
SmpLength lookaheadOffset = 3 * InterpolationLookaheadBufferSize + chn.pModSample->nLength - chn.nLoopEnd;
if(inSustainLoop)
{
lookaheadOffset += 4 * InterpolationLookaheadBufferSize;
}
lookaheadPointer = samplePointer + lookaheadOffset * chn.pModSample->GetBytesPerSample();
}
}
}
// Returns the buffer length required to render a certain amount of samples, based on the channel's playback speed.
static MPT_FORCEINLINE uint32 DistanceToBufferLength(SamplePosition from, SamplePosition to, SamplePosition inc)
{
return static_cast<uint32>((to - from - SamplePosition(1)) / inc) + 1;
}
// Check how many samples can be rendered without encountering loop or sample end, and also update loop position / direction
MPT_FORCEINLINE uint32 GetSampleCount(ModChannel &chn, uint32 nSamples) const
{
int32 nLoopStart = chn.dwFlags[CHN_LOOP] ? chn.nLoopStart : 0;
SamplePosition nInc = chn.increment;
if(nSamples <= 0 || nInc.IsZero() || !chn.nLength || !samplePointer)
return 0;
// Part 1: Making sure the play position is valid, and if necessary, invert the play direction in case we reached a loop boundary of a ping-pong loop.
chn.pCurrentSample = samplePointer;
// Under zero ?
if (chn.position.GetInt() < nLoopStart)
{
if (nInc.IsNegative())
{
// Invert loop direction for bidi loops
chn.position = SamplePosition(nLoopStart + nLoopStart, 0) - chn.position;
if ((chn.position.GetInt() < nLoopStart) || (chn.position.GetUInt() >= (nLoopStart + chn.nLength) / 2))
{
chn.position.Set(nLoopStart, 0);
}
if(chn.dwFlags[CHN_PINGPONGLOOP])
{
chn.dwFlags.reset(CHN_PINGPONGFLAG); // go forward
nInc.Negate();
chn.increment = nInc;
} else
{
chn.position.SetInt(chn.nLength - 1);
}
if(!chn.dwFlags[CHN_LOOP] || chn.position.GetUInt() >= chn.nLength)
{
chn.position.Set(chn.nLength);
return 0;
}
} else
{
// We probably didn't hit the loop end yet (first loop), so we do nothing
if (chn.position.GetInt() < 0) chn.position.SetInt(0);
}
} else if (chn.position.GetUInt() >= chn.nLength)
{
// Past the end
if(!chn.dwFlags[CHN_LOOP])
return 0; // not looping -> stop this channel
if(chn.dwFlags[CHN_PINGPONGLOOP])
{
// Invert loop
if (nInc.IsPositive())
{
nInc.Negate();
chn.increment = nInc;
}
chn.dwFlags.set(CHN_PINGPONGFLAG);
// Adjust loop position
if(precisePingPongLoops)
{
// More accurate loop end overshoot calculation.
// Test cases: BidiPrecision.it, BidiPrecision.xm
const auto overshoot = chn.position - SamplePosition(chn.nLength, 0);
const auto loopLength = chn.nLoopEnd - chn.nLoopStart - ITPingPongDiff;
if(overshoot.GetUInt() < loopLength)
chn.position = SamplePosition(chn.nLength - ITPingPongDiff, 0) - overshoot;
else
chn.position = SamplePosition(chn.nLoopStart, 0);
} else
{
SamplePosition invFract = chn.position.GetInvertedFract();
chn.position = SamplePosition(chn.nLength - (chn.position.GetInt() - chn.nLength) - invFract.GetInt(), invFract.GetFract());
if(chn.position.GetUInt() <= chn.nLoopStart || chn.position.GetUInt() >= chn.nLength)
{
// Impulse Tracker's software mixer would put a -2 (instead of -1) in the following line (doesn't happen on a GUS)
chn.position.SetInt(chn.nLength - std::min(chn.nLength, static_cast<SmpLength>(ITPingPongDiff + 1)));
}
}
} else
{
if (nInc.IsNegative()) // This is a bug
{
nInc.Negate();
chn.increment = nInc;
}
// Restart at loop start
chn.position += SamplePosition(nLoopStart - chn.nLength, 0);
MPT_ASSERT(chn.position.GetInt() >= nLoopStart);
// Interpolate correctly after wrapping around
chn.dwFlags.set(CHN_WRAPPED_LOOP);
}
}
// Part 2: Compute how many samples we can render until we reach the end of sample / loop boundary / etc.
SamplePosition nPos = chn.position;
const SmpLength nPosInt = nPos.GetUInt();
if(nPos.GetInt() < nLoopStart)
{
// too big increment, and/or too small loop length
if(nPos.IsNegative() || nInc.IsNegative())
return 0;
} else
{
// Not testing for equality since we might be going backwards from the very end of the sample
if(nPosInt > chn.nLength)
return 0;
// If going forwards and we're preceisely at the end, there's no point in going further
if(nPosInt == chn.nLength && nInc.IsPositive())
return 0;
}
uint32 nSmpCount = nSamples;
SamplePosition nInv = nInc;
if (nInc.IsNegative())
{
nInv.Negate();
}
LimitMax(nSamples, maxSamples);
SamplePosition incSamples = nInc * (nSamples - 1);
int32 nPosDest = (nPos + incSamples).GetInt();
const bool isAtLoopStart = (nPosInt >= chn.nLoopStart && nPosInt < chn.nLoopStart + InterpolationLookaheadBufferSize);
if(!isAtLoopStart)
{
chn.dwFlags.reset(CHN_WRAPPED_LOOP);
}
// Loop wrap-around magic.
bool checkDest = true;
if(lookaheadPointer != nullptr)
{
if(nPosInt >= lookaheadStart)
{
#if 0
const uint32 oldCount = nSmpCount;
// When going backwards - we can only go back up to lookaheadStart.
// When going forwards - read through the whole pre-computed wrap-around buffer if possible.
// TODO: ProTracker sample swapping needs hard cut at sample end.
int32 samplesToRead = nInc.IsNegative()
? (nPosInt - lookaheadStart)
//: 2 * InterpolationMaxLookahead - (nPosInt - mixLoopState.lookaheadStart);
: (chn.nLoopEnd - nPosInt);
//LimitMax(samplesToRead, chn.nLoopEnd - chn.nLoopStart);
nSmpCount = SamplesToBufferLength(samplesToRead, chn);
Limit(nSmpCount, 1u, oldCount);
#else
if (nInc.IsNegative())
{
nSmpCount = DistanceToBufferLength(SamplePosition(lookaheadStart, 0), nPos, nInv);
} else
{
nSmpCount = DistanceToBufferLength(nPos, SamplePosition(chn.nLoopEnd, 0), nInv);
}
#endif
chn.pCurrentSample = lookaheadPointer;
checkDest = false;
} else if(chn.dwFlags[CHN_WRAPPED_LOOP] && isAtLoopStart)
{
// We just restarted the loop, so interpolate correctly after wrapping around
nSmpCount = DistanceToBufferLength(nPos, SamplePosition(nLoopStart + InterpolationLookaheadBufferSize, 0), nInv);
chn.pCurrentSample = lookaheadPointer + (chn.nLoopEnd - nLoopStart) * chn.pModSample->GetBytesPerSample();
checkDest = false;
} else if(nInc.IsPositive() && static_cast<SmpLength>(nPosDest) >= lookaheadStart && nSmpCount > 1)
{
// We shouldn't read that far if we're not using the pre-computed wrap-around buffer.
nSmpCount = DistanceToBufferLength(nPos, SamplePosition(lookaheadStart, 0), nInv);
checkDest = false;
}
}
if(checkDest)
{
// Fix up sample count if target position is invalid
if (nInc.IsNegative())
{
if (nPosDest < nLoopStart)
{
nSmpCount = DistanceToBufferLength(SamplePosition(nLoopStart, 0), nPos, nInv);
}
} else
{
if (nPosDest >= (int32)chn.nLength)
{
nSmpCount = DistanceToBufferLength(nPos, SamplePosition(chn.nLength, 0), nInv);
}
}
}
Limit(nSmpCount, uint32(1u), nSamples);
#ifdef MPT_BUILD_DEBUG
{
SmpLength posDest = (nPos + nInc * (nSmpCount - 1)).GetUInt();
if (posDest < 0 || posDest > chn.nLength)
{
// We computed an invalid delta!
MPT_ASSERT_NOTREACHED();
return 0;
}
}
#endif
return nSmpCount;
}
};
// Render count * number of channels samples
void CSoundFile::CreateStereoMix(int count)
{
mixsample_t *pOfsL, *pOfsR;
if(!count)
return;
// Resetting sound buffer
StereoFill(MixSoundBuffer, count, m_dryROfsVol, m_dryLOfsVol);
if(m_MixerSettings.gnChannels > 2)
StereoFill(MixRearBuffer, count, m_surroundROfsVol, m_surroundLOfsVol);
CHANNELINDEX nchmixed = 0;
for(uint32 nChn = 0; nChn < m_nMixChannels; nChn++)
{
ModChannel &chn = m_PlayState.Chn[m_PlayState.ChnMix[nChn]];
if(!chn.pCurrentSample && !chn.nLOfs && !chn.nROfs)
continue;
pOfsR = &m_dryROfsVol;
pOfsL = &m_dryLOfsVol;
uint32 functionNdx = MixFuncTable::ResamplingModeToMixFlags(static_cast<ResamplingMode>(chn.resamplingMode));
if(chn.dwFlags[CHN_16BIT]) functionNdx |= MixFuncTable::ndx16Bit;
if(chn.dwFlags[CHN_STEREO]) functionNdx |= MixFuncTable::ndxStereo;
#ifndef NO_FILTER
if(chn.dwFlags[CHN_FILTER]) functionNdx |= MixFuncTable::ndxFilter;
#endif
mixsample_t *pbuffer = MixSoundBuffer;
#ifndef NO_REVERB
if(((m_MixerSettings.DSPMask & SNDDSP_REVERB) && !chn.dwFlags[CHN_NOREVERB]) || chn.dwFlags[CHN_REVERB])
{
m_Reverb.TouchReverbSendBuffer(ReverbSendBuffer, m_RvbROfsVol, m_RvbLOfsVol, count);
pbuffer = ReverbSendBuffer;
pOfsR = &m_RvbROfsVol;
pOfsL = &m_RvbLOfsVol;
}
#endif
if(chn.dwFlags[CHN_SURROUND] && m_MixerSettings.gnChannels > 2)
{
pbuffer = MixRearBuffer;
pOfsR = &m_surroundROfsVol;
pOfsL = &m_surroundLOfsVol;
}
//Look for plugins associated with this implicit tracker channel.
#ifndef NO_PLUGINS
PLUGINDEX nMixPlugin = GetBestPlugin(m_PlayState, m_PlayState.ChnMix[nChn], PrioritiseInstrument, RespectMutes);
if ((nMixPlugin > 0) && (nMixPlugin <= MAX_MIXPLUGINS) && m_MixPlugins[nMixPlugin - 1].pMixPlugin != nullptr)
{
// Render into plugin buffer instead of global buffer
SNDMIXPLUGINSTATE &mixState = m_MixPlugins[nMixPlugin - 1].pMixPlugin->m_MixState;
if (mixState.pMixBuffer)
{
pbuffer = mixState.pMixBuffer;
pOfsR = &mixState.nVolDecayR;
pOfsL = &mixState.nVolDecayL;
if (!(mixState.dwFlags & SNDMIXPLUGINSTATE::psfMixReady))
{
StereoFill(pbuffer, count, *pOfsR, *pOfsL);
mixState.dwFlags |= SNDMIXPLUGINSTATE::psfMixReady;
}
}
}
#endif // NO_PLUGINS
if(chn.isPaused)
{
EndChannelOfs(chn, pbuffer, count);
*pOfsR += chn.nROfs;
*pOfsL += chn.nLOfs;
chn.nROfs = chn.nLOfs = 0;
continue;
}
MixLoopState mixLoopState(*this, chn);
////////////////////////////////////////////////////
CHANNELINDEX naddmix = 0;
int nsamples = count;
// Keep mixing this sample until the buffer is filled.
do
{
uint32 nrampsamples = nsamples;
int32 nSmpCount;
if(chn.nRampLength > 0)
{
if (nrampsamples > chn.nRampLength) nrampsamples = chn.nRampLength;
}
if((nSmpCount = mixLoopState.GetSampleCount(chn, nrampsamples)) <= 0)
{
// Stopping the channel
chn.pCurrentSample = nullptr;
chn.nLength = 0;
chn.position.Set(0);
chn.nRampLength = 0;
EndChannelOfs(chn, pbuffer, nsamples);
*pOfsR += chn.nROfs;
*pOfsL += chn.nLOfs;
chn.nROfs = chn.nLOfs = 0;
chn.dwFlags.reset(CHN_PINGPONGFLAG);
break;
}
// Should we mix this channel ?
if((nchmixed >= m_MixerSettings.m_nMaxMixChannels) // Too many channels
|| (!chn.nRampLength && !(chn.leftVol | chn.rightVol))) // Channel is completely silent
{
chn.position += chn.increment * nSmpCount;
chn.nROfs = chn.nLOfs = 0;
pbuffer += nSmpCount * 2;
naddmix = 0;
}
#ifdef MODPLUG_TRACKER
else if(m_SamplePlayLengths != nullptr)
{
// Detecting the longest play time for each sample for optimization
SmpLength pos = chn.position.GetUInt();
chn.position += chn.increment * nSmpCount;
if(!chn.increment.IsNegative())
{
pos = chn.position.GetUInt();
}
size_t smp = std::distance(static_cast<const ModSample*>(static_cast<std::decay<decltype(Samples)>::type>(Samples)), chn.pModSample);
if(smp < m_SamplePlayLengths->size())
{
(*m_SamplePlayLengths)[smp] = std::max((*m_SamplePlayLengths)[smp], pos);
}
}
#endif
else
{
// Do mixing
mixsample_t *pbufmax = pbuffer + (nSmpCount * 2);
chn.nROfs = -*(pbufmax - 2);
chn.nLOfs = -*(pbufmax - 1);
#ifdef MPT_BUILD_DEBUG
SamplePosition targetpos = chn.position + chn.increment * nSmpCount;
#endif
MixFuncTable::Functions[functionNdx | (chn.nRampLength ? MixFuncTable::ndxRamp : 0)](chn, m_Resampler, pbuffer, nSmpCount);
#ifdef MPT_BUILD_DEBUG
MPT_ASSERT(chn.position.GetUInt() == targetpos.GetUInt());
#endif
chn.nROfs += *(pbufmax - 2);
chn.nLOfs += *(pbufmax - 1);
pbuffer = pbufmax;
naddmix = 1;
}
nsamples -= nSmpCount;
if (chn.nRampLength)
{
if (chn.nRampLength <= static_cast<uint32>(nSmpCount))
{
// Ramping is done
chn.nRampLength = 0;
chn.leftVol = chn.newLeftVol;
chn.rightVol = chn.newRightVol;
chn.rightRamp = chn.leftRamp = 0;
if(chn.dwFlags[CHN_NOTEFADE] && !chn.nFadeOutVol)
{
chn.nLength = 0;
chn.pCurrentSample = nullptr;
}
} else
{
chn.nRampLength -= nSmpCount;
}
}
const bool pastLoopEnd = chn.position.GetUInt() >= chn.nLoopEnd && chn.dwFlags[CHN_LOOP];
const bool pastSampleEnd = chn.position.GetUInt() >= chn.nLength && !chn.dwFlags[CHN_LOOP] && chn.nLength && !chn.nMasterChn;
const bool doSampleSwap = m_playBehaviour[kMODSampleSwap] && chn.nNewIns && chn.nNewIns <= GetNumSamples() && chn.pModSample != &Samples[chn.nNewIns];
if((pastLoopEnd || pastSampleEnd) && doSampleSwap)
{
// ProTracker compatibility: Instrument changes without a note do not happen instantly, but rather when the sample loop has finished playing.
// Test case: PTInstrSwap.mod, PTSwapNoLoop.mod
const ModSample &smp = Samples[chn.nNewIns];
chn.pModSample = &smp;
chn.pCurrentSample = smp.samplev();
chn.dwFlags = (chn.dwFlags & CHN_CHANNELFLAGS) | smp.uFlags;
chn.nLength = smp.uFlags[CHN_LOOP] ? smp.nLoopEnd : 0; // non-looping sample continue in oneshot mode (i.e. they will most probably just play silence)
chn.nLoopStart = smp.nLoopStart;
chn.nLoopEnd = smp.nLoopEnd;
chn.position.SetInt(chn.nLoopStart);
mixLoopState.UpdateLookaheadPointers(chn);
if(!chn.pCurrentSample)
{
break;
}
} else if(pastLoopEnd && !doSampleSwap && m_playBehaviour[kMODOneShotLoops] && chn.nLoopStart == 0)
{
// ProTracker "oneshot" loops (if loop start is 0, play the whole sample once and then repeat until loop end)
chn.position.SetInt(0);
chn.nLoopEnd = chn.nLength = chn.pModSample->nLoopEnd;
}
} while(nsamples > 0);
// Restore sample pointer in case it got changed through loop wrap-around
chn.pCurrentSample = mixLoopState.samplePointer;
nchmixed += naddmix;
#ifndef NO_PLUGINS
if(naddmix && nMixPlugin > 0 && nMixPlugin <= MAX_MIXPLUGINS && m_MixPlugins[nMixPlugin - 1].pMixPlugin)
{
m_MixPlugins[nMixPlugin - 1].pMixPlugin->ResetSilence();
}
#endif // NO_PLUGINS
}
m_nMixStat = std::max(m_nMixStat, nchmixed);
}
void CSoundFile::ProcessPlugins(uint32 nCount)
{
#ifndef NO_PLUGINS
// If any sample channels are active or any plugin has some input, possibly suspended master plugins need to be woken up.
bool masterHasInput = (m_nMixStat > 0);
#ifdef MPT_INTMIXER
const float IntToFloat = m_PlayConfig.getIntToFloat();
const float FloatToInt = m_PlayConfig.getFloatToInt();
#endif // MPT_INTMIXER
// Setup float inputs from samples
for(PLUGINDEX plug = 0; plug < MAX_MIXPLUGINS; plug++)
{
SNDMIXPLUGIN &plugin = m_MixPlugins[plug];
if(plugin.pMixPlugin != nullptr
&& plugin.pMixPlugin->m_MixState.pMixBuffer != nullptr
&& plugin.pMixPlugin->m_mixBuffer.Ok())
{
IMixPlugin *mixPlug = plugin.pMixPlugin;
SNDMIXPLUGINSTATE &state = mixPlug->m_MixState;
//We should only ever reach this point if the song is playing.
if (!mixPlug->IsSongPlaying())
{
//Plugin doesn't know it is in a song that is playing;
//we must have added it during playback. Initialise it!
mixPlug->NotifySongPlaying(true);
mixPlug->Resume();
}
// Setup float input
float *plugInputL = mixPlug->m_mixBuffer.GetInputBuffer(0);
float *plugInputR = mixPlug->m_mixBuffer.GetInputBuffer(1);
if (state.dwFlags & SNDMIXPLUGINSTATE::psfMixReady)
{
#ifdef MPT_INTMIXER
StereoMixToFloat(state.pMixBuffer, plugInputL, plugInputR, nCount, IntToFloat);
#else
DeinterleaveStereo(state.pMixBuffer, plugInputL, plugInputR, nCount);
#endif // MPT_INTMIXER
} else if (state.nVolDecayR || state.nVolDecayL)
{
StereoFill(state.pMixBuffer, nCount, state.nVolDecayR, state.nVolDecayL);
#ifdef MPT_INTMIXER
StereoMixToFloat(state.pMixBuffer, plugInputL, plugInputR, nCount, IntToFloat);
#else
DeinterleaveStereo(state.pMixBuffer, plugInputL, plugInputR, nCount);
#endif // MPT_INTMIXER
} else
{
memset(plugInputL, 0, nCount * sizeof(plugInputL[0]));
memset(plugInputR, 0, nCount * sizeof(plugInputR[0]));
}
state.dwFlags &= ~SNDMIXPLUGINSTATE::psfMixReady;
if(!plugin.IsMasterEffect() && !(state.dwFlags & SNDMIXPLUGINSTATE::psfSilenceBypass))
{
masterHasInput = true;
}
}
}
// Convert mix buffer
#ifdef MPT_INTMIXER
StereoMixToFloat(MixSoundBuffer, MixFloatBuffer[0], MixFloatBuffer[1], nCount, IntToFloat);
#else
DeinterleaveStereo(MixSoundBuffer, MixFloatBuffer[0], MixFloatBuffer[1], nCount);
#endif // MPT_INTMIXER
float *pMixL = MixFloatBuffer[0];
float *pMixR = MixFloatBuffer[1];
const bool positionChanged = HasPositionChanged();
// Process Plugins
for(PLUGINDEX plug = 0; plug < MAX_MIXPLUGINS; plug++)
{
SNDMIXPLUGIN &plugin = m_MixPlugins[plug];
if (plugin.pMixPlugin != nullptr
&& plugin.pMixPlugin->m_MixState.pMixBuffer != nullptr
&& plugin.pMixPlugin->m_mixBuffer.Ok())
{
IMixPlugin *pObject = plugin.pMixPlugin;
if(!plugin.IsMasterEffect() && !plugin.pMixPlugin->ShouldProcessSilence() && !(plugin.pMixPlugin->m_MixState.dwFlags & SNDMIXPLUGINSTATE::psfHasInput))
{
// If plugin has no inputs and isn't a master plugin, we shouldn't let it process silence if possible.
// I have yet to encounter a VST plugin which actually sets this flag.
bool hasInput = false;
for(PLUGINDEX inPlug = 0; inPlug < plug; inPlug++)
{
if(m_MixPlugins[inPlug].GetOutputPlugin() == plug)
{
hasInput = true;
break;
}
}
if(!hasInput)
{
continue;
}
}
bool isMasterMix = false;
float *plugInputL = pObject->m_mixBuffer.GetInputBuffer(0);
float *plugInputR = pObject->m_mixBuffer.GetInputBuffer(1);
if (pMixL == plugInputL)
{
isMasterMix = true;
pMixL = MixFloatBuffer[0];
pMixR = MixFloatBuffer[1];
}
SNDMIXPLUGINSTATE &state = plugin.pMixPlugin->m_MixState;
float *pOutL = pMixL;
float *pOutR = pMixR;
if (!plugin.IsOutputToMaster())
{
PLUGINDEX nOutput = plugin.GetOutputPlugin();
if(nOutput > plug && nOutput < MAX_MIXPLUGINS
&& m_MixPlugins[nOutput].pMixPlugin != nullptr)
{
IMixPlugin *outPlugin = m_MixPlugins[nOutput].pMixPlugin;
if(!(state.dwFlags & SNDMIXPLUGINSTATE::psfSilenceBypass)) outPlugin->ResetSilence();
if(outPlugin->m_mixBuffer.Ok())
{
pOutL = outPlugin->m_mixBuffer.GetInputBuffer(0);
pOutR = outPlugin->m_mixBuffer.GetInputBuffer(1);
}
}
}
/*
if (plugin.multiRouting) {
int nOutput=0;
for (int nOutput=0; nOutput < plugin.nOutputs / 2; nOutput++) {
destinationPlug = plugin.multiRoutingDestinations[nOutput];
pOutState = m_MixPlugins[destinationPlug].pMixState;
pOutputs[2 * nOutput] = plugInputL;
pOutputs[2 * (nOutput + 1)] = plugInputR;
}
}*/
if (plugin.IsMasterEffect())
{
if (!isMasterMix)
{
float *pInL = plugInputL;
float *pInR = plugInputR;
for (uint32 i=0; i<nCount; i++)
{
pInL[i] += pMixL[i];
pInR[i] += pMixR[i];
pMixL[i] = 0;
pMixR[i] = 0;
}
}
pMixL = pOutL;
pMixR = pOutR;
if(masterHasInput)
{
// Samples or plugins are being rendered, so turn off auto-bypass for this master effect.
if(plugin.pMixPlugin != nullptr) plugin.pMixPlugin->ResetSilence();
SNDMIXPLUGIN *chain = &plugin;
PLUGINDEX out = chain->GetOutputPlugin(), prevOut = plug;
while(out > prevOut && out < MAX_MIXPLUGINS)
{
chain = &m_MixPlugins[out];
prevOut = out;
out = chain->GetOutputPlugin();
if(chain->pMixPlugin)
{
chain->pMixPlugin->ResetSilence();
}
}
}
}
if(plugin.IsBypassed() || (plugin.IsAutoSuspendable() && (state.dwFlags & SNDMIXPLUGINSTATE::psfSilenceBypass)))
{
const float * const pInL = plugInputL;
const float * const pInR = plugInputR;
for (uint32 i=0; i<nCount; i++)
{
pOutL[i] += pInL[i];
pOutR[i] += pInR[i];
}
} else
{
if(positionChanged)
pObject->PositionChanged();
pObject->Process(pOutL, pOutR, nCount);
state.inputSilenceCount += nCount;
if(plugin.IsAutoSuspendable() && pObject->GetNumOutputChannels() > 0 && state.inputSilenceCount >= m_MixerSettings.gdwMixingFreq * 4)
{
bool isSilent = true;
for(uint32 i = 0; i < nCount; i++)
{
if(pOutL[i] >= FLT_EPSILON || pOutL[i] <= -FLT_EPSILON
|| pOutR[i] >= FLT_EPSILON || pOutR[i] <= -FLT_EPSILON)
{
isSilent = false;
break;
}
}
if(isSilent)
{
state.dwFlags |= SNDMIXPLUGINSTATE::psfSilenceBypass;
} else
{
state.inputSilenceCount = 0;
}
}
}
state.dwFlags &= ~SNDMIXPLUGINSTATE::psfHasInput;
}
}
#ifdef MPT_INTMIXER
FloatToStereoMix(pMixL, pMixR, MixSoundBuffer, nCount, FloatToInt);
#else
InterleaveStereo(pMixL, pMixR, MixSoundBuffer, nCount);
#endif // MPT_INTMIXER
#else
MPT_UNREFERENCED_PARAMETER(nCount);
#endif // NO_PLUGINS
}
OPENMPT_NAMESPACE_END