bool EffectAmplify::ProcessOne(int count, WaveTrack *t,
sampleCount start, sampleCount len)
{
sampleCount s = start;
sampleCount originalLen = len;
sampleCount blockSize = t->GetMaxBlockSize();
sampleType *buffer = new sampleType[blockSize];
while (len) {
unsigned int block = t->GetBestBlockSize(s);
if (block > len)
block = len;
t->Get(buffer, s, block);
for (unsigned int i = 0; i < block; i++) {
buffer[i] = (sampleType) (buffer[i] * ratio);
}
t->Set(buffer, s, block);
len -= block;
s += block;
if (TrackProgress(count, (s-start)/(double)originalLen))
break;
}
delete[] buffer;
return true;
}
bool EffectInvert::ProcessOne(int count, WaveTrack *t,
sampleCount start, sampleCount len)
{
// keep track of two blocks whose data we will swap
sampleCount s = start;
sampleCount originalLen = len;
sampleCount blockSize = t->GetMaxBlockSize();
sampleType *buffer = new sampleType[blockSize];
while (len > 0) {
unsigned int block = t->GetBestBlockSize(s);
if (block > len)
block = len;
t->Get(buffer, s, block);
for (unsigned int i = 0; i < block; i++) {
buffer[i] = (sampleType) (-buffer[i]);
}
t->Set(buffer, s, block);
len -= block;
s += block;
TrackProgress(count, (s-start)/(double)originalLen);
}
delete[] buffer;
return true;
}
bool BlockGenerator::GenerateTrack(WaveTrack *tmp,
const WaveTrack &track,
int ntrack)
{
bool bGoodResult = true;
numSamples = track.TimeToLongSamples(GetDuration());
sampleCount i = 0;
float *data = new float[tmp->GetMaxBlockSize()];
sampleCount block = 0;
while ((i < numSamples) && bGoodResult) {
block = tmp->GetBestBlockSize(i);
if (block > (numSamples - i))
block = numSamples - i;
GenerateBlock(data, track, block);
// Add the generated data to the temporary track
tmp->Append((samplePtr)data, floatSample, block);
i += block;
// Update the progress meter
if (TrackProgress(ntrack, (double)i / numSamples))
bGoodResult = false;
}
delete[] data;
return bGoodResult;
}
bool EffectRepeat::Process()
{
this->CopyInputWaveTracks(); // Set up mOutputWaveTracks.
bool bGoodResult = true;
TrackListIterator iter(mOutputWaveTracks);
WaveTrack *track = (WaveTrack *) iter.First();
int nTrack = 0;
double maxDestLen = 0.0; // used to change selection to generated bit
while ((track != NULL) && bGoodResult) {
double trackStart = track->GetStartTime();
double trackEnd = track->GetEndTime();
double t0 = mT0 < trackStart? trackStart: mT0;
double t1 = mT1 > trackEnd? trackEnd: mT1;
if (t1 <= t0)
continue;
sampleCount start = track->TimeToLongSamples(t0);
sampleCount end = track->TimeToLongSamples(t1);
sampleCount len = (sampleCount)(end - start);
double tLen = track->LongSamplesToTime(len);
double tc = t0 + tLen;
if (len <= 0)
continue;
Track *dest;
track->Copy(t0, t1, &dest);
for(int j=0; j<repeatCount; j++)
{
if (!track->Paste(tc, dest) ||
TrackProgress(nTrack, j / repeatCount)) // TrackProgress returns true on Cancel.
{
bGoodResult = false;
break;
}
tc += tLen;
}
if (tc > maxDestLen)
maxDestLen = tc;
delete dest;
track = (WaveTrack *) iter.Next();
nTrack++;
}
if (bGoodResult)
{
// Change selection to just the generated bits.
mT0 = mT1;
mT1 = maxDestLen;
}
this->ReplaceProcessedWaveTracks(bGoodResult);
return bGoodResult;
}
bool EffectClickRemoval::ProcessOne(int count, WaveTrack * track,
sampleCount start, sampleCount len)
{
bool rc = true;
sampleCount s = 0;
sampleCount idealBlockLen = track->GetMaxBlockSize() * 4;
if (idealBlockLen % windowSize != 0)
idealBlockLen += (windowSize - (idealBlockLen % windowSize));
float *buffer = new float[idealBlockLen];
float *datawindow = new float[windowSize];
int i;
while((s < len)&&((len-s)>(windowSize/2))) {
sampleCount block = idealBlockLen;
if (s + block > len)
block = len - s;
track->Get((samplePtr) buffer, floatSample, start + s, block);
for(i=0; i<(block-windowSize/2); i+=windowSize/2) {
int wcopy = windowSize;
if (i + wcopy > block)
wcopy = block - i;
int j;
for(j=0; j<wcopy; j++)
datawindow[j] = buffer[i+j];
for(j=wcopy; j<windowSize; j++)
datawindow[j] = 0;
RemoveClicks(windowSize, datawindow);
for(j=0; j<wcopy; j++)
buffer[i+j] = datawindow[j];
}
track->Set((samplePtr) buffer, floatSample, start + s, block);
s += block;
if (TrackProgress(count, s / (double) len)) {
rc = false;
break;
}
}
delete[] buffer;
delete[] datawindow;
return rc;
}
//ProcessOne() takes a track, transforms it to bunch of buffer-blocks,
//and executes ProcessData, on it...
// uses mMult and mOffset to normalize a track. Needs to have them set before being called
bool EffectNormalize::ProcessOne(WaveTrack * track, wxString msg)
{
bool rc = true;
sampleCount s;
//Transform the marker timepoints to samples
sampleCount start = track->TimeToLongSamples(mCurT0);
sampleCount end = track->TimeToLongSamples(mCurT1);
//Get the length of the buffer (as double). len is
//used simply to calculate a progress meter, so it is easier
//to make it a double now than it is to do it later
double len = (double)(end - start);
//Initiate a processing buffer. This buffer will (most likely)
//be shorter than the length of the track being processed.
float *buffer = new float[track->GetMaxBlockSize()];
//Go through the track one buffer at a time. s counts which
//sample the current buffer starts at.
s = start;
while (s < end) {
//Get a block of samples (smaller than the size of the buffer)
sampleCount block = track->GetBestBlockSize(s);
//Adjust the block size if it is the final block in the track
if (s + block > end)
block = end - s;
//Get the samples from the track and put them in the buffer
track->Get((samplePtr) buffer, floatSample, s, block);
//Process the buffer.
ProcessData(buffer, block);
//Copy the newly-changed samples back onto the track.
track->Set((samplePtr) buffer, floatSample, s, block);
//Increment s one blockfull of samples
s += block;
//Update the Progress meter
if (TrackProgress(mCurTrackNum,
0.5+((double)(s - start) / len)/2.0, msg)) {
rc = false; //lda .. break, not return, so that buffer is deleted
break;
}
}
//Clean up the buffer
delete[] buffer;
//Return true because the effect processing succeeded ... unless cancelled
return rc;
}
//ProcessOne() takes a track, transforms it to bunch of buffer-blocks,
//and executes ProcessSimpleMono on these blocks
bool EffectSimpleMono::ProcessOne(WaveTrack * track,
longSampleCount start, longSampleCount end)
{
longSampleCount s;
//Get the length of the buffer (as double). len is
//used simple to calculate a progress meter, so it is easier
//to make it a double now than it is to do it later
double len = (double)(end - start);
//Initiate a processing buffer. This buffer will (most likely)
//be shorter than the length of the track being processed.
float *buffer = new float[track->GetMaxBlockSize()];
//Go through the track one buffer at a time. s counts which
//sample the current buffer starts at.
s = start;
while (s < end) {
//Get a block of samples (smaller than the size of the buffer)
sampleCount block = track->GetBestBlockSize(s);
//Adjust the block size if it is the final block in the track
if (s + block > end)
block = end - s;
//Get the samples from the track and put them in the buffer
track->Get((samplePtr) buffer, floatSample, s, block);
//Process the buffer. If it fails, clean up and exit.
if (!ProcessSimpleMono(buffer, block)) {
delete[]buffer;
//Return false because the effect failed.
return false;
}
//Processing succeeded. copy the newly-changed samples back
//onto the track.
track->Set((samplePtr) buffer, floatSample, s, block);
//Increment s one blockfull of samples
s += block;
//Update the Progress meter
if (TrackProgress(mCurTrackNum, s / len))
return false;
}
//Clean up the buffer
delete[]buffer;
//Return true because the effect processing succeeded.
return true;
}
bool EffectRepair::ProcessOne(int count, WaveTrack * track,
sampleCount start,
size_t len,
size_t repairStart, size_t repairLen)
{
Floats buffer{ len };
track->Get((samplePtr) buffer.get(), floatSample, start, len);
InterpolateAudio(buffer.get(), len, repairStart, repairLen);
track->Set((samplePtr)&buffer[repairStart], floatSample,
start + repairStart, repairLen);
return !TrackProgress(count, 1.0); // TrackProgress returns true on Cancel.
}
bool EffectEcho::ProcessOne(int count, WaveTrack * track,
sampleCount start, sampleCount len)
{
sampleCount s = 0;
sampleCount blockSize = (sampleCount) (track->GetRate() * delay);
//do nothing if the delay is less than 1 sample or greater than
//the length of the selection
if (blockSize < 1 || blockSize > len)
return true;
float *buffer0 = new float[blockSize];
float *buffer1 = new float[blockSize];
float *ptr0 = buffer0;
float *ptr1 = buffer1;
bool first = true;
while (s < len) {
sampleCount block = blockSize;
if (s + block > len)
block = len - s;
track->Get((samplePtr)ptr0, floatSample, start + s, block);
if (!first) {
for (sampleCount i = 0; i < block; i++)
ptr0[i] += ptr1[i] * decay;
track->Set((samplePtr)ptr0, floatSample, start + s, block);
}
float *ptrtemp = ptr0;
ptr0 = ptr1;
ptr1 = ptrtemp;
first = false;
s += block;
if (TrackProgress(count, s / (double) len)) {
delete[]buffer0;
delete[]buffer1;
return false;
}
}
delete[]buffer0;
delete[]buffer1;
return true;
}
bool EffectRepair::ProcessOne(int count, WaveTrack * track,
sampleCount start,
sampleCount len,
sampleCount repairStart, sampleCount repairLen)
{
float *buffer = new float[len];
track->Get((samplePtr) buffer, floatSample, start, len);
InterpolateAudio(buffer, len, repairStart, repairLen);
track->Set((samplePtr)&buffer[repairStart], floatSample,
start + repairStart, repairLen);
delete[] buffer;
return !TrackProgress(count, 1.0); // TrackProgress returns true on Cancel.
}
bool EffectEcho::ProcessOne(int count, WaveTrack * t,
sampleCount start, sampleCount len)
{
sampleCount s = start;
sampleCount blockSize = (sampleCount) (t->rate * delay);
sampleCount originalLen = len;
if (blockSize < 1 || blockSize > len)
return true;
sampleType *buffer0 = new sampleType[blockSize];
sampleType *buffer1 = new sampleType[blockSize];
sampleType *ptr0 = buffer0;
sampleType *ptr1 = buffer1;
bool first = true;
while (len) {
sampleCount block = blockSize;
if (block > len)
block = len;
t->Get(ptr0, s, block);
if (!first) {
for (sampleCount i = 0; i < block; i++)
ptr0[i] += (sampleType) (ptr1[i] * decay);
t->Set(ptr0, s, block);
}
sampleType *ptrtemp = ptr0;
ptr0 = ptr1;
ptr1 = ptrtemp;
first = false;
len -= block;
s += block;
TrackProgress(count, (s-start)/(double)originalLen);
}
delete[]buffer0;
delete[]buffer1;
return true;
}
bool EffectStereoToMono::ProcessOne(int count)
{
float curLeftFrame;
float curRightFrame;
float curMonoFrame;
sampleCount idealBlockLen = mLeftTrack->GetMaxBlockSize() * 2;
sampleCount index = mStart;
float *leftBuffer = new float[idealBlockLen];
float *rightBuffer = new float[idealBlockLen];
bool bResult = true;
while (index < mEnd) {
bResult &= mLeftTrack->Get((samplePtr)leftBuffer, floatSample, index, idealBlockLen);
bResult &= mRightTrack->Get((samplePtr)rightBuffer, floatSample, index, idealBlockLen);
sampleCount limit = idealBlockLen;
if ((index + idealBlockLen) > mEnd) {
limit = mEnd - index;
}
for (sampleCount i = 0; i < limit; ++i) {
index++;
curLeftFrame = leftBuffer[i];
curRightFrame = rightBuffer[i];
curMonoFrame = (curLeftFrame + curRightFrame) / 2.0;
leftBuffer[i] = curMonoFrame;
}
bResult &= mOutTrack->Append((samplePtr)leftBuffer, floatSample, limit);
if (TrackProgress(count, 2.*((double)index / (double)(mEnd - mStart))))
return false;
}
double minStart = wxMin(mLeftTrack->GetStartTime(), mRightTrack->GetStartTime());
bResult &= mLeftTrack->Clear(mLeftTrack->GetStartTime(), mLeftTrack->GetEndTime());
bResult &= mOutTrack->Flush();
bResult &= mLeftTrack->Paste(minStart, mOutTrack);
mLeftTrack->SetLinked(false);
mRightTrack->SetLinked(false);
mLeftTrack->SetChannel(Track::MonoChannel);
mOutputTracks->Remove(mRightTrack);
delete mRightTrack;
delete [] leftBuffer;
delete [] rightBuffer;
return bResult;
}
bool EffectReverse::ProcessOneClip(int count, WaveTrack *track,
sampleCount start, sampleCount len,
sampleCount originalStart, sampleCount originalEnd)
{
bool rc = true;
// keep track of two blocks whose data we will swap
sampleCount first = start;
sampleCount second;
sampleCount blockSize = track->GetMaxBlockSize();
float tmp;
float *buffer1 = new float[blockSize];
float *buffer2 = new float[blockSize];
sampleCount originalLen = (sampleCount)originalEnd-originalStart;
while (len > 1) {
sampleCount block = track->GetBestBlockSize(first);
if (block > len / 2)
block = len / 2;
second = first + (len - block);
track->Get((samplePtr)buffer1, floatSample, first, block);
track->Get((samplePtr)buffer2, floatSample, second, block);
for (int i = 0; i < block; i++) {
tmp = buffer1[i];
buffer1[i] = buffer2[block-i-1];
buffer2[block-i-1] = tmp;
}
track->Set((samplePtr)buffer1, floatSample, first, block);
track->Set((samplePtr)buffer2, floatSample, second, block);
len -= 2 * block;
first += block;
if( TrackProgress(count, 2*(first-originalStart) / (double) originalLen) ) {
rc = false;
break;
}
}
delete[] buffer1;
delete[] buffer2;
return rc;
}
bool EffectReverse::ProcessOneClip(int count, WaveTrack *track,
sampleCount start, sampleCount len,
sampleCount originalStart, sampleCount originalEnd)
{
bool rc = true;
// keep track of two blocks whose data we will swap
auto first = start;
auto blockSize = track->GetMaxBlockSize();
float tmp;
Floats buffer1{ blockSize };
Floats buffer2{ blockSize };
auto originalLen = originalEnd - originalStart;
while (len > 1) {
auto block =
limitSampleBufferSize( track->GetBestBlockSize(first), len / 2 );
auto second = first + (len - block);
track->Get((samplePtr)buffer1.get(), floatSample, first, block);
track->Get((samplePtr)buffer2.get(), floatSample, second, block);
for (decltype(block) i = 0; i < block; i++) {
tmp = buffer1[i];
buffer1[i] = buffer2[block-i-1];
buffer2[block-i-1] = tmp;
}
track->Set((samplePtr)buffer1.get(), floatSample, first, block);
track->Set((samplePtr)buffer2.get(), floatSample, second, block);
len -= 2 * block;
first += block;
if( TrackProgress(count, 2 * ( first - originalStart ).as_double() /
originalLen.as_double() ) ) {
rc = false;
break;
}
}
return rc;
}
bool EffectSineSweepGenerator::GenerateTrack(WaveTrack *tmp, const WaveTrack &track, int ntrack)
{
#ifdef __AUDEBUG__
printf("elsg: GENERATETRACK\n"); fflush(stdout); // ************DEBUG
#endif
bool bGoodResult = true;
//numSamples = track.TimeToLongSamples(mDuration);
sampleCount numSamples;
//Filter has different length
if(ntrack == m_pSsg->GetFilterChannel())
numSamples = sampleCount(m_pSsg->GetFilterLength());
else
numSamples = sampleCount(m_pSsg->GetBuffersLength());
sampleCount i = 0;
AFSample *data = new AFSample[tmp->GetMaxBlockSize()];
sampleCount block = 0;
while ((i < numSamples) && bGoodResult)
{
block = tmp->GetBestBlockSize(i);
if (block > (numSamples - i))
block = numSamples - i;
// GenerateBlock(data, track, block);
m_pSsg->FillBlock(data, AFSampleCount(block), AFSampleCount(i), ntrack);
// Add the generated data to the temporary track
tmp->Append((samplePtr)data, floatSample, block);
i += block;
// Update the progress meter
if (TrackProgress(ntrack, (double)i / numSamples))
bGoodResult = false;
}
tmp->Flush();
delete[] data;
return bGoodResult;
}
void EffectStereoToMono::ProcessOne()
{
float curLeftFrame;
float curRightFrame;
float curMonoFrame;
sampleCount idealBlockLen = mLeftTrack->GetMaxBlockSize() * 2;
sampleCount index = 0;
sampleCount outTrackOffset = 0;
float *leftBuffer = new float[idealBlockLen];
float *rightBuffer = new float[idealBlockLen];
bool rc;
while (index < mLeftTrackLen) {
rc = mLeftTrack->Get((samplePtr)leftBuffer, floatSample, index, idealBlockLen);
rc = mRightTrack->Get((samplePtr)rightBuffer, floatSample, index, idealBlockLen);
sampleCount limit = idealBlockLen;
if ((index + idealBlockLen) > mLeftTrackLen) {
limit = mLeftTrackLen - index;
}
for (sampleCount i = 0; i < limit; ++i) {
index++;
curLeftFrame = leftBuffer[i];
curRightFrame = rightBuffer[i];
curMonoFrame = (curLeftFrame + curRightFrame) / 2.0;
leftBuffer[i] = curMonoFrame;
}
rc = mLeftTrack->Set((samplePtr)leftBuffer, floatSample, outTrackOffset, limit);
outTrackOffset += limit;
TrackProgress(0, ((double)index / (double)mLeftTrackLen));
}
mLeftTrack->SetLinked(false);
mRightTrack->SetLinked(false);
mLeftTrack->SetChannel(Track::MonoChannel);
mTracks->Remove(mRightTrack);
delete mRightTrack;
delete [] leftBuffer;
delete [] rightBuffer;
}
bool EffectFilter::ProcessOne(int count, WaveTrack * t,
sampleCount start, sampleCount len)
{
sampleCount s = start;
sampleCount idealBlockLen = t->GetMaxBlockSize() * 4;
if (idealBlockLen % windowSize != 0)
idealBlockLen += (windowSize - (idealBlockLen % windowSize));
sampleType *buffer = new sampleType[idealBlockLen];
sampleType *window1 = new sampleType[windowSize];
sampleType *window2 = new sampleType[windowSize];
sampleType *thisWindow = window1;
sampleType *lastWindow = window2;
sampleCount originalLen = len;
unsigned int i;
for(i=0; i<windowSize; i++)
lastWindow[i] = 0;
while(len) {
unsigned int block = idealBlockLen;
if (block > len)
block = len;
t->Get(buffer, s, block);
for(i=0; i<block; i+=windowSize/2) {
unsigned int wcopy = windowSize;
if (i + wcopy > block)
wcopy = block - i;
unsigned int j;
for(j=0; j<wcopy; j++)
thisWindow[j] = buffer[i+j];
for(j=wcopy; j<windowSize; j++)
thisWindow[j] = 0;
Filter(windowSize, thisWindow);
for(j=0; j<windowSize/2; j++)
buffer[i+j] = thisWindow[j] + lastWindow[windowSize/2 + j];
sampleType *tempP = thisWindow;
thisWindow = lastWindow;
lastWindow = tempP;
}
if (len > block && len > windowSize/2)
block -= windowSize/2;
t->Set(buffer, s, block);
len -= block;
s += block;
TrackProgress(count, (s-start)/(double)originalLen);
}
delete[] buffer;
delete[] window1;
delete[] window2;
return true;
}
bool VampEffect::Process()
{
if (!mPlugin) return false;
TrackListIterator iter(mWaveTracks);
int count = 0;
WaveTrack *left = (WaveTrack *)iter.First();
bool multiple = false;
int prevTrackChannels = 0;
TrackListIterator scooter(iter);
if (left->GetLinked()) scooter.Next();
if (scooter.Next()) {
// if there is another track beyond this one and any linked one,
// then we're processing more than one track. That means we
// should use the originating track name in each new label
// track's name, to make clear which is which
multiple = true;
}
while (left) {
sampleCount lstart, rstart;
sampleCount len;
GetSamples(left, &lstart, &len);
WaveTrack *right = NULL;
int channels = 1;
if (left->GetLinked()) {
right = (WaveTrack *)iter.Next();
channels = 2;
GetSamples(right, &rstart, &len);
}
size_t step = mPlugin->getPreferredStepSize();
size_t block = mPlugin->getPreferredBlockSize();
bool initialiseRequired = true;
if (block == 0) {
if (step != 0) block = step;
else block = 1024;
}
if (step == 0) {
step = block;
}
if (prevTrackChannels > 0) {
// Plugin has already been initialised, so if the number of
// channels remains the same, we only need to do a reset.
// Otherwise we need to re-construct the whole plugin,
// because a Vamp plugin can't be re-initialised.
if (prevTrackChannels == channels) {
mPlugin->reset();
initialiseRequired = false;
} else {
//!!! todo: retain parameters previously set
Init();
}
}
if (initialiseRequired) {
if (!mPlugin->initialise(channels, step, block)) {
wxMessageBox(_("Sorry, Vamp Plug-in failed to initialize."));
return false;
}
}
LabelTrack *ltrack = mFactory->NewLabelTrack();
if (!multiple) {
ltrack->SetName(GetEffectName());
} else {
ltrack->SetName(wxString::Format(wxT("%s: %s"),
left->GetName().c_str(),
GetEffectName().c_str()));
}
mTracks->Add(ltrack);
float **data = new float*[channels];
for (int c = 0; c < channels; ++c) data[c] = new float[block];
sampleCount originalLen = len;
sampleCount ls = lstart;
sampleCount rs = rstart;
while (len) {
int request = block;
if (request > len) request = len;
if (left) left->Get((samplePtr)data[0], floatSample, ls, request);
if (right) right->Get((samplePtr)data[1], floatSample, rs, request);
if (request < (int)block) {
for (int c = 0; c < channels; ++c) {
for (int i = request; i < (int)block; ++i) {
data[c][i] = 0.f;
}
}
}
Vamp::RealTime timestamp = Vamp::RealTime::frame2RealTime
(ls, (int)(mRate + 0.5));
Vamp::Plugin::FeatureSet features = mPlugin->process(data, timestamp);
AddFeatures(ltrack, features);
if (len > (int)step) len -= step;
else len = 0;
ls += step;
rs += step;
if (channels > 1) {
if (TrackGroupProgress(count, (ls - lstart) / double(originalLen)))
return false;
} else {
if (TrackProgress(count, (ls - lstart) / double(originalLen)))
return false;
}
}
Vamp::Plugin::FeatureSet features = mPlugin->getRemainingFeatures();
AddFeatures(ltrack, features);
prevTrackChannels = channels;
left = (WaveTrack *)iter.Next();
}
return true;
}
//ProcessOne() takes a track, transforms it to bunch of buffer-blocks,
//and executes ProcessSoundTouch on these blocks
bool EffectSoundTouch::ProcessOne(WaveTrack *track,
longSampleCount start, longSampleCount end)
{
WaveTrack *outputTrack;
longSampleCount s;
mSoundTouch->setSampleRate((unsigned int)(track->GetRate()+0.5));
outputTrack = mFactory->NewWaveTrack(track->GetSampleFormat());
//Get the length of the buffer (as double). len is
//used simple to calculate a progress meter, so it is easier
//to make it a double now than it is to do it later
double len = (double)(end - start);
//Initiate a processing buffer. This buffer will (most likely)
//be shorter than the length of the track being processed.
float *buffer = new float[track->GetMaxBlockSize()];
//Go through the track one buffer at a time. s counts which
//sample the current buffer starts at.
s = start;
while (s < end) {
//Get a block of samples (smaller than the size of the buffer)
sampleCount block = track->GetBestBlockSize(s);
//Adjust the block size if it is the final block in the track
if (s + block > end)
block = end - s;
//Get the samples from the track and put them in the buffer
track->Get((samplePtr) buffer, floatSample, s, block);
//Add samples to SoundTouch
mSoundTouch->putSamples(buffer, block);
//Get back samples from SoundTouch
unsigned int outputCount = mSoundTouch->numSamples();
if (outputCount > 0) {
float *buffer2 = new float[outputCount];
mSoundTouch->receiveSamples(buffer2, outputCount);
outputTrack->Append((samplePtr)buffer2, floatSample, outputCount);
delete[] buffer2;
}
//Increment s one blockfull of samples
s += block;
//Update the Progress meter
if (TrackProgress(mCurTrackNum, (s - start) / len))
return false;
}
// Tell SoundTouch to finish processing any remaining samples
mSoundTouch->flush();
unsigned int outputCount = mSoundTouch->numSamples();
if (outputCount > 0) {
float *buffer2 = new float[outputCount];
mSoundTouch->receiveSamples(buffer2, outputCount);
outputTrack->Append((samplePtr)buffer2, floatSample, outputCount);
delete[] buffer2;
}
// Flush the output WaveTrack (since it's buffered, too)
outputTrack->Flush();
// Clean up the buffer
delete[]buffer;
// Take the output track and insert it in place of the original
// sample data
track->Clear(mT0, mT1);
track->Paste(mT0, outputTrack);
double newLength = outputTrack->GetEndTime();
if (newLength > m_maxNewLength)
m_maxNewLength = newLength;
// Delete the outputTrack now that its data is inserted in place
delete outputTrack;
//Return true because the effect processing succeeded.
return true;
}
bool EffectBassBoost::ProcessOne(int count, WaveTrack * t,
sampleCount start, sampleCount len)
{
float samplerate = (float) (t->GetRate());
/* Compute coefficents of the biquand IIR filter */
float omega = 2 * 3.141592653589 * frequency / samplerate;
float sn = sin(omega);
float cs = cos(omega);
float a = exp(log(10) * dB_boost / 40);
float shape = 1.0; /*Low Shelf filter's shape, if this is too large
or too small it will result an unstable filter */
float beta = sqrt((a * a + 1) / shape - (pow((a - 1), 2)));
/* Coefficients */
float b0 = a * ((a + 1) - (a - 1) * cs + beta * sn);
float b1 = 2 * a * ((a - 1) - (a + 1) * cs);
float b2 = a * ((a + 1) - (a - 1) * cs - beta * sn);
float a0 = ((a + 1) + (a - 1) * cs + beta * sn);
float a1 = -2 * ((a - 1) + (a + 1) * cs);
float a2 = (a + 1) + (a - 1) * cs - beta * sn;
/* initialise the filter */
float xn1 = 0, xn2 = 0, yn1 = 0, yn2 = 0;
float out, in = 0;
sampleCount s = start;
sampleCount originalLen = len;
sampleCount blockSize = t->GetMaxBlockSize();
float *buffer = new float[blockSize];
while (len) {
sampleCount block = t->GetBestBlockSize(s);
if (block > len)
block = len;
t->Get(buffer, s, block);
for (int i = 0; i < block; i++) {
in = buffer[i];
out = (b0 * in + b1 * xn1 + b2 * xn2 - a1 * yn1 - a2 * yn2) / a0;
xn2 = xn1;
xn1 = in;
yn2 = yn1;
yn1 = out;
if (out < -1.0)
out = -1.0;
else if (out > 1.0)
out = 1.0; //Prevents clipping
buffer[i] = out;
}
t->Set(buffer, s, block);
len -= block;
s += block;
TrackProgress(count, (s-start)/(double)originalLen);
}
delete[]buffer;
return true;
}
bool EffectPaulstretch::ProcessOne(WaveTrack *track,double t0,double t1,int count)
{
auto badAllocMessage = _("Requested value exceeds memory capacity.");
const auto stretch_buf_size = GetBufferSize(track->GetRate());
if (stretch_buf_size == 0) {
::Effect::MessageBox( badAllocMessage );
return false;
}
double amount = this->mAmount;
auto start = track->TimeToLongSamples(t0);
auto end = track->TimeToLongSamples(t1);
auto len = end - start;
const auto minDuration = stretch_buf_size * 2 + 1;
if (minDuration < stretch_buf_size) {
// overflow!
::Effect::MessageBox( badAllocMessage );
return false;
}
if (len < minDuration) { //error because the selection is too short
float maxTimeRes = log( len.as_double() ) / log(2.0);
maxTimeRes = pow(2.0, floor(maxTimeRes) + 0.5);
maxTimeRes = maxTimeRes / track->GetRate();
if (this->IsPreviewing()) {
double defaultPreviewLen;
gPrefs->Read(wxT("/AudioIO/EffectsPreviewLen"), &defaultPreviewLen, 6.0);
/* i18n-hint: 'Time Resolution' is the name of a control in the Paulstretch effect.*/
if ((minDuration / mProjectRate) < defaultPreviewLen) {
::Effect::MessageBox (wxString::Format(_("Audio selection too short to preview.\n\n"
"Try increasing the audio selection to at least %.1f seconds,\n"
"or reducing the 'Time Resolution' to less than %.1f seconds."),
(minDuration / track->GetRate()) + 0.05, // round up to 1/10 s.
floor(maxTimeRes * 10.0) / 10.0),
wxOK | wxICON_EXCLAMATION);
}
else {
/* i18n-hint: 'Time Resolution' is the name of a control in the Paulstretch effect.*/
::Effect::MessageBox (wxString::Format(_("Unable to Preview.\n\n"
"For the current audio selection, the maximum\n"
"'Time Resolution' is %.1f seconds."),
floor(maxTimeRes * 10.0) / 10.0),
wxOK | wxICON_EXCLAMATION);
}
}
else {
/* i18n-hint: 'Time Resolution' is the name of a control in the Paulstretch effect.*/
::Effect::MessageBox (wxString::Format(_("The 'Time Resolution' is too long for the selection.\n\n"
"Try increasing the audio selection to at least %.1f seconds,\n"
"or reducing the 'Time Resolution' to less than %.1f seconds."),
(minDuration / track->GetRate()) + 0.05, // round up to 1/10 s.
floor(maxTimeRes * 10.0) / 10.0),
wxOK | wxICON_EXCLAMATION);
}
return false;
}
auto dlen = len.as_double();
double adjust_amount = dlen /
(dlen - ((double)stretch_buf_size * 2.0));
amount = 1.0 + (amount - 1.0) * adjust_amount;
auto outputTrack = mFactory->NewWaveTrack(track->GetSampleFormat(),track->GetRate());
try {
// This encloses all the allocations of buffers, including those in
// the constructor of the PaulStretch object
PaulStretch stretch(amount, stretch_buf_size, track->GetRate());
auto nget = stretch.get_nsamples_for_fill();
auto bufsize = stretch.poolsize;
Floats buffer0{ bufsize };
float *bufferptr0 = buffer0.get();
bool first_time = true;
const auto fade_len = std::min<size_t>(100, bufsize / 2 - 1);
bool cancelled = false;
{
Floats fade_track_smps{ fade_len };
decltype(len) s=0;
while (s < len) {
track->Get((samplePtr)bufferptr0, floatSample, start + s, nget);
stretch.process(buffer0.get(), nget);
if (first_time) {
stretch.process(buffer0.get(), 0);
};
s += nget;
if (first_time){//blend the the start of the selection
track->Get((samplePtr)fade_track_smps.get(), floatSample, start, fade_len);
first_time = false;
for (size_t i = 0; i < fade_len; i++){
float fi = (float)i / (float)fade_len;
stretch.out_buf[i] =
stretch.out_buf[i] * fi + (1.0 - fi) * fade_track_smps[i];
}
}
if (s >= len){//blend the end of the selection
track->Get((samplePtr)fade_track_smps.get(), floatSample, end - fade_len, fade_len);
for (size_t i = 0; i < fade_len; i++){
float fi = (float)i / (float)fade_len;
auto i2 = bufsize / 2 - 1 - i;
stretch.out_buf[i2] =
stretch.out_buf[i2] * fi + (1.0 - fi) *
fade_track_smps[fade_len - 1 - i];
}
}
outputTrack->Append((samplePtr)stretch.out_buf.get(), floatSample, stretch.out_bufsize);
nget = stretch.get_nsamples();
if (TrackProgress(count,
s.as_double() / len.as_double()
)) {
cancelled = true;
break;
}
}
}
if (!cancelled){
outputTrack->Flush();
track->Clear(t0,t1);
track->Paste(t0, outputTrack.get());
m_t1 = mT0 + outputTrack->GetEndTime();
}
return !cancelled;
}
catch ( const std::bad_alloc& ) {
::Effect::MessageBox( badAllocMessage );
return false;
}
};
bool LadspaEffect::ProcessStereo(int count, WaveTrack *left, WaveTrack *right,
sampleCount lstart,
sampleCount rstart,
sampleCount len)
{
/* Allocate buffers */
if (mBlockSize == 0) {
mBlockSize = left->GetMaxBlockSize() * 2;
fInBuffer = new float *[inputs];
unsigned long i;
for (i = 0; i < inputs; i++)
fInBuffer[i] = new float[mBlockSize];
fOutBuffer = new float *[outputs];
for (i = 0; i < outputs; i++)
fOutBuffer[i] = new float[mBlockSize];
}
/* Instantiate the plugin */
unsigned long rate = (unsigned long)(left->GetRate() + 0.5);
LADSPA_Handle handle = mData->instantiate(mData, rate);
unsigned long p;
for(p=0; p<inputs; p++) {
mData->connect_port(handle, inputPorts[p], fInBuffer[p]);
}
for(p=0; p<outputs; p++) {
mData->connect_port(handle, outputPorts[p], fOutBuffer[p]);
}
for(p=0; p<mData->PortCount; p++) {
LADSPA_PortDescriptor d = mData->PortDescriptors[p];
if (LADSPA_IS_PORT_CONTROL(d)) {
if (LADSPA_IS_PORT_INPUT(d)) {
mData->connect_port(handle, p, &inputControls[p]);
}
else
mData->connect_port(handle, p, &outputControls[p]);
}
}
if (mData->activate)
mData->activate(handle);
// Actually perform the effect here
sampleCount originalLen = len;
sampleCount ls = lstart;
sampleCount rs = rstart;
while (len) {
int block = mBlockSize;
if (block > len)
block = len;
if (left && inputs > 0) {
left->Get((samplePtr)fInBuffer[0], floatSample, ls, block);
}
if (right && inputs > 1) {
right->Get((samplePtr)fInBuffer[1], floatSample, rs, block);
}
mData->run(handle, block);
if (left && outputs > 0) {
left->Set((samplePtr)fOutBuffer[0], floatSample, ls, block);
}
if (right && outputs > 1) {
right->Set((samplePtr)fOutBuffer[1], floatSample, rs, block);
}
len -= block;
ls += block;
rs += block;
if (inputs > 1) {
if (TrackGroupProgress(count, (ls-lstart)/(double)originalLen))
return false;
}
else {
if (TrackProgress(count, (ls-lstart)/(double)originalLen))
return false;
}
}
if (mData->deactivate)
mData->deactivate(handle);
if (mData->cleanup)
mData->cleanup(handle);
return true;
}
bool VSTEffect::ProcessStereo(int count,
WaveTrack *left, WaveTrack *right,
sampleCount lstart, sampleCount rstart,
sampleCount len)
{
bool rc = true;
// Initialize time info
mTimeInfo.samplePos = 0.0;
mTimeInfo.sampleRate = left->GetRate();
mTimeInfo.flags |= kVstTransportPlaying;
// Turn the power on
callDispatcher(effMainsChanged, 0, 1, NULL, 0.0);
// Tell effect we're starting to process
callDispatcher(effStartProcess, 0, 0, NULL, 0.0);
// Actually perform the effect here
sampleCount originalLen = len;
sampleCount ls = lstart;
sampleCount rs = rstart;
while (len) {
int block = mBlockSize;
if (block > len) {
block = len;
}
left->Get((samplePtr)mInBuffer[0], floatSample, ls, block);
if (right) {
right->Get((samplePtr)mInBuffer[1], floatSample, rs, block);
}
callProcessReplacing(mInBuffer, mOutBuffer, block);
left->Set((samplePtr)mOutBuffer[0], floatSample, ls, block);
if (right) {
right->Set((samplePtr)mOutBuffer[1], floatSample, rs, block);
}
len -= block;
ls += block;
rs += block;
mTimeInfo.samplePos += ((double) block / mTimeInfo.sampleRate);
if (mInputs > 1) {
if (TrackGroupProgress(count, (ls - lstart) / (double)originalLen)) {
rc = false;
break;
}
}
else {
if (TrackProgress(count, (ls - lstart) / (double)originalLen)) {
rc = false;
break;
}
}
}
// Tell effect we're done
callDispatcher(effStopProcess, 0, 0, NULL, 0.0);
// Turn the power off
callDispatcher(effMainsChanged, 0, 0, NULL, 0.0);
// No longer playing
mTimeInfo.samplePos = 0.0;
mTimeInfo.sampleRate = 44100.0;
mTimeInfo.tempo = 120.0;
mTimeInfo.timeSigNumerator = 4;
mTimeInfo.timeSigDenominator = 4;
mTimeInfo.flags = kVstTempoValid | kVstNanosValid;
return rc;
}
bool EffectNoiseRemoval::ProcessOne(int count, WaveTrack * track,
longSampleCount start, sampleCount len)
{
bool retCode = true;
sampleCount s = 0;
sampleCount idealBlockLen = track->GetMaxBlockSize() * 4;
if (idealBlockLen % windowSize != 0)
idealBlockLen += (windowSize - (idealBlockLen % windowSize));
float *buffer = new float[idealBlockLen];
float *window1 = new float[windowSize];
float *window2 = new float[windowSize];
float *thisWindow = window1;
float *lastWindow = window2;
int i;
for(i=0; i<windowSize; i++) {
lastWindow[i] = 0;
smoothing[i] = float(0.0);
}
while((s < len)&&((len-s)>(windowSize/2))) {
sampleCount block = idealBlockLen;
if (s + block > len)
block = len - s;
track->Get((samplePtr) buffer, floatSample, start + s, block);
for(i=0; i<(block-windowSize/2); i+=windowSize/2) {
int wcopy = windowSize;
if (i + wcopy > block)
wcopy = block - i;
int j;
for(j=0; j<wcopy; j++)
thisWindow[j] = buffer[i+j];
for(j=wcopy; j<windowSize; j++)
thisWindow[j] = 0;
if (mDoProfile)
GetProfile(windowSize, thisWindow);
else {
//TIDY-ME: could we just test mLevel=<0 in CheckWhetherSkipEffect?
if (mLevel > 0) { // Skip NoiseRemoval if zero ... may apply for CleanChain
RemoveNoise(windowSize, thisWindow);
for(j=0; j<windowSize/2; j++) {
buffer[i+j] = thisWindow[j] + lastWindow[windowSize/2 + j];
}
}
}
float *tempP = thisWindow;
thisWindow = lastWindow;
lastWindow = tempP;
}
// Shift by half-a-window less than the block size we loaded
// (so that the blocks properly overlap)
block -= windowSize/2;
if (!mDoProfile)
track->Set((samplePtr) buffer, floatSample, start + s, block);
s += block;
if (TrackProgress(count, s / (double) len)) {
retCode = false;
break;
}
}
delete[] buffer;
delete[] window1;
delete[] window2;
return retCode;
}
// ProcessOne() takes a track, transforms it to bunch of buffer-blocks,
// and calls libsamplerate code on these blocks.
bool EffectChangeSpeed::ProcessOne(WaveTrack * track,
sampleCount start, sampleCount end)
{
if (track == NULL)
return false;
// initialization, per examples of Mixer::Mixer and
// EffectSoundTouch::ProcessOne
WaveTrack * outputTrack = mFactory->NewWaveTrack(track->GetSampleFormat(),
track->GetRate());
//Get the length of the selection (as double). len is
//used simple to calculate a progress meter, so it is easier
//to make it a double now than it is to do it later
double len = (double)(end - start);
// Initiate processing buffers, most likely shorter than
// the length of the selection being processed.
sampleCount inBufferSize = track->GetMaxBlockSize();
float * inBuffer = new float[inBufferSize];
sampleCount outBufferSize =
(sampleCount)((mFactor * inBufferSize) + 10);
float * outBuffer = new float[outBufferSize];
// Set up the resampling stuff for this track.
Resample resample(true, mFactor, mFactor);
//Go through the track one buffer at a time. samplePos counts which
//sample the current buffer starts at.
bool bResult = true;
sampleCount blockSize;
sampleCount samplePos = start;
while (samplePos < end) {
//Get a blockSize of samples (smaller than the size of the buffer)
blockSize = track->GetBestBlockSize(samplePos);
//Adjust the block size if it is the final block in the track
if (samplePos + blockSize > end)
blockSize = end - samplePos;
//Get the samples from the track and put them in the buffer
track->Get((samplePtr) inBuffer, floatSample, samplePos, blockSize);
int inUsed;
int outgen = resample.Process(mFactor,
inBuffer,
blockSize,
((samplePos + blockSize) >= end),
&inUsed,
outBuffer,
outBufferSize);
if (outgen < 0) {
bResult = false;
break;
}
if (outgen > 0)
outputTrack->Append((samplePtr)outBuffer, floatSample,
outgen);
// Increment samplePos
samplePos += inUsed;
// Update the Progress meter
if (TrackProgress(mCurTrackNum, (samplePos - start) / len)) {
bResult = false;
break;
}
}
// Flush the output WaveTrack (since it's buffered, too)
outputTrack->Flush();
// Clean up the buffers
delete [] inBuffer;
delete [] outBuffer;
// Take the output track and insert it in place of the original
// sample data
double newLength = outputTrack->GetEndTime();
if (bResult)
{
SetTimeWarper(new LinearTimeWarper(mCurT0, mCurT0, mCurT1, mCurT0 + newLength));
bResult = track->ClearAndPaste(mCurT0, mCurT1, outputTrack, true, false, GetTimeWarper());
}
if (newLength > mMaxNewLength)
mMaxNewLength = newLength;
// Delete the outputTrack now that its data is inserted in place
delete outputTrack;
return bResult;
}
bool LV2Effect::ProcessStereo(int count,
WaveTrack *left,
WaveTrack *right,
sampleCount lstart,
sampleCount rstart,
sampleCount len)
{
/* Allocate buffers */
if (mBlockSize == 0)
{
mBlockSize = left->GetMaxBlockSize() * 2;
fInBuffer = new float *[mAudioInputs.GetCount()];
for (size_t i = 0; i < mAudioInputs.GetCount(); i++)
{
fInBuffer[i] = new float[mBlockSize];
}
fOutBuffer = new float *[mAudioOutputs.GetCount()];
for (size_t i = 0; i < mAudioOutputs.GetCount(); i++)
{
fOutBuffer[i] = new float[mBlockSize];
}
}
/* Instantiate the plugin */
LilvInstance *handle = lilv_plugin_instantiate(mData,
left->GetRate(),
gLV2Features);
if (!handle)
{
wxMessageBox(wxString::Format(_("Unable to load plug-in %s"), pluginName.c_str()));
return false;
}
/* Write the Note On to the MIDI event buffer and connect it */
LV2_Event_Buffer *midiBuffer = NULL;
int noteOffTime;
if (mMidiInput)
{
midiBuffer = lv2_event_buffer_new(40, 2);
LV2_Event_Iterator iter;
lv2_event_begin(&iter, midiBuffer);
uint8_t noteOn[] = { 0x90, mNoteKey, mNoteVelocity };
lv2_event_write(&iter, 0, 0, 1, 3, noteOn);
noteOffTime = mNoteLength * left->GetRate();
if (noteOffTime < len && noteOffTime < mBlockSize) {
uint8_t noteOff[] = { 0x80, mNoteKey, 64 };
lv2_event_write(&iter, noteOffTime, 0, 1, 3, noteOff);
}
lilv_instance_connect_port(handle, mMidiInput->mIndex, midiBuffer);
}
for (size_t p = 0; p < mAudioInputs.GetCount(); p++)
{
lilv_instance_connect_port(handle, mAudioInputs[p].mIndex, fInBuffer[p]);
}
for (size_t p = 0; p < mAudioOutputs.GetCount(); p++)
{
lilv_instance_connect_port(handle, mAudioOutputs[p].mIndex, fOutBuffer[p]);
}
for (size_t p = 0; p < mControlInputs.GetCount(); p++)
{
lilv_instance_connect_port(handle, mControlInputs[p].mIndex,
&mControlInputs[p].mControlBuffer);
}
for (size_t p = 0; p < mControlOutputs.GetCount(); p++)
{
lilv_instance_connect_port(handle, mControlOutputs[p].mIndex,
&mControlOutputs[p].mControlBuffer);
}
float latency = 0.0;
if (mLatencyPortIndex >= 0)
{
lilv_instance_connect_port(handle, mLatencyPortIndex, &latency);
}
lilv_instance_activate(handle);
// Actually perform the effect here
sampleCount originalLen = len;
sampleCount ls = lstart;
sampleCount rs = rstart;
sampleCount ols = ls;
sampleCount ors = rs;
bool noteOver = false;
sampleCount delayed = 0;
sampleCount delay = 0;
bool cleared = false;
while (len || delayed)
{
int block = mBlockSize;
if (len)
{
if (block > len)
{
block = len;
}
if (left && mAudioInputs.GetCount() > 0)
{
left->Get((samplePtr)fInBuffer[0], floatSample, ls, block);
}
if (right && mAudioInputs.GetCount() > 1)
{
right->Get((samplePtr)fInBuffer[1], floatSample, rs, block);
}
}
else if (delayed)
{
// At the end if we don't have enough left for a whole block
if (block > delayed)
{
block = delayed;
}
// Clear the input buffer so that we only pass zeros to the effect.
if (!cleared)
{
for (int i = 0; i < mBlockSize; i++)
{
fInBuffer[0][i] = 0.0;
}
if (right)
{
memcpy(fInBuffer[1], fOutBuffer[0], mBlockSize);
}
cleared = true;
}
}
lilv_instance_run(handle, block);
if (delayed == 0 && latency != 0)
{
delayed = delay = latency;
}
if (delay >= block)
{
delay -= block;
}
else if (delay > 0)
{
sampleCount oblock = block - delay;
if (left && mAudioOutputs.GetCount() > 0)
{
left->Set((samplePtr)(fOutBuffer[0] + delay), floatSample, ols, oblock);
}
if (right && mAudioOutputs.GetCount() > 1)
{
right->Set((samplePtr)(fOutBuffer[1] + delay), floatSample, ors, oblock);
}
ols += oblock;
ors += oblock;
delay = 0;
}
else
{
if (left && mAudioOutputs.GetCount() > 0)
{
left->Set((samplePtr)fOutBuffer[0], floatSample, ols, block);
}
if (right && mAudioOutputs.GetCount() > 1)
{
right->Set((samplePtr)fOutBuffer[1], floatSample, ors, block);
}
ols += block;
ors += block;
}
if (len)
{
len -= block;
noteOffTime -= block;
// Clear the event buffer and add the note off event if needed
if (mMidiInput)
{
lv2_event_buffer_reset(midiBuffer, 1,
(uint8_t *)midiBuffer +
sizeof(LV2_Event_Buffer));
if (!noteOver && noteOffTime < len && noteOffTime < block)
{
LV2_Event_Iterator iter;
lv2_event_begin(&iter, midiBuffer);
uint8_t noteOff[] = { 0x80, mNoteKey, 64 };
lv2_event_write(&iter, noteOffTime, 0, 1, 3, noteOff);
noteOver = true;
}
}
}
else if (delayed)
{
delayed -= block;
}
ls += block;
rs += block;
if (mAudioInputs.GetCount() > 1)
{
if (TrackGroupProgress(count, (ls-lstart)/(double)originalLen))
{
return false;
}
}
else
{
if (TrackProgress(count, (ls-lstart)/(double)originalLen))
{
return false;
}
}
}
lilv_instance_deactivate(handle);
lilv_instance_free(handle);
return true;
}
//AnalyseDC() takes a track, transforms it to bunch of buffer-blocks,
//and executes AnalyzeData on it...
// sets mOffset
bool EffectNormalize::AnalyseDC(WaveTrack * track, wxString msg)
{
bool rc = true;
sampleCount s;
mOffset = 0.0; // we might just return
if(!mDC) // don't do analysis if not doing dc removal
return(rc);
//Transform the marker timepoints to samples
sampleCount start = track->TimeToLongSamples(mCurT0);
sampleCount end = track->TimeToLongSamples(mCurT1);
//Get the length of the buffer (as double). len is
//used simply to calculate a progress meter, so it is easier
//to make it a double now than it is to do it later
double len = (double)(end - start);
//Initiate a processing buffer. This buffer will (most likely)
//be shorter than the length of the track being processed.
float *buffer = new float[track->GetMaxBlockSize()];
mSum = 0.0; // dc offset inits
mCount = 0;
//Go through the track one buffer at a time. s counts which
//sample the current buffer starts at.
s = start;
while (s < end) {
//Get a block of samples (smaller than the size of the buffer)
sampleCount block = track->GetBestBlockSize(s);
//Adjust the block size if it is the final block in the track
if (s + block > end)
block = end - s;
//Get the samples from the track and put them in the buffer
track->Get((samplePtr) buffer, floatSample, s, block);
//Process the buffer.
AnalyzeData(buffer, block);
//Increment s one blockfull of samples
s += block;
//Update the Progress meter
if (TrackProgress(mCurTrackNum,
((double)(s - start) / len)/2.0, msg)) {
rc = false; //lda .. break, not return, so that buffer is deleted
break;
}
}
//Clean up the buffer
delete[] buffer;
mOffset = (float)(-mSum / mCount); // calculate actual offset (amount that needs to be added on)
//Return true because the effect processing succeeded ... unless cancelled
return rc;
}
bool EffectNoise::Process()
{
if (noiseDuration <= 0.0)
noiseDuration = sDefaultGenerateLen;
//Iterate over each track
int ntrack = 0;
this->CopyInputWaveTracks(); // Set up mOutputWaveTracks.
bool bGoodResult = true;
TrackListIterator iter(mOutputWaveTracks);
WaveTrack *track = (WaveTrack *)iter.First();
while (track) {
WaveTrack *tmp = mFactory->NewWaveTrack(track->GetSampleFormat(), track->GetRate());
numSamples = (longSampleCount)(noiseDuration * track->GetRate() + 0.5);
longSampleCount i = 0;
float *data = new float[tmp->GetMaxBlockSize()];
sampleCount block;
while ((i < numSamples) && bGoodResult) {
block = tmp->GetBestBlockSize(i);
if (block > (numSamples - i))
block = numSamples - i;
MakeNoise(data, block, track->GetRate(), noiseAmplitude);
tmp->Append((samplePtr)data, floatSample, block);
i += block;
//Update the Progress meter
if (TrackProgress(ntrack, (double)i / numSamples))
bGoodResult = false;
}
delete[] data;
tmp->Flush();
track->Clear(mT0, mT1);
track->Paste(mT0, tmp);
delete tmp;
if (!bGoodResult)
break;
//Iterate to the next track
ntrack++;
track = (WaveTrack *)iter.Next();
}
if (bGoodResult)
{
/*
save last used values
save duration unless value was got from selection, so we save only
when user explicitely setup a value
*/
if (mT1 == mT0)
gPrefs->Write(wxT("/CsPresets/NoiseGen_Duration"), noiseDuration);
gPrefs->Write(wxT("/CsPresets/NoiseGen_Type"), noiseType);
gPrefs->Write(wxT("/CsPresets/NoiseGen_Amp"), noiseAmplitude);
mT1 = mT0 + noiseDuration; // Update selection.
}
this->ReplaceProcessedWaveTracks(bGoodResult);
return bGoodResult;
}
bool EffectDtmf::GenerateTrack(WaveTrack *tmp,
const WaveTrack &track,
int ntrack)
{
bool bGoodResult = true;
// all dtmf sequence durations in samples from seconds
numSamplesSequence = (sampleCount)(mDuration * track.GetRate() + 0.5);
numSamplesTone = (sampleCount)(dtmfTone * track.GetRate() + 0.5);
numSamplesSilence = (sampleCount)(dtmfSilence * track.GetRate() + 0.5);
// recalculate the sum, and spread the difference - due to approximations.
// Since diff should be in the order of "some" samples, a division (resulting in zero)
// is not sufficient, so we add the additional remaining samples in each tone/silence block,
// at least until available.
int diff = numSamplesSequence - (dtmfNTones*numSamplesTone) - (dtmfNTones-1)*numSamplesSilence;
if (diff>dtmfNTones) {
// in this case, both these values would change, so it makes sense to recalculate diff
// otherwise just keep the value we already have
// should always be the case that dtmfNTones>1, as if 0, we don't even start processing,
// and with 1 there is no difference to spread (no silence slot)...
wxASSERT(dtmfNTones > 1);
numSamplesTone += (diff/(dtmfNTones));
numSamplesSilence += (diff/(dtmfNTones-1));
diff = numSamplesSequence - (dtmfNTones*numSamplesTone) - (dtmfNTones-1)*numSamplesSilence;
}
// this var will be used as extra samples distributor
int extra=0;
sampleCount i = 0;
sampleCount j = 0;
int n=0; // pointer to string in dtmfString
sampleCount block;
bool isTone = true;
float *data = new float[tmp->GetMaxBlockSize()];
// for the whole dtmf sequence, we will be generating either tone or silence
// according to a bool value, and this might be done in small chunks of size
// 'block', as a single tone might sometimes be larger than the block
// tone and silence generally have different duration, thus two generation blocks
//
// Note: to overcome a 'clicking' noise introduced by the abrupt transition from/to
// silence, I added a fade in/out of 1/250th of a second (4ms). This can still be
// tweaked but gives excellent results at 44.1kHz: I haven't tried other freqs.
// A problem might be if the tone duration is very short (<10ms)... (?)
//
// One more problem is to deal with the approximations done when calculating the duration
// of both tone and silence: in some cases the final sum might not be same as the initial
// duration. So, to overcome this, we had a redistribution block up, and now we will spread
// the remaining samples in every bin in order to achieve the full duration: test case was
// to generate an 11 tone DTMF sequence, in 4 seconds, and with DutyCycle=75%: after generation
// you ended up with 3.999s or in other units: 3 seconds and 44097 samples.
//
while ((i < numSamplesSequence) && bGoodResult) {
if (isTone)
// generate tone
{
// the statement takes care of extracting one sample from the diff bin and
// adding it into the tone block until depletion
extra=(diff-- > 0?1:0);
for(j=0; j < numSamplesTone+extra && bGoodResult; j+=block) {
block = tmp->GetBestBlockSize(j);
if (block > (numSamplesTone+extra - j))
block = numSamplesTone+extra - j;
// generate the tone and append
MakeDtmfTone(data, block, track.GetRate(), dtmfString[n], j, numSamplesTone, dtmfAmplitude);
tmp->Append((samplePtr)data, floatSample, block);
//Update the Progress meter
if (TrackProgress(ntrack, (double)(i+j) / numSamplesSequence))
bGoodResult = false;
}
i += numSamplesTone;
n++;
if(n>=dtmfNTones)break;
}
else
// generate silence
{
// the statement takes care of extracting one sample from the diff bin and
// adding it into the silence block until depletion
extra=(diff-- > 0?1:0);
for(j=0; j < numSamplesSilence+extra && bGoodResult; j+=block) {
block = tmp->GetBestBlockSize(j);
if (block > (numSamplesSilence+extra - j))
block = numSamplesSilence+extra - j;
// generate silence and append
memset(data, 0, sizeof(float)*block);
tmp->Append((samplePtr)data, floatSample, block);
//Update the Progress meter
if (TrackProgress(ntrack, (double)(i+j) / numSamplesSequence))
bGoodResult = false;
}
i += numSamplesSilence;
}
// flip flag
isTone=!isTone;
} // finished the whole dtmf sequence
delete[] data;
return bGoodResult;
}
bool EffectTruncSilence::Process()
{
SelectedTrackListOfKindIterator iter(Track::Wave, mTracks);
WaveTrack *t;
double t0 = mT0;
double t1 = mT1;
int tndx;
int tcount = 0;
int fr;
// Init using first track
t = (WaveTrack *) iter.First();
double rate = t->GetRate();
sampleCount blockLen = t->GetMaxBlockSize();
// Get the left and right bounds for all tracks
while (t) {
// Make sure all tracks have the same sample rate
if (rate != t->GetRate()) {
wxMessageBox(_("All tracks must have the same sample rate"), _("Truncate Silence"));
return false;
}
// Count the tracks
tcount++;
// Set the current bounds to whichever left marker is
// greater and whichever right marker is less
t0 = wxMax(mT0, t->GetStartTime());
t1 = wxMin(mT1, t->GetEndTime());
// Use the smallest block size of all the tracks
blockLen = wxMin(blockLen, t->GetMaxBlockSize());
// Iterate to the next track
t = (WaveTrack*) iter.Next();
}
// Just a sanity check, really it should be much higher
if(blockLen < 4*mBlendFrameCount)
blockLen = 4*mBlendFrameCount;
// Transform the marker timepoints to samples
t = (WaveTrack *) iter.First();
sampleCount start = t->TimeToLongSamples(t0);
sampleCount end = t->TimeToLongSamples(t1);
// Bigger buffers reduce 'reset'
//blockLen *= 8;
// Stress-test the logic for cutting samples through block endpoints
//blockLen /= 8;
// Set thresholds
// We have a lower bound on the amount of silence we chop out at a time
// to avoid chopping up low frequency sounds. We're good down to 10Hz
// if we use 100ms.
const float minTruncMs = 1.0f;
double truncDbSilenceThreshold = Enums::Db2Signal[mTruncDbChoiceIndex];
int truncInitialAllowedSilentSamples =
int((wxMax( mTruncInitialAllowedSilentMs, minTruncMs) * rate) / 1000.0);
int truncLongestAllowedSilentSamples =
int((wxMax( mTruncLongestAllowedSilentMs, minTruncMs) * rate) / 1000.0);
// Require at least 4 samples for lengths
if(truncInitialAllowedSilentSamples < 4)
truncInitialAllowedSilentSamples = 4;
if(truncLongestAllowedSilentSamples < 4)
truncLongestAllowedSilentSamples = 4;
// If the cross-fade is longer than the minimum length,
// then limit the cross-fade length to the minimum length
// This allows us to have reasonable cross-fade by default
// and still allow for 1ms minimum lengths
if(truncInitialAllowedSilentSamples < mBlendFrameCount)
mBlendFrameCount = truncInitialAllowedSilentSamples;
if(truncLongestAllowedSilentSamples < mBlendFrameCount)
mBlendFrameCount = truncLongestAllowedSilentSamples;
// For sake of efficiency, don't let blockLen be less than double the longest silent samples
// up until a sane limit of 1Meg samples
while((blockLen > 0) && (blockLen < truncLongestAllowedSilentSamples*2) && (blockLen < 1048576)) {
blockLen *= 2;
}
// Don't allow either value to be more than half of the block length
if(truncLongestAllowedSilentSamples > blockLen/2)
truncLongestAllowedSilentSamples = blockLen/2;
if(truncInitialAllowedSilentSamples > truncLongestAllowedSilentSamples)
truncInitialAllowedSilentSamples = truncLongestAllowedSilentSamples;
// We use the 'longest' variable as additive to the 'initial' variable
truncLongestAllowedSilentSamples -= truncInitialAllowedSilentSamples;
// Perform the crossfade half-way through the minimum removed silence duration
int rampInFrames = (truncInitialAllowedSilentSamples + mBlendFrameCount) / 2;
if(rampInFrames > truncInitialAllowedSilentSamples)
rampInFrames = truncInitialAllowedSilentSamples;
// Allocate buffers
float **buffer = new float*[tcount];
for (tndx = 0; tndx < tcount; tndx++) {
buffer[tndx] = new float[blockLen];
}
// Start processing
//Track::All is needed because this effect has clear functionality
this->CopyInputTracks(Track::All); // Set up mOutputTracks.
SelectedTrackListOfKindIterator iterOut(Track::Wave, mOutputTracks);
sampleCount index = start;
sampleCount outTrackOffset = start;
bool cancelled = false;
// Reset
bool ignoringFrames = false;
bool truncToMinimum = true; // Ignore the initial samples until we get above the noise floor
sampleCount consecutiveSilentFrames = 0;
sampleCount truncIndex = 0;
sampleCount i = 0;
sampleCount keep;
while (index < end) {
// Limit size of current block if we've reached the end
sampleCount count = blockLen-i;
if ((index + count) > end) {
count = end - index;
}
// Fill the buffers
tndx = 0;
t = (WaveTrack *) iter.First();
while (t) {
t->Get((samplePtr)(buffer[tndx++]+i), floatSample, index, count);
t = (WaveTrack *) iter.Next();
}
// Shift over to account for samples remaining from prior block
sampleCount limit = count+i;
// Look for silences in current block
for ( ; i < limit; i++) {
// Is current frame in all tracks below threshold
bool below = true;
for (tndx = 0; tndx < tcount; tndx++) {
if (fabs(buffer[tndx][i]) >= truncDbSilenceThreshold) {
below = false;
break;
}
}
// Make sure we cross-fade and output the last silence
// so we get a smooth transition into whatever follows the selected region
// Also set the 'truncToMinimum' flag so that the last silence is truncated to the minimum amount
if(below && ((index+i+1) == end)) {
below = false;
truncToMinimum = true;
}
// Count frame if it's below threshold
if (below) {
consecutiveSilentFrames++;
// Ignore this frame (equivalent to cutting it)
// otherwise, keep sample to be part of allowed silence
if (consecutiveSilentFrames > truncInitialAllowedSilentSamples) {
ignoringFrames = true;
continue;
}
}
else {
if (ignoringFrames == true) {
// Scale the consectiveSilentFrames so we keep a silence duration
// which is proportional to the original silence up to the limit
keep = consecutiveSilentFrames - truncInitialAllowedSilentSamples;
keep /= mSilenceCompressRatio;
// The first and last samples always get truncated to the minimum amount
if(truncToMinimum == true)
keep = 0;
if(keep > truncLongestAllowedSilentSamples)
keep = truncLongestAllowedSilentSamples;
if(keep < 0)
keep = 0;
// Compute the location of the cross-fade to be halfway through the silence
// with restriction to the samples we still have available to use
rampInFrames = (truncInitialAllowedSilentSamples - keep + mBlendFrameCount) / 2;
if(rampInFrames > truncInitialAllowedSilentSamples)
rampInFrames = truncInitialAllowedSilentSamples;
if(rampInFrames < mBlendFrameCount)
rampInFrames = mBlendFrameCount;
// Include the cross-fade samples in the count to make the loop logic easier
keep += rampInFrames;
truncIndex -= rampInFrames;
// back up for cross-fade
sampleCount curOffset = i - keep;
if(curOffset < 0) {
// This should never happen, but just in case...
keep += curOffset - rampInFrames;
if(keep < mBlendFrameCount)
keep = mBlendFrameCount;
curOffset = 0;
}
if(truncIndex < 0) {
// This should never happen, but just in case...
truncIndex = 0;
}
for (tndx = 0; tndx < tcount; tndx++) {
// Cross fade the cut point
for (fr = 0; fr < mBlendFrameCount; fr++) {
buffer[tndx][truncIndex+fr] = ((mBlendFrameCount-fr)*buffer[tndx][truncIndex+fr] + fr*buffer[tndx][curOffset + fr]) / mBlendFrameCount;
}
// Append the 'keep' samples, if any
for ( ; fr < keep; fr++) {
buffer[tndx][truncIndex+fr] = buffer[tndx][curOffset + fr];
}
}
truncIndex += keep;
}
consecutiveSilentFrames = 0;
ignoringFrames = false;
truncToMinimum = false;
}
// Can get here either because > dbThreshold
// or silence duration isn't longer than allowed
for (tndx = 0; tndx < tcount; tndx++) {
buffer[tndx][truncIndex] = buffer[tndx][i];
}
truncIndex++;
}
// Update tracks if any samples were removed, now or before
if (outTrackOffset + truncIndex != index + limit) {
// Put updated sample back into output tracks.
tndx = 0;
t = (WaveTrack *) iterOut.First();
while (t) {
t->Set((samplePtr)buffer[tndx++], floatSample, outTrackOffset, truncIndex);
t = (WaveTrack *) iterOut.Next();
}
}
// If currently in a silent section, retain samples for the next pass
if(ignoringFrames) {
keep = consecutiveSilentFrames - truncInitialAllowedSilentSamples;
if(keep > (truncLongestAllowedSilentSamples+mBlendFrameCount))
keep = truncLongestAllowedSilentSamples+mBlendFrameCount;
for (tndx = 0; tndx < tcount; tndx++) {
for(fr = 0; fr < truncInitialAllowedSilentSamples; fr++) {
buffer[tndx][fr] = buffer[tndx][truncIndex-truncInitialAllowedSilentSamples+fr];
}
for(fr = 0; fr < keep; fr++) {
buffer[tndx][truncInitialAllowedSilentSamples+fr] = buffer[tndx][i-keep+fr];
}
}
// Update the output index, less what we are retaining for next time
outTrackOffset += truncIndex - truncInitialAllowedSilentSamples;
// Append the following buffer to the existing data
i = consecutiveSilentFrames = truncInitialAllowedSilentSamples + keep;
truncIndex = truncInitialAllowedSilentSamples;
} else {
// Maintain output index
outTrackOffset += truncIndex;
// Reset the buffer pointers to the beginning
i = 0;
truncIndex = 0;
consecutiveSilentFrames = 0;
}
// Update progress and bail if user cancelled
cancelled = TrackProgress(0, ((double)index / (double)end));
if (cancelled) {
break;
}
// Bump to next block
index += count;
}
AudacityProject *p = GetActiveProject();
if (!p)
return false;
// Remove stale data at end of output tracks.
if (!cancelled && (outTrackOffset < end)) {
t = (WaveTrack *) iterOut.First();
if( p->IsSticky() )
t->Clear(outTrackOffset / rate, t1, mOutputTracks);
else
while(t) {
t->Clear(outTrackOffset / rate, t1, mOutputTracks);
t = (WaveTrack *) iterOut.Next();
}
t1 = outTrackOffset / rate;
}
// Free buffers
for (tndx = 0; tndx < tcount; tndx++) {
delete [] buffer[tndx];
}
delete [] buffer;
mT0 = t0;
mT1 = t1;
this->ReplaceProcessedTracks(!cancelled);
return !cancelled;
}
