cog/Frameworks/OpenMPT/OpenMPT/soundlib/ModSample.cpp

580 lines
15 KiB
C++

/*
* ModSample.h
* -----------
* Purpose: Module Sample header class and helpers
* 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 "Sndfile.h"
#include "ModSample.h"
#include "modsmp_ctrl.h"
#include "mpt/base/numbers.hpp"
#include <cmath>
OPENMPT_NAMESPACE_BEGIN
// Translate sample properties between two given formats.
void ModSample::Convert(MODTYPE fromType, MODTYPE toType)
{
// Convert between frequency and transpose values if necessary.
if((!(toType & (MOD_TYPE_MOD | MOD_TYPE_XM))) && (fromType & (MOD_TYPE_MOD | MOD_TYPE_XM)))
{
TransposeToFrequency();
RelativeTone = 0;
nFineTune = 0;
// TransposeToFrequency assumes NTSC middle-C frequency like FT2, but we play MODs with PAL middle-C!
if(fromType == MOD_TYPE_MOD)
nC5Speed = Util::muldivr_unsigned(nC5Speed, 8272, 8363);
} else if((toType & (MOD_TYPE_MOD | MOD_TYPE_XM)) && (!(fromType & (MOD_TYPE_MOD | MOD_TYPE_XM))))
{
// FrequencyToTranspose assumes NTSC middle-C frequency like FT2, but we play MODs with PAL middle-C!
if(toType == MOD_TYPE_MOD)
nC5Speed = Util::muldivr_unsigned(nC5Speed, 8363, 8272);
FrequencyToTranspose();
}
// No ping-pong loop, panning and auto-vibrato for MOD / S3M samples
if(toType & (MOD_TYPE_MOD | MOD_TYPE_S3M))
{
uFlags.reset(CHN_PINGPONGLOOP | CHN_PANNING);
nVibDepth = 0;
nVibRate = 0;
nVibSweep = 0;
nVibType = VIB_SINE;
RelativeTone = 0;
}
// No global volume / sustain loops for MOD/S3M/XM
if(toType & (MOD_TYPE_MOD | MOD_TYPE_XM | MOD_TYPE_S3M))
{
nGlobalVol = 64;
// Sustain loops - convert to normal loops
if(uFlags[CHN_SUSTAINLOOP])
{
// We probably overwrite a normal loop here, but since sustain loops are evaluated before normal loops, this is just correct.
nLoopStart = nSustainStart;
nLoopEnd = nSustainEnd;
uFlags.set(CHN_LOOP);
uFlags.set(CHN_PINGPONGLOOP, uFlags[CHN_PINGPONGSUSTAIN]);
}
nSustainStart = nSustainEnd = 0;
uFlags.reset(CHN_SUSTAINLOOP | CHN_PINGPONGSUSTAIN);
}
// All XM samples have default panning, and XM's autovibrato settings are rather limited.
if(toType & MOD_TYPE_XM)
{
if(!uFlags[CHN_PANNING])
{
uFlags.set(CHN_PANNING);
nPan = 128;
}
LimitMax(nVibDepth, uint8(15));
LimitMax(nVibRate, uint8(63));
}
// Autovibrato sweep setting is inverse in XM (0 = "no sweep") and IT (0 = "no vibrato")
if(((fromType & MOD_TYPE_XM) && (toType & (MOD_TYPE_IT | MOD_TYPE_MPT))) || ((toType & MOD_TYPE_XM) && (fromType & (MOD_TYPE_IT | MOD_TYPE_MPT))))
{
if(nVibRate != 0 && nVibDepth != 0)
{
if(nVibSweep != 0)
nVibSweep = mpt::saturate_cast<decltype(nVibSweep)>(Util::muldivr_unsigned(nVibDepth, 256, nVibSweep));
else
nVibSweep = 255;
}
}
// Convert incompatible autovibrato types
if(toType == MOD_TYPE_IT && nVibType == VIB_RAMP_UP)
{
nVibType = VIB_RAMP_DOWN;
} else if(toType == MOD_TYPE_XM && nVibType == VIB_RANDOM)
{
nVibType = VIB_SINE;
}
// No external samples in formats other than MPTM.
if(toType != MOD_TYPE_MPT)
{
uFlags.reset(SMP_KEEPONDISK);
}
// No Adlib instruments in formats that can't handle it.
if(!CSoundFile::SupportsOPL(toType) && uFlags[CHN_ADLIB])
{
SetAdlib(false);
} else if(toType == MOD_TYPE_S3M && uFlags[CHN_ADLIB])
{
// No support for OPL3 waveforms in S3M
adlib[8] &= 0x03;
adlib[9] &= 0x03;
}
}
// Initialize sample slot with default values.
void ModSample::Initialize(MODTYPE type)
{
FreeSample();
nLength = 0;
nLoopStart = nLoopEnd = 0;
nSustainStart = nSustainEnd = 0;
nC5Speed = 8363;
nPan = 128;
nVolume = 256;
nGlobalVol = 64;
uFlags.reset(CHN_PANNING | CHN_SUSTAINLOOP | CHN_LOOP | CHN_PINGPONGLOOP | CHN_PINGPONGSUSTAIN | CHN_ADLIB | SMP_MODIFIED | SMP_KEEPONDISK);
if(type == MOD_TYPE_XM)
{
uFlags.set(CHN_PANNING);
}
RelativeTone = 0;
nFineTune = 0;
nVibType = VIB_SINE;
nVibSweep = 0;
nVibDepth = 0;
nVibRate = 0;
rootNote = 0;
filename = "";
RemoveAllCuePoints();
}
// Returns sample rate of the sample.
uint32 ModSample::GetSampleRate(const MODTYPE type) const
{
uint32 rate;
if(CSoundFile::UseFinetuneAndTranspose(type))
rate = TransposeToFrequency(RelativeTone, nFineTune);
else
rate = nC5Speed;
// TransposeToFrequency assumes NTSC middle-C frequency like FT2, but we play MODs with PAL middle-C!
if(type == MOD_TYPE_MOD)
rate = Util::muldivr_unsigned(rate, 8272, 8363);
return (rate > 0) ? rate : 8363;
}
// Copies sample data from another sample slot and ensures that the 16-bit/stereo flags are set accordingly.
bool ModSample::CopyWaveform(const ModSample &smpFrom)
{
if(!smpFrom.HasSampleData())
return false;
// If we duplicate a sample slot, avoid deleting the sample we just copy from
if(smpFrom.sampleb() == sampleb())
pData.pSample = nullptr;
LimitMax(nLength, smpFrom.nLength);
uFlags.set(CHN_16BIT, smpFrom.uFlags[CHN_16BIT]);
uFlags.set(CHN_STEREO, smpFrom.uFlags[CHN_STEREO]);
if(AllocateSample())
{
memcpy(sampleb(), smpFrom.sampleb(), GetSampleSizeInBytes());
return true;
}
return false;
}
// Allocate sample based on a ModSample's properties.
// Returns number of bytes allocated, 0 on failure.
size_t ModSample::AllocateSample()
{
FreeSample();
if((pData.pSample = AllocateSample(nLength, GetBytesPerSample())) == nullptr)
{
return 0;
} else
{
return GetSampleSizeInBytes();
}
}
// Allocate sample memory. On success, a pointer to the silenced sample buffer is returned. On failure, nullptr is returned.
// numFrames must contain the sample length, bytesPerSample the size of a sampling point multiplied with the number of channels.
void *ModSample::AllocateSample(SmpLength numFrames, size_t bytesPerSample)
{
const size_t allocSize = GetRealSampleBufferSize(numFrames, bytesPerSample);
if(allocSize != 0)
{
char *p = new(std::nothrow) char[allocSize];
if(p != nullptr)
{
memset(p, 0, allocSize);
return p + (InterpolationLookaheadBufferSize * MaxSamplingPointSize);
}
}
return nullptr;
}
// Compute sample buffer size in bytes, including any overhead introduced by pre-computed loops and such. Returns 0 if sample is too big.
size_t ModSample::GetRealSampleBufferSize(SmpLength numSamples, size_t bytesPerSample)
{
// Number of required lookahead samples:
// * 1x InterpolationMaxLookahead samples before the actual sample start. This is set to MaxSamplingPointSize due to the way AllocateSample/FreeSample currently work.
// * 1x InterpolationMaxLookahead samples of silence after the sample end (if normal loop end == sample end, this can be optimized out).
// * 2x InterpolationMaxLookahead before the loop point (because we start at InterpolationMaxLookahead before the loop point and will look backwards from there as well)
// * 2x InterpolationMaxLookahead after the loop point (for wrap-around)
// * 4x InterpolationMaxLookahead for the sustain loop (same as the two points above)
const SmpLength maxSize = Util::MaxValueOfType(numSamples);
const SmpLength lookaheadBufferSize = (MaxSamplingPointSize + 1 + 4 + 4) * InterpolationLookaheadBufferSize;
if(numSamples == 0 || numSamples > MAX_SAMPLE_LENGTH || lookaheadBufferSize > maxSize - numSamples)
{
return 0;
}
numSamples += lookaheadBufferSize;
if(maxSize / bytesPerSample < numSamples)
{
return 0;
}
return numSamples * bytesPerSample;
}
void ModSample::FreeSample()
{
FreeSample(pData.pSample);
pData.pSample = nullptr;
}
void ModSample::FreeSample(void *samplePtr)
{
if(samplePtr)
{
delete[](((char *)samplePtr) - (InterpolationLookaheadBufferSize * MaxSamplingPointSize));
}
}
// Set loop points and update loop wrap-around buffer
void ModSample::SetLoop(SmpLength start, SmpLength end, bool enable, bool pingpong, CSoundFile &sndFile)
{
nLoopStart = start;
nLoopEnd = end;
LimitMax(nLoopEnd, nLength);
if(nLoopStart < nLoopEnd)
{
uFlags.set(CHN_LOOP, enable);
uFlags.set(CHN_PINGPONGLOOP, pingpong && enable);
} else
{
nLoopStart = nLoopEnd = 0;
uFlags.reset(CHN_LOOP | CHN_PINGPONGLOOP);
}
PrecomputeLoops(sndFile, true);
}
// Set sustain loop points and update loop wrap-around buffer
void ModSample::SetSustainLoop(SmpLength start, SmpLength end, bool enable, bool pingpong, CSoundFile &sndFile)
{
nSustainStart = start;
nSustainEnd = end;
LimitMax(nLoopEnd, nLength);
if(nSustainStart < nSustainEnd)
{
uFlags.set(CHN_SUSTAINLOOP, enable);
uFlags.set(CHN_PINGPONGSUSTAIN, pingpong && enable);
} else
{
nSustainStart = nSustainEnd = 0;
uFlags.reset(CHN_SUSTAINLOOP | CHN_PINGPONGSUSTAIN);
}
PrecomputeLoops(sndFile, true);
}
namespace // Unnamed namespace for local implementation functions.
{
template <typename T>
class PrecomputeLoop
{
protected:
T *target;
const T *sampleData;
SmpLength loopEnd;
int numChannels;
bool pingpong;
bool ITPingPongMode;
public:
PrecomputeLoop(T *target, const T *sampleData, SmpLength loopEnd, int numChannels, bool pingpong, bool ITPingPongMode)
: target(target), sampleData(sampleData), loopEnd(loopEnd), numChannels(numChannels), pingpong(pingpong), ITPingPongMode(ITPingPongMode)
{
if(loopEnd > 0)
{
CopyLoop(true);
CopyLoop(false);
}
}
void CopyLoop(bool direction) const
{
// Direction: true = start reading and writing forward, false = start reading and writing backward (write direction never changes)
const int numSamples = 2 * InterpolationLookaheadBufferSize + (direction ? 1 : 0); // Loop point is included in forward loop expansion
T *dest = target + numChannels * (2 * InterpolationLookaheadBufferSize - 1); // Write buffer offset
SmpLength readPosition = loopEnd - 1;
const int writeIncrement = direction ? 1 : -1;
int readIncrement = writeIncrement;
for(int i = 0; i < numSamples; i++)
{
// Copy sample over to lookahead buffer
for(int c = 0; c < numChannels; c++)
{
dest[c] = sampleData[readPosition * numChannels + c];
}
dest += writeIncrement * numChannels;
if(readPosition == loopEnd - 1 && readIncrement > 0)
{
// Reached end of loop while going forward
if(pingpong)
{
readIncrement = -1;
if(ITPingPongMode && readPosition > 0)
{
readPosition--;
}
} else
{
readPosition = 0;
}
} else if(readPosition == 0 && readIncrement < 0)
{
// Reached start of loop while going backward
if(pingpong)
{
readIncrement = 1;
} else
{
readPosition = loopEnd - 1;
}
} else
{
readPosition += readIncrement;
}
}
}
};
template <typename T>
void PrecomputeLoopsImpl(ModSample &smp, const CSoundFile &sndFile)
{
const int numChannels = smp.GetNumChannels();
const int copySamples = numChannels * InterpolationLookaheadBufferSize;
T *sampleData = static_cast<T *>(smp.samplev());
T *afterSampleStart = sampleData + smp.nLength * numChannels;
T *loopLookAheadStart = afterSampleStart + copySamples;
T *sustainLookAheadStart = loopLookAheadStart + 4 * copySamples;
// Hold sample on the same level as the last sampling point at the end to prevent extra pops with interpolation.
// Do the same at the sample start, too.
for(int i = 0; i < (int)InterpolationLookaheadBufferSize; i++)
{
for(int c = 0; c < numChannels; c++)
{
afterSampleStart[i * numChannels + c] = afterSampleStart[-numChannels + c];
sampleData[-(i + 1) * numChannels + c] = sampleData[c];
}
}
if(smp.uFlags[CHN_LOOP])
{
PrecomputeLoop<T>(loopLookAheadStart,
sampleData + smp.nLoopStart * numChannels,
smp.nLoopEnd - smp.nLoopStart,
numChannels,
smp.uFlags[CHN_PINGPONGLOOP],
sndFile.m_playBehaviour[kITPingPongMode]);
}
if(smp.uFlags[CHN_SUSTAINLOOP])
{
PrecomputeLoop<T>(sustainLookAheadStart,
sampleData + smp.nSustainStart * numChannels,
smp.nSustainEnd - smp.nSustainStart,
numChannels,
smp.uFlags[CHN_PINGPONGSUSTAIN],
sndFile.m_playBehaviour[kITPingPongMode]);
}
}
} // unnamed namespace
void ModSample::PrecomputeLoops(CSoundFile &sndFile, bool updateChannels)
{
if(!HasSampleData())
return;
SanitizeLoops();
// Update channels with possibly changed loop values
if(updateChannels)
{
ctrlSmp::UpdateLoopPoints(*this, sndFile);
}
if(GetElementarySampleSize() == 2)
PrecomputeLoopsImpl<int16>(*this, sndFile);
else if(GetElementarySampleSize() == 1)
PrecomputeLoopsImpl<int8>(*this, sndFile);
}
// Remove loop points if they're invalid.
void ModSample::SanitizeLoops()
{
LimitMax(nSustainEnd, nLength);
LimitMax(nLoopEnd, nLength);
if(nSustainStart >= nSustainEnd)
{
nSustainStart = nSustainEnd = 0;
uFlags.reset(CHN_SUSTAINLOOP | CHN_PINGPONGSUSTAIN);
}
if(nLoopStart >= nLoopEnd)
{
nLoopStart = nLoopEnd = 0;
uFlags.reset(CHN_LOOP | CHN_PINGPONGLOOP);
}
}
/////////////////////////////////////////////////////////////
// Transpose <-> Frequency conversions
uint32 ModSample::TransposeToFrequency(int transpose, int finetune)
{
return mpt::saturate_round<uint32>(std::pow(2.0, (transpose * 128.0 + finetune) * (1.0 / (12.0 * 128.0))) * 8363.0);
}
void ModSample::TransposeToFrequency()
{
nC5Speed = TransposeToFrequency(RelativeTone, nFineTune);
}
// Return a pair of {transpose, finetune}
std::pair<int8, int8> ModSample::FrequencyToTranspose(uint32 freq)
{
if(!freq)
return {};
const auto f2t = mpt::saturate_round<int32>(std::log(freq * (1.0 / 8363.0)) * (12.0 * 128.0 * (1.0 / mpt::numbers::ln2)));
const auto fine = std::div(Clamp(f2t, -16384, 16383), int32(128));
return {static_cast<int8>(fine.quot), static_cast<int8>(fine.rem)};
}
void ModSample::FrequencyToTranspose()
{
std::tie(RelativeTone, nFineTune) = FrequencyToTranspose(nC5Speed);
}
// Transpose the sample by amount specified in octaves (i.e. amount=1 transposes one octave up)
void ModSample::Transpose(double amount)
{
nC5Speed = mpt::saturate_round<uint32>(nC5Speed * std::pow(2.0, amount));
}
// Check if the sample has any valid cue points
bool ModSample::HasAnyCuePoints() const
{
if(uFlags[CHN_ADLIB])
return false;
for(auto pt : cues)
{
if(pt < nLength)
return true;
}
return false;
}
// Check if the sample's cue points are the default cue point set.
bool ModSample::HasCustomCuePoints() const
{
if(uFlags[CHN_ADLIB])
return false;
for(SmpLength i = 0; i < std::size(cues); i++)
{
if(cues[i] != (i + 1) << 11)
return true;
}
return false;
}
void ModSample::SetDefaultCuePoints()
{
// Default cues compatible with old-style volume column offset
for(int i = 0; i < 9; i++)
{
cues[i] = (i + 1) << 11;
}
}
void ModSample::Set16BitCuePoints()
{
// Cue points that are useful for extending regular offset command
for(int i = 0; i < 9; i++)
{
cues[i] = (i + 1) << 16;
}
}
void ModSample::RemoveAllCuePoints()
{
if(!uFlags[CHN_ADLIB])
cues.fill(MAX_SAMPLE_LENGTH);
}
void ModSample::SetAdlib(bool enable, OPLPatch patch)
{
if(!enable && uFlags[CHN_ADLIB])
{
SetDefaultCuePoints();
}
uFlags.set(CHN_ADLIB, enable);
if(enable)
{
// Bogus sample to make playback work
uFlags.reset(CHN_16BIT | CHN_STEREO);
nLength = 4;
AllocateSample();
adlib = patch;
}
}
OPENMPT_NAMESPACE_END