mirror of
https://github.com/WinampDesktop/winamp.git
synced 2024-12-19 18:25:58 +01:00
582 lines
17 KiB
C++
582 lines
17 KiB
C++
/*
|
|
* SamplEdit.cpp
|
|
* -------------
|
|
* Purpose: Basic sample editing code (resizing, adding silence, normalizing, ...).
|
|
* Notes : (currently none)
|
|
* Authors: OpenMPT Devs
|
|
* The OpenMPT source code is released under the BSD license. Read LICENSE for more details.
|
|
*/
|
|
|
|
|
|
#include "stdafx.h"
|
|
#include "SampleEdit.h"
|
|
#include "../soundlib/AudioCriticalSection.h"
|
|
#include "../soundlib/Sndfile.h"
|
|
#include "../soundlib/modsmp_ctrl.h"
|
|
#include "openmpt/soundbase/SampleConvert.hpp"
|
|
#include "openmpt/soundbase/SampleDecode.hpp"
|
|
#include "../soundlib/SampleCopy.h"
|
|
|
|
OPENMPT_NAMESPACE_BEGIN
|
|
|
|
namespace SampleEdit
|
|
{
|
|
|
|
std::vector<std::reference_wrapper<SmpLength>> GetCuesAndLoops(ModSample &smp)
|
|
{
|
|
std::vector<std::reference_wrapper<SmpLength>> loopPoints = {smp.nLoopStart, smp.nLoopEnd, smp.nSustainStart, smp.nSustainEnd};
|
|
loopPoints.insert(loopPoints.end(), std::begin(smp.cues), std::end(smp.cues));
|
|
return loopPoints;
|
|
}
|
|
|
|
|
|
SmpLength InsertSilence(ModSample &smp, const SmpLength silenceLength, const SmpLength startFrom, CSoundFile &sndFile)
|
|
{
|
|
if(silenceLength == 0 || silenceLength > MAX_SAMPLE_LENGTH || smp.nLength > MAX_SAMPLE_LENGTH - silenceLength || startFrom > smp.nLength)
|
|
return smp.nLength;
|
|
|
|
const bool wasEmpty = !smp.HasSampleData();
|
|
const SmpLength newLength = smp.nLength + silenceLength;
|
|
|
|
char *pNewSmp = static_cast<char *>(ModSample::AllocateSample(newLength, smp.GetBytesPerSample()));
|
|
if(pNewSmp == nullptr)
|
|
return smp.nLength; //Sample allocation failed.
|
|
|
|
if(!wasEmpty)
|
|
{
|
|
// Copy over old sample
|
|
const SmpLength silenceOffset = startFrom * smp.GetBytesPerSample();
|
|
const SmpLength silenceBytes = silenceLength * smp.GetBytesPerSample();
|
|
if(startFrom > 0)
|
|
{
|
|
memcpy(pNewSmp, smp.samplev(), silenceOffset);
|
|
}
|
|
if(startFrom < smp.nLength)
|
|
{
|
|
memcpy(pNewSmp + silenceOffset + silenceBytes, smp.sampleb() + silenceOffset, smp.GetSampleSizeInBytes() - silenceOffset);
|
|
}
|
|
|
|
// Update loop points if necessary.
|
|
for(SmpLength &point : GetCuesAndLoops(smp))
|
|
{
|
|
if(point >= startFrom) point += silenceLength;
|
|
}
|
|
} else
|
|
{
|
|
// Set loop points automatically
|
|
smp.nLoopStart = 0;
|
|
smp.nLoopEnd = newLength;
|
|
smp.uFlags.set(CHN_LOOP);
|
|
}
|
|
|
|
ctrlSmp::ReplaceSample(smp, pNewSmp, newLength, sndFile);
|
|
smp.PrecomputeLoops(sndFile, true);
|
|
|
|
return smp.nLength;
|
|
}
|
|
|
|
|
|
namespace
|
|
{
|
|
// Update loop points and cues after deleting a sample selection
|
|
void AdjustLoopPoints(SmpLength selStart, SmpLength selEnd, SmpLength &loopStart, SmpLength &loopEnd, SmpLength length)
|
|
{
|
|
Util::DeleteRange(selStart, selEnd - 1, loopStart, loopEnd);
|
|
|
|
LimitMax(loopEnd, length);
|
|
if(loopStart + 2 >= loopEnd)
|
|
{
|
|
loopStart = loopEnd = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
SmpLength RemoveRange(ModSample &smp, SmpLength selStart, SmpLength selEnd, CSoundFile &sndFile)
|
|
{
|
|
LimitMax(selEnd, smp.nLength);
|
|
if(selEnd <= selStart)
|
|
{
|
|
return smp.nLength;
|
|
}
|
|
const uint8 bps = smp.GetBytesPerSample();
|
|
memmove(smp.sampleb() + selStart * bps, smp.sampleb() + selEnd * bps, (smp.nLength - selEnd) * bps);
|
|
smp.nLength -= (selEnd - selStart);
|
|
|
|
// Did loops or cue points cover the deleted selection?
|
|
AdjustLoopPoints(selStart, selEnd, smp.nLoopStart, smp.nLoopEnd, smp.nLength);
|
|
AdjustLoopPoints(selStart, selEnd, smp.nSustainStart, smp.nSustainEnd, smp.nLength);
|
|
|
|
if(smp.nLoopEnd == 0) smp.uFlags.reset(CHN_LOOP | CHN_PINGPONGLOOP);
|
|
if(smp.nSustainEnd == 0) smp.uFlags.reset(CHN_SUSTAINLOOP | CHN_PINGPONGSUSTAIN);
|
|
|
|
for(auto &cue : smp.cues)
|
|
{
|
|
if(cue >= selEnd)
|
|
cue -= (selEnd - selStart);
|
|
else if(cue >= selStart && selStart == 0)
|
|
cue = smp.nLength;
|
|
else if(cue >= selStart)
|
|
cue = selStart;
|
|
}
|
|
|
|
smp.PrecomputeLoops(sndFile);
|
|
return smp.nLength;
|
|
}
|
|
|
|
|
|
SmpLength ResizeSample(ModSample &smp, const SmpLength newLength, CSoundFile &sndFile)
|
|
{
|
|
// Invalid sample size
|
|
if(newLength > MAX_SAMPLE_LENGTH || newLength == smp.nLength)
|
|
return smp.nLength;
|
|
|
|
// New sample will be bigger so we'll just use "InsertSilence" as it's already there.
|
|
if(newLength > smp.nLength)
|
|
return InsertSilence(smp, newLength - smp.nLength, smp.nLength, sndFile);
|
|
|
|
// Else: Shrink sample
|
|
|
|
const SmpLength newSmpBytes = newLength * smp.GetBytesPerSample();
|
|
|
|
void *newData = ModSample::AllocateSample(newLength, smp.GetBytesPerSample());
|
|
if(newData == nullptr && newLength > 0)
|
|
return smp.nLength; //Sample allocation failed.
|
|
|
|
// Copy over old data and replace sample by the new one
|
|
if(newData != nullptr)
|
|
memcpy(newData, smp.sampleb(), newSmpBytes);
|
|
ctrlSmp::ReplaceSample(smp, newData, newLength, sndFile);
|
|
|
|
// Sanitize loops and update loop wrap-around buffers
|
|
smp.PrecomputeLoops(sndFile);
|
|
|
|
return smp.nLength;
|
|
}
|
|
|
|
|
|
void ResetSamples(CSoundFile &sndFile, ResetFlag resetflag, SAMPLEINDEX minSample, SAMPLEINDEX maxSample)
|
|
{
|
|
if(minSample == SAMPLEINDEX_INVALID)
|
|
minSample = 1;
|
|
if(maxSample == SAMPLEINDEX_INVALID)
|
|
maxSample = sndFile.GetNumSamples();
|
|
Limit(minSample, SAMPLEINDEX(1), SAMPLEINDEX(MAX_SAMPLES - 1));
|
|
Limit(maxSample, SAMPLEINDEX(1), SAMPLEINDEX(MAX_SAMPLES - 1));
|
|
|
|
if(minSample > maxSample)
|
|
std::swap(minSample, maxSample);
|
|
|
|
for(SAMPLEINDEX i = minSample; i <= maxSample; i++)
|
|
{
|
|
ModSample &sample = sndFile.GetSample(i);
|
|
switch(resetflag)
|
|
{
|
|
case SmpResetInit:
|
|
sndFile.m_szNames[i] = "";
|
|
sample.filename = "";
|
|
sample.nC5Speed = 8363;
|
|
[[fallthrough]];
|
|
case SmpResetCompo:
|
|
sample.nPan = 128;
|
|
sample.nGlobalVol = 64;
|
|
sample.nVolume = 256;
|
|
sample.nVibDepth = 0;
|
|
sample.nVibRate = 0;
|
|
sample.nVibSweep = 0;
|
|
sample.nVibType = VIB_SINE;
|
|
sample.uFlags.reset(CHN_PANNING | SMP_NODEFAULTVOLUME);
|
|
break;
|
|
case SmpResetVibrato:
|
|
sample.nVibDepth = 0;
|
|
sample.nVibRate = 0;
|
|
sample.nVibSweep = 0;
|
|
sample.nVibType = VIB_SINE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
namespace
|
|
{
|
|
struct OffsetData
|
|
{
|
|
double max = 0.0, min = 0.0, offset = 0.0;
|
|
};
|
|
|
|
// Returns maximum sample amplitude for given sample type (int8/int16).
|
|
template <class T>
|
|
constexpr double GetMaxAmplitude() {return 1.0 + (std::numeric_limits<T>::max)();}
|
|
|
|
// Calculates DC offset and returns struct with DC offset, max and min values.
|
|
// DC offset value is average of [-1.0, 1.0[-normalized offset values.
|
|
template<class T>
|
|
OffsetData CalculateOffset(const T *pStart, const SmpLength length)
|
|
{
|
|
OffsetData offsetVals;
|
|
if(length < 1)
|
|
return offsetVals;
|
|
|
|
const double intToFloatScale = 1.0 / GetMaxAmplitude<T>();
|
|
double max = -1, min = 1, sum = 0;
|
|
|
|
const T *p = pStart;
|
|
for(SmpLength i = 0; i < length; i++, p++)
|
|
{
|
|
const double val = static_cast<double>(*p) * intToFloatScale;
|
|
sum += val;
|
|
if(val > max) max = val;
|
|
if(val < min) min = val;
|
|
}
|
|
|
|
offsetVals.max = max;
|
|
offsetVals.min = min;
|
|
offsetVals.offset = (-sum / (double)(length));
|
|
return offsetVals;
|
|
}
|
|
|
|
template <class T>
|
|
void RemoveOffsetAndNormalize(T *pStart, const SmpLength length, const double offset, const double amplify)
|
|
{
|
|
T *p = pStart;
|
|
for(SmpLength i = 0; i < length; i++, p++)
|
|
{
|
|
double var = (*p) * amplify + offset;
|
|
*p = mpt::saturate_round<T>(var);
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
// Remove DC offset
|
|
double RemoveDCOffset(ModSample &smp, SmpLength start, SmpLength end, CSoundFile &sndFile)
|
|
{
|
|
if(!smp.HasSampleData())
|
|
return 0;
|
|
|
|
if(end > smp.nLength) end = smp.nLength;
|
|
if(start > end) start = end;
|
|
if(start == end)
|
|
{
|
|
start = 0;
|
|
end = smp.nLength;
|
|
}
|
|
|
|
start *= smp.GetNumChannels();
|
|
end *= smp.GetNumChannels();
|
|
|
|
const double maxAmplitude = (smp.GetElementarySampleSize() == 2) ? GetMaxAmplitude<int16>() : GetMaxAmplitude<int8>();
|
|
|
|
// step 1: Calculate offset.
|
|
OffsetData oData;
|
|
if(smp.GetElementarySampleSize() == 2)
|
|
oData = CalculateOffset(smp.sample16() + start, end - start);
|
|
else if(smp.GetElementarySampleSize() == 1)
|
|
oData = CalculateOffset(smp.sample8() + start, end - start);
|
|
else
|
|
return 0;
|
|
|
|
double offset = oData.offset;
|
|
|
|
if((int)(offset * maxAmplitude) == 0)
|
|
return 0;
|
|
|
|
// those will be changed...
|
|
oData.max += offset;
|
|
oData.min += offset;
|
|
|
|
// ... and that might cause distortion, so we will normalize this.
|
|
const double amplify = 1 / std::max(oData.max, -oData.min);
|
|
|
|
// step 2: centralize + normalize sample
|
|
offset *= maxAmplitude * amplify;
|
|
if(smp.GetElementarySampleSize() == 2)
|
|
RemoveOffsetAndNormalize(smp.sample16() + start, end - start, offset, amplify);
|
|
else if(smp.GetElementarySampleSize() == 1)
|
|
RemoveOffsetAndNormalize(smp.sample8() + start, end - start, offset, amplify);
|
|
|
|
// step 3: adjust global vol (if available)
|
|
if((sndFile.GetType() & (MOD_TYPE_IT | MOD_TYPE_MPT)) && (start == 0) && (end == smp.nLength * smp.GetNumChannels()))
|
|
{
|
|
CriticalSection cs;
|
|
|
|
smp.nGlobalVol = std::min(mpt::saturate_round<uint16>(smp.nGlobalVol / amplify), uint16(64));
|
|
for(auto &chn : sndFile.m_PlayState.Chn)
|
|
{
|
|
if(chn.pModSample == &smp)
|
|
{
|
|
chn.UpdateInstrumentVolume(&smp, chn.pModInstrument);
|
|
}
|
|
}
|
|
}
|
|
|
|
smp.PrecomputeLoops(sndFile, false);
|
|
|
|
return oData.offset;
|
|
}
|
|
|
|
|
|
template<typename T>
|
|
static void ApplyAmplifyImpl(T * MPT_RESTRICT pSample, const SmpLength length, const double amplifyStart, const double amplifyEnd, const bool isFadeIn, const Fade::Law fadeLaw)
|
|
{
|
|
Fade::Func fadeFunc = Fade::GetFadeFunc(fadeLaw);
|
|
|
|
if(amplifyStart != amplifyEnd)
|
|
{
|
|
const double fadeOffset = isFadeIn ? amplifyStart : amplifyEnd;
|
|
const double fadeDiff = isFadeIn ? (amplifyEnd - amplifyStart) : (amplifyStart - amplifyEnd);
|
|
const double lengthInv = 1.0 / length;
|
|
for(SmpLength i = 0; i < length; i++)
|
|
{
|
|
const double amp = fadeOffset + fadeFunc(static_cast<double>(isFadeIn ? i : (length - i)) * lengthInv) * fadeDiff;
|
|
pSample[i] = mpt::saturate_round<T>(amp * pSample[i]);
|
|
}
|
|
} else
|
|
{
|
|
const double amp = fadeFunc(amplifyStart);
|
|
for(SmpLength i = 0; i < length; i++)
|
|
{
|
|
pSample[i] = mpt::saturate_round<T>(amp * pSample[i]);
|
|
}
|
|
}
|
|
}
|
|
|
|
bool AmplifySample(ModSample &smp, SmpLength start, SmpLength end, double amplifyStart, double amplifyEnd, bool isFadeIn, Fade::Law fadeLaw, CSoundFile &sndFile)
|
|
{
|
|
if(!smp.HasSampleData()) return false;
|
|
if(end == 0 || start >= end || end > smp.nLength)
|
|
{
|
|
start = 0;
|
|
end = smp.nLength;
|
|
}
|
|
|
|
if(end - start < 2) return false;
|
|
|
|
start *= smp.GetNumChannels();
|
|
end *= smp.GetNumChannels();
|
|
|
|
if (smp.GetElementarySampleSize() == 2)
|
|
ApplyAmplifyImpl(smp.sample16() + start, end - start, amplifyStart, amplifyEnd, isFadeIn, fadeLaw);
|
|
else if (smp.GetElementarySampleSize() == 1)
|
|
ApplyAmplifyImpl(smp.sample8() + start, end - start, amplifyStart, amplifyEnd, isFadeIn, fadeLaw);
|
|
else
|
|
return false;
|
|
|
|
smp.PrecomputeLoops(sndFile, false);
|
|
return true;
|
|
}
|
|
|
|
|
|
// Reverse sample data
|
|
bool ReverseSample(ModSample &smp, SmpLength start, SmpLength end, CSoundFile &sndFile)
|
|
{
|
|
return ctrlSmp::ReverseSample(smp, start, end, sndFile);
|
|
}
|
|
|
|
|
|
template <class T>
|
|
static void UnsignSampleImpl(T *pStart, const SmpLength length)
|
|
{
|
|
const T offset = (std::numeric_limits<T>::min)();
|
|
for(SmpLength i = 0; i < length; i++)
|
|
{
|
|
pStart[i] += offset;
|
|
}
|
|
}
|
|
|
|
// Virtually unsign sample data
|
|
bool UnsignSample(ModSample &smp, SmpLength start, SmpLength end, CSoundFile &sndFile)
|
|
{
|
|
if(!smp.HasSampleData()) return false;
|
|
if(end == 0 || start > smp.nLength || end > smp.nLength)
|
|
{
|
|
start = 0;
|
|
end = smp.nLength;
|
|
}
|
|
start *= smp.GetNumChannels();
|
|
end *= smp.GetNumChannels();
|
|
if(smp.GetElementarySampleSize() == 2)
|
|
UnsignSampleImpl(smp.sample16() + start, end - start);
|
|
else if(smp.GetElementarySampleSize() == 1)
|
|
UnsignSampleImpl(smp.sample8() + start, end - start);
|
|
else
|
|
return false;
|
|
|
|
smp.PrecomputeLoops(sndFile, false);
|
|
return true;
|
|
}
|
|
|
|
|
|
// Invert sample data (flip by 180 degrees)
|
|
bool InvertSample(ModSample &smp, SmpLength start, SmpLength end, CSoundFile &sndFile)
|
|
{
|
|
return ctrlSmp::InvertSample(smp, start, end, sndFile);
|
|
}
|
|
|
|
// Crossfade sample data to create smooth loops
|
|
bool XFadeSample(ModSample &smp, SmpLength fadeLength, int fadeLaw, bool afterloopFade, bool useSustainLoop, CSoundFile &sndFile)
|
|
{
|
|
return ctrlSmp::XFadeSample(smp, fadeLength, fadeLaw, afterloopFade, useSustainLoop, sndFile);
|
|
}
|
|
|
|
|
|
template <class T>
|
|
static void SilenceSampleImpl(T *p, SmpLength length, SmpLength inc, bool fromStart, bool toEnd)
|
|
{
|
|
const int dest = toEnd ? 0 : p[(length - 1) * inc];
|
|
const int base = fromStart ? 0 :p[0];
|
|
const int delta = dest - base;
|
|
const int64 len_m1 = length - 1;
|
|
for(SmpLength i = 0; i < length; i++)
|
|
{
|
|
int n = base + static_cast<int>((static_cast<int64>(delta) * static_cast<int64>(i)) / len_m1);
|
|
*p = static_cast<T>(n);
|
|
p += inc;
|
|
}
|
|
}
|
|
|
|
// Silence parts of the sample data
|
|
bool SilenceSample(ModSample &smp, SmpLength start, SmpLength end, CSoundFile &sndFile)
|
|
{
|
|
LimitMax(end, smp.nLength);
|
|
if(!smp.HasSampleData() || start >= end) return false;
|
|
|
|
const SmpLength length = end - start;
|
|
const bool fromStart = start == 0;
|
|
const bool toEnd = end == smp.nLength;
|
|
const uint8 numChn = smp.GetNumChannels();
|
|
|
|
for(uint8 chn = 0; chn < numChn; chn++)
|
|
{
|
|
if(smp.GetElementarySampleSize() == 2)
|
|
SilenceSampleImpl(smp.sample16() + start * numChn + chn, length, numChn, fromStart, toEnd);
|
|
else if(smp.GetElementarySampleSize() == 1)
|
|
SilenceSampleImpl(smp.sample8() + start * numChn + chn, length, numChn, fromStart, toEnd);
|
|
else
|
|
return false;
|
|
}
|
|
|
|
smp.PrecomputeLoops(sndFile, false);
|
|
return true;
|
|
}
|
|
|
|
|
|
template <class T>
|
|
static void StereoSepSampleImpl(T *p, SmpLength length, int32 separation)
|
|
{
|
|
const int32 fac1 = static_cast<int32>(32768 + separation / 2), fac2 = static_cast<int32>(32768 - separation / 2);
|
|
while(length--)
|
|
{
|
|
const int32 l = p[0], r = p[1];
|
|
p[0] = mpt::saturate_cast<T>((Util::mul32to64(l, fac1) + Util::mul32to64(r, fac2)) >> 16);
|
|
p[1] = mpt::saturate_cast<T>((Util::mul32to64(l, fac2) + Util::mul32to64(r, fac1)) >> 16);
|
|
p += 2;
|
|
}
|
|
}
|
|
|
|
// Change stereo separation
|
|
bool StereoSepSample(ModSample &smp, SmpLength start, SmpLength end, double separation, CSoundFile &sndFile)
|
|
{
|
|
LimitMax(end, smp.nLength);
|
|
if(!smp.HasSampleData() || start >= end || smp.GetNumChannels() != 2) return false;
|
|
|
|
const SmpLength length = end - start;
|
|
const uint8 numChn = smp.GetNumChannels();
|
|
const int32 sep32 = mpt::saturate_round<int32>(separation * (65536.0 / 100.0));
|
|
|
|
if(smp.GetElementarySampleSize() == 2)
|
|
StereoSepSampleImpl(smp.sample16() + start * numChn, length, sep32);
|
|
else if(smp.GetElementarySampleSize() == 1)
|
|
StereoSepSampleImpl(smp.sample8() + start * numChn, length, sep32);
|
|
else
|
|
return false;
|
|
|
|
smp.PrecomputeLoops(sndFile, false);
|
|
return true;
|
|
}
|
|
|
|
|
|
// Convert 16-bit sample to 8-bit
|
|
bool ConvertTo8Bit(ModSample &smp, CSoundFile &sndFile)
|
|
{
|
|
if(!smp.HasSampleData() || smp.GetElementarySampleSize() != 2)
|
|
return false;
|
|
|
|
CopySample<SC::ConversionChain<SC::Convert<int8, int16>, SC::DecodeIdentity<int16>>>(static_cast<int8 *>(smp.samplev()), smp.nLength * smp.GetNumChannels(), 1, smp.sample16(), smp.GetSampleSizeInBytes(), 1);
|
|
smp.uFlags.reset(CHN_16BIT);
|
|
for(auto &chn : sndFile.m_PlayState.Chn)
|
|
{
|
|
if(chn.pModSample == &smp)
|
|
chn.dwFlags.reset(CHN_16BIT);
|
|
}
|
|
|
|
smp.PrecomputeLoops(sndFile, false);
|
|
return true;
|
|
}
|
|
|
|
|
|
// Convert 8-bit sample to 16-bit
|
|
bool ConvertTo16Bit(ModSample &smp, CSoundFile &sndFile)
|
|
{
|
|
if(!smp.HasSampleData() || smp.GetElementarySampleSize() != 1)
|
|
return false;
|
|
|
|
int16 *newSample = static_cast<int16 *>(ModSample::AllocateSample(smp.nLength, 2 * smp.GetNumChannels()));
|
|
if(newSample == nullptr)
|
|
return false;
|
|
|
|
CopySample<SC::ConversionChain<SC::Convert<int16, int8>, SC::DecodeIdentity<int8>>>(newSample, smp.nLength * smp.GetNumChannels(), 1, smp.sample8(), smp.GetSampleSizeInBytes(), 1);
|
|
smp.uFlags.set(CHN_16BIT);
|
|
ctrlSmp::ReplaceSample(smp, newSample, smp.nLength, sndFile);
|
|
smp.PrecomputeLoops(sndFile, false);
|
|
return true;
|
|
}
|
|
|
|
|
|
template <class T>
|
|
static void ConvertPingPongLoopImpl(T *pStart, SmpLength length)
|
|
{
|
|
auto *out = pStart, *in = pStart;
|
|
while(length--)
|
|
{
|
|
*(out++) = *(--in);
|
|
}
|
|
}
|
|
|
|
// Convert ping-pong loops to regular loops
|
|
bool ConvertPingPongLoop(ModSample &smp, CSoundFile &sndFile, bool sustainLoop)
|
|
{
|
|
if(!smp.HasSampleData()
|
|
|| (!smp.HasPingPongLoop() && !sustainLoop)
|
|
|| (!smp.HasPingPongSustainLoop() && sustainLoop))
|
|
return false;
|
|
|
|
const SmpLength loopStart = sustainLoop ? smp.nSustainStart : smp.nLoopStart;
|
|
const SmpLength loopEnd = sustainLoop ? smp.nSustainEnd : smp.nLoopEnd;
|
|
const SmpLength oldLoopLength = loopEnd - loopStart;
|
|
const SmpLength oldLength = smp.nLength;
|
|
|
|
if(InsertSilence(smp, oldLoopLength, loopEnd, sndFile) <= oldLength)
|
|
return false;
|
|
|
|
static_assert(MaxSamplingPointSize <= 4);
|
|
if(smp.GetBytesPerSample() == 4) // 16 bit stereo
|
|
ConvertPingPongLoopImpl(static_cast<int32 *>(smp.samplev()) + loopEnd, oldLoopLength);
|
|
else if(smp.GetBytesPerSample() == 2) // 16 bit mono / 8 bit stereo
|
|
ConvertPingPongLoopImpl(static_cast<int16 *>(smp.samplev()) + loopEnd, oldLoopLength);
|
|
else if(smp.GetBytesPerSample() == 1) // 8 bit mono
|
|
ConvertPingPongLoopImpl(static_cast<int8 *>(smp.samplev()) + loopEnd, oldLoopLength);
|
|
else
|
|
return false;
|
|
|
|
smp.uFlags.reset(sustainLoop ? CHN_PINGPONGSUSTAIN : CHN_PINGPONGLOOP);
|
|
smp.PrecomputeLoops(sndFile, true);
|
|
return true;
|
|
}
|
|
|
|
} // namespace SampleEdit
|
|
|
|
OPENMPT_NAMESPACE_END
|