bool EffectSoundTouch::Process()
{
// Assumes that mSoundTouch has already been initialized
// by the subclass for subclass-specific parameters.
//Iterate over each track
TrackListIterator iter(mWaveTracks);
WaveTrack* leftTrack = (WaveTrack*)(iter.First());
WaveTrack* rightTrack = NULL;
mCurTrackNum = 0;
m_maxNewLength = 0.0;
while (leftTrack) {
//Get start and end times from track
double trackStart = leftTrack->GetStartTime();
double trackEnd = leftTrack->GetEndTime();
//Set the current bounds to whichever left marker is
//greater and whichever right marker is less:
mCurT0 = mT0 < trackStart? trackStart: mT0;
mCurT1 = mT1 > trackEnd? trackEnd: mT1;
// Process only if the right marker is to the right of the left marker
if (mCurT1 > mCurT0) {
//Transform the marker timepoints to samples
longSampleCount start = leftTrack->TimeToLongSamples(mCurT0);
longSampleCount end = leftTrack->TimeToLongSamples(mCurT1);
rightTrack = NULL;
if (leftTrack->GetLinked()) {
rightTrack = (WaveTrack*)(iter.Next());
mSoundTouch->setChannels(2);
if (!ProcessStereo(leftTrack, rightTrack, start, end))
return false;
mCurTrackNum++; // Increment for rightTrack, too.
} else {
mSoundTouch->setChannels(1);
//ProcessOne() (implemented below) processes a single track
if (!ProcessOne(leftTrack, start, end))
return false;
}
}
//Iterate to the next track
leftTrack = (WaveTrack*)(iter.Next());
mCurTrackNum++;
}
delete mSoundTouch;
mSoundTouch = NULL;
mT1 = mT0 + m_maxNewLength; // Update selection.
return true;
}
//TODO: There are a lot of places where a track is being checked
// to see if it is stereo. Consolidate these
bool EffectStereoToMono::CheckWhetherSkipEffect()
{
TrackListIterator iter(mWaveTracks);
WaveTrack *t = (WaveTrack*)iter.First();
while (t) {
if (t->GetLinked()) {
return false;
}
t = (WaveTrack *)iter.Next();
}
return true;
}
bool VSTEffect::Init()
{
if (!mAEffect) {
Load();
}
if (!mAEffect) {
return false;
}
mBlockSize = 0;
TrackListIterator iter(mOutputTracks);
WaveTrack *left = (WaveTrack *) iter.First();
while (left) {
sampleCount lstart;
sampleCount llen;
GetSamples(left, &lstart, &llen);
if (left->GetLinked()) {
WaveTrack *right = (WaveTrack *) iter.Next();
sampleCount rstart;
sampleCount rlen;
GetSamples(right, &rstart, &rlen);
if (left->GetRate() != right->GetRate()) {
wxMessageBox(_("Both channels of a stereo track must be the same sample rate."));
return false;
}
if (llen != rlen) {
wxMessageBox(_("Both channels of a stereo track must be the same length."));
return false;
}
}
left = (WaveTrack *) iter.Next();
}
return true;
}
bool VampEffect::Init()
{
Vamp::HostExt::PluginLoader *loader =
Vamp::HostExt::PluginLoader::getInstance();
delete mPlugin;
mPlugin = 0;
TrackListIterator iter(mWaveTracks);
WaveTrack *left = (WaveTrack *)iter.First();
mRate = 0.0;
while (left) {
if (mRate == 0.0) mRate = left->GetRate();
if (left->GetLinked()) {
WaveTrack *right = (WaveTrack *)iter.Next();
if (left->GetRate() != right->GetRate()) {
wxMessageBox(_("Sorry, Vamp Plug-ins cannot be run on stereo tracks where the individual channels of the track do not match."));
return false;
}
}
left = (WaveTrack *)iter.Next();
}
if (mRate <= 0.0) mRate = mProjectRate;
mPlugin = loader->loadPlugin
(mKey, mRate, Vamp::HostExt::PluginLoader::ADAPT_ALL);
if (!mPlugin) {
wxMessageBox(_("Sorry, failed to load Vamp Plug-in."));
return false;
}
return true;
}
bool EffectNormalize::Process()
{
if (mGain == false && mDC == false)
return true;
float ratio;
if( mGain )
ratio = pow(10.0,TrapDouble(mLevel, // same value used for all tracks
NORMALIZE_DB_MIN,
NORMALIZE_DB_MAX)/20.0);
else
ratio = 1.0;
//Iterate over each track
this->CopyInputTracks(); // Set up mOutputTracks.
bool bGoodResult = true;
SelectedTrackListOfKindIterator iter(Track::Wave, mOutputTracks);
WaveTrack *track = (WaveTrack *) iter.First();
WaveTrack *prevTrack;
prevTrack = track;
mCurTrackNum = 0;
wxString topMsg;
if(mDC & mGain)
topMsg = _("Removing DC offset and Normalizing...\n");
else if(mDC & !mGain)
topMsg = _("Removing DC offset...\n");
else if(!mDC & mGain)
topMsg = _("Normalizing without removing DC offset...\n");
else if(!mDC & !mGain)
topMsg = wxT("Not doing anything)...\n"); // shouldn't get here
while (track) {
//Get start and end times from track
double trackStart = track->GetStartTime();
double trackEnd = track->GetEndTime();
//Set the current bounds to whichever left marker is
//greater and whichever right marker is less:
mCurT0 = mT0 < trackStart? trackStart: mT0;
mCurT1 = mT1 > trackEnd? trackEnd: mT1;
// Process only if the right marker is to the right of the left marker
if (mCurT1 > mCurT0) {
wxString msg;
wxString trackName = track->GetName();
if(!track->GetLinked() || mStereoInd)
msg = topMsg + _("Analyzing: ") + trackName;
else
msg = topMsg + _("Analyzing first track of stereo pair: ") + trackName;
AnalyseTrack(track, msg); // sets mOffset and offset-adjusted mMin and mMax
if(!track->GetLinked() || mStereoInd) { // mono or 'stereo tracks independently'
float extent = wxMax(fabs(mMax), fabs(mMin));
if( (extent > 0) && mGain )
mMult = ratio / extent;
else
mMult = 1.0;
msg = topMsg + _("Processing: ") + trackName;
if(track->GetLinked() || prevTrack->GetLinked()) // only get here if there is a linked track but we are processing independently
msg = topMsg + _("Processing stereo channels independently: ") + trackName;
if (!ProcessOne(track, msg))
{
bGoodResult = false;
break;
}
}
else
{
// we have a linked stereo track
// so we need to find it's min, max and offset
// as they are needed to calc the multiplier for both tracks
float offset1 = mOffset; // remember ones from first track
float min1 = mMin;
float max1 = mMax;
track = (WaveTrack *) iter.Next(); // get the next one
mCurTrackNum++; // keeps progress bar correct
msg = topMsg + _("Analyzing second track of stereo pair: ") + trackName;
AnalyseTrack(track, msg); // sets mOffset and offset-adjusted mMin and mMax
float offset2 = mOffset; // ones for second track
float min2 = mMin;
float max2 = mMax;
float extent = wxMax(fabs(min1), fabs(max1));
extent = wxMax(extent, fabs(min2));
extent = wxMax(extent, fabs(max2));
if( (extent > 0) && mGain )
mMult = ratio / extent; // we need to use this for both linked tracks
else
mMult = 1.0;
mOffset = offset1;
track = (WaveTrack *) iter.Prev(); // go back to the first linked one
mCurTrackNum--; // keeps progress bar correct
msg = topMsg + _("Processing first track of stereo pair: ") + trackName;
if (!ProcessOne(track, msg))
{
bGoodResult = false;
break;
}
mOffset = offset2;
track = (WaveTrack *) iter.Next(); // go to the second linked one
mCurTrackNum++; // keeps progress bar correct
msg = topMsg + _("Processing second track of stereo pair: ") + trackName;
if (!ProcessOne(track, msg))
{
bGoodResult = false;
break;
}
}
}
//Iterate to the next track
prevTrack = track;
track = (WaveTrack *) iter.Next();
mCurTrackNum++;
}
this->ReplaceProcessedTracks(bGoodResult);
return bGoodResult;
}
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;
}
bool ContrastDialog::GetDB(float &dB)
{
float rms = float(0.0);
int numberSelecteTracks = 0;
// For stereo tracks: sqrt((mean(L)+mean(R))/2)
bool isStereo = false;
double meanSq = 0.0;
AudacityProject *p = GetActiveProject();
SelectedTrackListOfKindIterator iter(Track::Wave, p->GetTracks());
WaveTrack *t = (WaveTrack *) iter.First();
while (t) {
numberSelecteTracks++;
if (numberSelecteTracks > 1 && !isStereo) {
AudacityMessageDialog m(NULL, _("You can only measure one track at a time."), _("Error"), wxOK);
m.ShowModal();
return false;
}
isStereo = t->GetLinked();
wxASSERT(mT0 <= mT1);
// Ignore whitespace beyond ends of track.
if(mT0 < t->GetStartTime())
mT0 = t->GetStartTime();
if(mT1 > t->GetEndTime())
mT1 = t->GetEndTime();
auto SelT0 = t->TimeToLongSamples(mT0);
auto SelT1 = t->TimeToLongSamples(mT1);
if(SelT0 > SelT1)
{
AudacityMessageDialog m(NULL, _("Invalid audio selection.\nPlease ensure that audio is selected."), _("Error"), wxOK);
m.ShowModal();
return false;
}
if(SelT0 == SelT1)
{
AudacityMessageDialog m(NULL, _("Nothing to measure.\nPlease select a section of a track."), _("Error"), wxOK);
m.ShowModal();
return false;
}
// Don't throw in this analysis dialog
rms = ((WaveTrack *)t)->GetRMS(mT0, mT1, false);
meanSq += rms * rms;
t = (WaveTrack *) iter.Next();
}
// TODO: This works for stereo, provided the audio clips are in both channels.
// We should really count gaps between clips as silence.
rms = (meanSq > 0.0)? sqrt(meanSq/(double)numberSelecteTracks) : 0.0;
if(numberSelecteTracks == 0) {
AudacityMessageDialog m(NULL, _("Please select an audio track."), _("Error"), wxOK);
m.ShowModal();
return false;
}
// Gives warning C4056, Overflow in floating-point constant arithmetic
// -INFINITY is intentional here.
// Looks like we are stuck with this warning, as
// #pragma warning( disable : 4056)
// even around the whole function does not disable it successfully.
dB = (rms == 0.0)? -INFINITY : LINEAR_TO_DB(rms);
return true;
}
bool VSTEffect::Process()
{
CopyInputTracks();
bool bGoodResult = true;
mInBuffer = NULL;
mOutBuffer = NULL;
TrackListIterator iter(mOutputTracks);
int count = 0;
bool clear = false;
WaveTrack *left = (WaveTrack *) iter.First();
while (left) {
WaveTrack *right;
sampleCount len;
sampleCount lstart;
sampleCount rstart;
GetSamples(left, &lstart, &len);
mChannels = 1;
right = NULL;
rstart = 0;
if (left->GetLinked() && mInputs > 1) {
right = (WaveTrack *) iter.Next();
GetSamples(right, &rstart, &len);
clear = false;
mChannels = 2;
}
if (mBlockSize == 0) {
mBlockSize = left->GetMaxBlockSize() * 2;
// Some VST effects (Antress Modern is an example), do not like
// overly large block sizes. Unfortunately, I have not found a
// way to determine if the effect has a maximum it will support,
// so just limit to small value for now. This will increase
// processing time and, it's a shame, because most plugins seem
// to be able to handle much larger sizes.
if (mBlockSize > 8192) { // The Antress limit
mBlockSize = 8192;
}
mInBuffer = new float *[mInputs];
for (int i = 0; i < mInputs; i++) {
mInBuffer[i] = new float[mBlockSize];
}
mOutBuffer = new float *[mOutputs];
for (int i = 0; i < mOutputs; i++) {
mOutBuffer[i] = new float[mBlockSize];
}
// Turn the power off
callDispatcher(effMainsChanged, 0, 0, NULL, 0.0);
// Set processing parameters
callDispatcher(effSetSampleRate, 0, 0, NULL, left->GetRate());
callDispatcher(effSetBlockSize, 0, mBlockSize, NULL, 0.0);
}
// Clear unused input buffers
if (!right && !clear) {
for (int i = 1; i < mInputs; i++) {
for (int j = 0; j < mBlockSize; j++) {
mInBuffer[i][j] = 0.0;
}
}
clear = true;
}
bGoodResult = ProcessStereo(count, left, right, lstart, rstart, len);
if (!bGoodResult) {
break;
}
left = (WaveTrack *) iter.Next();
count++;
}
if (mOutBuffer) {
for (int i = 0; i < mOutputs; i++) {
delete mOutBuffer[i];
}
delete [] mOutBuffer;
mOutBuffer = NULL;
}
if (mInBuffer) {
for (int i = 0; i < mInputs; i++) {
delete mInBuffer[i];
}
delete [] mInBuffer;
mInBuffer = NULL;
}
ReplaceProcessedTracks(bGoodResult);
return bGoodResult;
}
bool ContrastDialog::GetDB(float &dB)
{
float rms = float(0.0);
int numberSelecteTracks = 0;
// For stereo tracks: sqrt((mean(L)+mean(R))/2)
bool isStereo = false;
double meanSq = 0.0;
AudacityProject *p = GetActiveProject();
SelectedTrackListOfKindIterator iter(Track::Wave, p->GetTracks());
WaveTrack *t = (WaveTrack *) iter.First();
while (t) {
numberSelecteTracks++;
if (numberSelecteTracks > 1 && !isStereo) {
wxMessageDialog m(NULL, _("You can only measure one track at a time."), _("Error"), wxOK);
m.ShowModal();
return false;
}
isStereo = t->GetLinked();
wxASSERT(mT0 <= mT1);
// Ignore whitespace beyond ends of track.
if(mT0 < t->GetStartTime())
mT0 = t->GetStartTime();
if(mT1 > t->GetEndTime())
mT1 = t->GetEndTime();
sampleCount SelT0 = t->TimeToLongSamples(mT0);
sampleCount SelT1 = t->TimeToLongSamples(mT1);
if(SelT0 > SelT1)
{
wxMessageDialog m(NULL, _("Invalid audio selection.\nPlease ensure that audio is selected."), _("Error"), wxOK);
m.ShowModal();
return false;
}
if(SelT0 == SelT1)
{
wxMessageDialog m(NULL, _("Nothing to measure.\nPlease select a section of a track."), _("Error"), wxOK);
m.ShowModal();
return false;
}
((WaveTrack *)t)->GetRMS(&rms, mT0, mT1);
meanSq += rms * rms;
t = (WaveTrack *) iter.Next();
}
// TODO: This works for stereo, provided the audio clips are in both channels.
// We should really count gaps between clips as silence.
rms = (meanSq > 0.0)? sqrt(meanSq/(double)numberSelecteTracks) : 0.0;
if(numberSelecteTracks == 0) {
wxMessageDialog m(NULL, _("Please select an audio track."), _("Error"), wxOK);
m.ShowModal();
return false;
}
dB = (rms == 0.0)? -INFINITY : LINEAR_TO_DB(rms);
return true;
}
bool EffectSBSMS::Process()
{
bool bGoodResult = true;
//Iterate over each track
//Track::All is needed because this effect needs to introduce silence in the group tracks to keep sync
this->CopyInputTracks(Track::All); // Set up mOutputTracks.
TrackListIterator iter(mOutputTracks);
Track* t;
mCurTrackNum = 0;
double maxDuration = 0.0;
// Must sync if selection length will change
bool mustSync = (rateStart != rateEnd);
Slide rateSlide(rateSlideType,rateStart,rateEnd);
Slide pitchSlide(pitchSlideType,pitchStart,pitchEnd);
mTotalStretch = rateSlide.getTotalStretch();
t = iter.First();
while (t != NULL) {
if (t->GetKind() == Track::Label &&
(t->GetSelected() || (mustSync && t->IsSyncLockSelected())) )
{
if (!ProcessLabelTrack(t)) {
bGoodResult = false;
break;
}
}
else if (t->GetKind() == Track::Wave && t->GetSelected() )
{
WaveTrack* leftTrack = (WaveTrack*)t;
//Get start and end times from track
mCurT0 = leftTrack->GetStartTime();
mCurT1 = leftTrack->GetEndTime();
//Set the current bounds to whichever left marker is
//greater and whichever right marker is less
mCurT0 = wxMax(mT0, mCurT0);
mCurT1 = wxMin(mT1, mCurT1);
// Process only if the right marker is to the right of the left marker
if (mCurT1 > mCurT0) {
sampleCount start;
sampleCount end;
start = leftTrack->TimeToLongSamples(mCurT0);
end = leftTrack->TimeToLongSamples(mCurT1);
WaveTrack* rightTrack = NULL;
if (leftTrack->GetLinked()) {
double t;
rightTrack = (WaveTrack*)(iter.Next());
//Adjust bounds by the right tracks markers
t = rightTrack->GetStartTime();
t = wxMax(mT0, t);
mCurT0 = wxMin(mCurT0, t);
t = rightTrack->GetEndTime();
t = wxMin(mT1, t);
mCurT1 = wxMax(mCurT1, t);
//Transform the marker timepoints to samples
start = leftTrack->TimeToLongSamples(mCurT0);
end = leftTrack->TimeToLongSamples(mCurT1);
mCurTrackNum++; // Increment for rightTrack, too.
}
sampleCount trackStart = leftTrack->TimeToLongSamples(leftTrack->GetStartTime());
sampleCount trackEnd = leftTrack->TimeToLongSamples(leftTrack->GetEndTime());
// SBSMS has a fixed sample rate - we just convert to its sample rate and then convert back
float srTrack = leftTrack->GetRate();
float srProcess = bLinkRatePitch?srTrack:44100.0;
// the resampler needs a callback to supply its samples
ResampleBuf rb;
sampleCount maxBlockSize = leftTrack->GetMaxBlockSize();
rb.blockSize = maxBlockSize;
rb.buf = (audio*)calloc(rb.blockSize,sizeof(audio));
rb.leftTrack = leftTrack;
rb.rightTrack = rightTrack?rightTrack:leftTrack;
rb.leftBuffer = (float*)calloc(maxBlockSize,sizeof(float));
rb.rightBuffer = (float*)calloc(maxBlockSize,sizeof(float));
// Samples in selection
sampleCount samplesIn = end-start;
// Samples for SBSMS to process after resampling
sampleCount samplesToProcess = (sampleCount) ((float)samplesIn*(srProcess/srTrack));
SlideType outSlideType;
SBSMSResampleCB outResampleCB;
sampleCount processPresamples = 0;
sampleCount trackPresamples = 0;
if(bLinkRatePitch) {
rb.bPitch = true;
outSlideType = rateSlideType;
outResampleCB = resampleCB;
rb.offset = start;
rb.end = end;
rb.iface = new SBSMSInterfaceSliding(&rateSlide,&pitchSlide,
bPitchReferenceInput,
samplesToProcess,0,
NULL);
} else {
rb.bPitch = false;
outSlideType = (srProcess==srTrack?SlideIdentity:SlideConstant);
outResampleCB = postResampleCB;
rb.ratio = srProcess/srTrack;
rb.quality = new SBSMSQuality(&SBSMSQualityStandard);
rb.resampler = new Resampler(resampleCB, &rb, srProcess==srTrack?SlideIdentity:SlideConstant);
rb.sbsms = new SBSMS(rightTrack?2:1,rb.quality,true);
rb.SBSMSBlockSize = rb.sbsms->getInputFrameSize();
rb.SBSMSBuf = (audio*)calloc(rb.SBSMSBlockSize,sizeof(audio));
processPresamples = wxMin(rb.quality->getMaxPresamples(),
(long)((float)(start-trackStart)*(srProcess/srTrack)));
trackPresamples = wxMin(start-trackStart,
(long)((float)(processPresamples)*(srTrack/srProcess)));
rb.offset = start - trackPresamples;
rb.end = trackEnd;
rb.iface = new SBSMSEffectInterface(rb.resampler,
&rateSlide,&pitchSlide,
bPitchReferenceInput,
samplesToProcess,processPresamples,
rb.quality);
}
Resampler resampler(outResampleCB,&rb,outSlideType);
audio outBuf[SBSMSOutBlockSize];
float outBufLeft[2*SBSMSOutBlockSize];
float outBufRight[2*SBSMSOutBlockSize];
// Samples in output after SBSMS
sampleCount samplesToOutput = rb.iface->getSamplesToOutput();
// Samples in output after resampling back
sampleCount samplesOut = (sampleCount) ((float)samplesToOutput * (srTrack/srProcess));
// Duration in track time
double duration = (mCurT1-mCurT0) * mTotalStretch;
if(duration > maxDuration)
maxDuration = duration;
TimeWarper *warper = createTimeWarper(mCurT0,mCurT1,maxDuration,rateStart,rateEnd,rateSlideType);
SetTimeWarper(warper);
rb.outputLeftTrack = mFactory->NewWaveTrack(leftTrack->GetSampleFormat(),
leftTrack->GetRate());
if(rightTrack)
rb.outputRightTrack = mFactory->NewWaveTrack(rightTrack->GetSampleFormat(),
rightTrack->GetRate());
long pos = 0;
long outputCount = -1;
// process
while(pos<samplesOut && outputCount) {
long frames;
if(pos+SBSMSOutBlockSize>samplesOut) {
frames = samplesOut - pos;
} else {
frames = SBSMSOutBlockSize;
}
outputCount = resampler.read(outBuf,frames);
for(int i = 0; i < outputCount; i++) {
outBufLeft[i] = outBuf[i][0];
if(rightTrack)
outBufRight[i] = outBuf[i][1];
}
pos += outputCount;
rb.outputLeftTrack->Append((samplePtr)outBufLeft, floatSample, outputCount);
if(rightTrack)
rb.outputRightTrack->Append((samplePtr)outBufRight, floatSample, outputCount);
double frac = (double)pos/(double)samplesOut;
int nWhichTrack = mCurTrackNum;
if(rightTrack) {
nWhichTrack = 2*(mCurTrackNum/2);
if (frac < 0.5)
frac *= 2.0; // Show twice as far for each track, because we're doing 2 at once.
else {
nWhichTrack++;
frac -= 0.5;
frac *= 2.0; // Show twice as far for each track, because we're doing 2 at once.
}
}
if (TrackProgress(nWhichTrack, frac))
return false;
}
rb.outputLeftTrack->Flush();
if(rightTrack)
rb.outputRightTrack->Flush();
bool bResult =
leftTrack->ClearAndPaste(mCurT0, mCurT1, rb.outputLeftTrack,
true, false, GetTimeWarper());
wxASSERT(bResult); // TO DO: Actually handle this.
wxUnusedVar(bResult);
if(rightTrack)
{
bResult =
rightTrack->ClearAndPaste(mCurT0, mCurT1, rb.outputRightTrack,
true, false, GetTimeWarper());
wxASSERT(bResult); // TO DO: Actually handle this.
}
}
mCurTrackNum++;
}
else if (mustSync && t->IsSyncLockSelected())
{
t->SyncLockAdjust(mCurT1, mCurT0 + (mCurT1 - mCurT0) * mTotalStretch);
}
//Iterate to the next track
t = iter.Next();
}
if (bGoodResult)
ReplaceProcessedTracks(bGoodResult);
// Update selection
mT0 = mCurT0;
mT1 = mCurT0 + maxDuration;
return bGoodResult;
}
bool EffectSoundTouch::Process()
{
// Assumes that mSoundTouch has already been initialized
// by the subclass for subclass-specific parameters. The
// time warper should also be set.
// Check if this effect will alter the selection length; if so, we need
// to operate on sync-lock selected tracks.
bool mustSync = true;
if (mT1 == GetTimeWarper()->Warp(mT1)) {
mustSync = false;
}
//Iterate over each track
// Needs Track::All for sync-lock grouping.
this->CopyInputTracks(Track::All);
bool bGoodResult = true;
TrackListIterator iter(mOutputTracks);
Track* t;
mCurTrackNum = 0;
m_maxNewLength = 0.0;
t = iter.First();
while (t != NULL) {
if (t->GetKind() == Track::Label &&
(t->GetSelected() || (mustSync && t->IsSyncLockSelected())) )
{
if (!ProcessLabelTrack(t))
{
bGoodResult = false;
break;
}
}
#ifdef USE_MIDI
else if (t->GetKind() == Track::Note &&
(t->GetSelected() || (mustSync && t->IsSyncLockSelected())))
{
if (!ProcessNoteTrack(t))
{
bGoodResult = false;
break;
}
}
#endif
else if (t->GetKind() == Track::Wave && t->GetSelected())
{
WaveTrack* leftTrack = (WaveTrack*)t;
//Get start and end times from track
mCurT0 = leftTrack->GetStartTime();
mCurT1 = leftTrack->GetEndTime();
//Set the current bounds to whichever left marker is
//greater and whichever right marker is less
mCurT0 = wxMax(mT0, mCurT0);
mCurT1 = wxMin(mT1, mCurT1);
// Process only if the right marker is to the right of the left marker
if (mCurT1 > mCurT0) {
sampleCount start, end;
if (leftTrack->GetLinked()) {
double t;
WaveTrack* rightTrack = (WaveTrack*)(iter.Next());
//Adjust bounds by the right tracks markers
t = rightTrack->GetStartTime();
t = wxMax(mT0, t);
mCurT0 = wxMin(mCurT0, t);
t = rightTrack->GetEndTime();
t = wxMin(mT1, t);
mCurT1 = wxMax(mCurT1, t);
//Transform the marker timepoints to samples
start = leftTrack->TimeToLongSamples(mCurT0);
end = leftTrack->TimeToLongSamples(mCurT1);
//Inform soundtouch there's 2 channels
mSoundTouch->setChannels(2);
//ProcessStereo() (implemented below) processes a stereo track
if (!ProcessStereo(leftTrack, rightTrack, start, end))
{
bGoodResult = false;
break;
}
mCurTrackNum++; // Increment for rightTrack, too.
} else {
//Transform the marker timepoints to samples
start = leftTrack->TimeToLongSamples(mCurT0);
end = leftTrack->TimeToLongSamples(mCurT1);
//Inform soundtouch there's a single channel
mSoundTouch->setChannels(1);
//ProcessOne() (implemented below) processes a single track
if (!ProcessOne(leftTrack, start, end))
{
bGoodResult = false;
break;
}
}
}
mCurTrackNum++;
}
else if (mustSync && t->IsSyncLockSelected()) {
t->SyncLockAdjust(mT1, GetTimeWarper()->Warp(mT1));
}
//Iterate to the next track
t = iter.Next();
}
if (bGoodResult)
ReplaceProcessedTracks(bGoodResult);
delete mSoundTouch;
mSoundTouch = NULL;
// mT0 = mCurT0;
// mT1 = mCurT0 + m_maxNewLength; // Update selection.
return bGoodResult;
}
bool EffectSBSMS::Process()
{
if(!bInit) {
sbsms_init(4096);
bInit = TRUE;
}
bool bGoodResult = true;
//Iterate over each track
//Track::All is needed because this effect needs to introduce silence in the group tracks to keep sync
this->CopyInputTracks(Track::All); // Set up mOutputTracks.
TrackListIterator iter(mOutputTracks);
Track* t;
mCurTrackNum = 0;
double maxDuration = 0.0;
if(rateStart == rateEnd)
mTotalStretch = 1.0/rateStart;
else
mTotalStretch = 1.0/(rateEnd-rateStart)*log(rateEnd/rateStart);
// Must sync if selection length will change
bool mustSync = (mTotalStretch != 1.0);
t = iter.First();
while (t != NULL) {
if (t->GetKind() == Track::Label &&
(t->GetSelected() || (mustSync && t->IsSynchroSelected())) )
{
if (!ProcessLabelTrack(t)) {
bGoodResult = false;
break;
}
}
else if (t->GetKind() == Track::Wave && t->GetSelected() )
{
WaveTrack* leftTrack = (WaveTrack*)t;
//Get start and end times from track
mCurT0 = leftTrack->GetStartTime();
mCurT1 = leftTrack->GetEndTime();
//Set the current bounds to whichever left marker is
//greater and whichever right marker is less
mCurT0 = wxMax(mT0, mCurT0);
mCurT1 = wxMin(mT1, mCurT1);
// Process only if the right marker is to the right of the left marker
if (mCurT1 > mCurT0) {
sampleCount start;
sampleCount end;
start = leftTrack->TimeToLongSamples(mCurT0);
end = leftTrack->TimeToLongSamples(mCurT1);
WaveTrack* rightTrack = NULL;
if (leftTrack->GetLinked()) {
double t;
rightTrack = (WaveTrack*)(iter.Next());
//Adjust bounds by the right tracks markers
t = rightTrack->GetStartTime();
t = wxMax(mT0, t);
mCurT0 = wxMin(mCurT0, t);
t = rightTrack->GetEndTime();
t = wxMin(mT1, t);
mCurT1 = wxMax(mCurT1, t);
//Transform the marker timepoints to samples
start = leftTrack->TimeToLongSamples(mCurT0);
end = leftTrack->TimeToLongSamples(mCurT1);
mCurTrackNum++; // Increment for rightTrack, too.
}
sampleCount trackEnd = leftTrack->TimeToLongSamples(leftTrack->GetEndTime());
// SBSMS has a fixed sample rate - we just convert to its sample rate and then convert back
float srIn = leftTrack->GetRate();
float srSBSMS = 44100.0;
// the resampler needs a callback to supply its samples
resampleBuf rb;
sampleCount maxBlockSize = leftTrack->GetMaxBlockSize();
rb.block = maxBlockSize;
rb.buf = (audio*)calloc(rb.block,sizeof(audio));
rb.leftTrack = leftTrack;
rb.rightTrack = rightTrack?rightTrack:leftTrack;
rb.leftBuffer = (float*)calloc(maxBlockSize,sizeof(float));
rb.rightBuffer = (float*)calloc(maxBlockSize,sizeof(float));
rb.offset = start;
rb.end = trackEnd;
rb.ratio = srSBSMS/srIn;
rb.resampler = new Resampler(resampleCB, &rb);
// Samples in selection
sampleCount samplesIn = end-start;
// Samples for SBSMS to process after resampling
sampleCount samplesToProcess = (sampleCount) ((real)samplesIn*(srSBSMS/srIn));
// Samples in output after resampling back
sampleCount samplesToGenerate = (sampleCount) ((real)samplesToProcess * mTotalStretch);
sampleCount samplesOut = (sampleCount) ((real)samplesIn * mTotalStretch);
double duration = (mCurT1-mCurT0) * mTotalStretch;
if(duration > maxDuration)
maxDuration = duration;
TimeWarper *warper = NULL;
if (rateStart == rateEnd)
{
warper = new LinearTimeWarper(mCurT0, mCurT0,
mCurT1, mCurT0+maxDuration);
} else
{
warper = new LogarithmicTimeWarper(mCurT0, mCurT1,
rateStart, rateEnd);
}
SetTimeWarper(warper);
sbsmsInfo si;
si.rs = rb.resampler;
si.samplesToProcess = samplesToProcess;
si.samplesToGenerate = samplesToGenerate;
si.stretch0 = rateStart;
si.stretch1 = rateEnd;
si.ratio0 = pitchStart;
si.ratio1 = pitchEnd;
rb.sbsmser = sbsms_create(&samplesCB,&stretchCB,&ratioCB,rightTrack?2:1,quality,bPreAnalyze,true);
rb.pitch = pitch_create(rb.sbsmser,&si,srIn/srSBSMS);
rb.outputLeftTrack = mFactory->NewWaveTrack(leftTrack->GetSampleFormat(),
leftTrack->GetRate());
if(rightTrack)
rb.outputRightTrack = mFactory->NewWaveTrack(rightTrack->GetSampleFormat(),
rightTrack->GetRate());
sampleCount blockSize = SBSMS_FRAME_SIZE[quality];
rb.outBuf = (audio*)calloc(blockSize,sizeof(audio));
rb.outputLeftBuffer = (float*)calloc(blockSize*2,sizeof(float));
if(rightTrack)
rb.outputRightBuffer = (float*)calloc(blockSize*2,sizeof(float));
long pos = 0;
long outputCount = -1;
// pre analysis
real fracPre = 0.0f;
if(bPreAnalyze) {
fracPre = 0.05f;
resampleBuf rbPre;
rbPre.block = maxBlockSize;
rbPre.buf = (audio*)calloc(rb.block,sizeof(audio));
rbPre.leftTrack = leftTrack;
rbPre.rightTrack = rightTrack?rightTrack:leftTrack;
rbPre.leftBuffer = (float*)calloc(maxBlockSize,sizeof(float));
rbPre.rightBuffer = (float*)calloc(maxBlockSize,sizeof(float));
rbPre.offset = start;
rbPre.end = end;
rbPre.ratio = srSBSMS/srIn;
rbPre.resampler = new Resampler(resampleCB, &rbPre);
si.rs = rbPre.resampler;
long pos = 0;
long lastPos = 0;
long ret = 0;
while(lastPos<samplesToProcess) {
ret = sbsms_pre_analyze(&samplesCB,&si,rb.sbsmser);
lastPos = pos;
pos += ret;
real completion = (real)lastPos/(real)samplesToProcess;
if (TrackProgress(0,fracPre*completion))
return false;
}
sbsms_pre_analyze_complete(rb.sbsmser);
sbsms_reset(rb.sbsmser);
si.rs = rb.resampler;
}
// process
while(pos<samplesOut && outputCount) {
long frames;
if(pos+blockSize>samplesOut) {
frames = samplesOut - pos;
} else {
frames = blockSize;
}
outputCount = pitch_process(rb.outBuf, frames, rb.pitch);
for(int i = 0; i < outputCount; i++) {
rb.outputLeftBuffer[i] = rb.outBuf[i][0];
if(rightTrack)
rb.outputRightBuffer[i] = rb.outBuf[i][1];
}
pos += outputCount;
rb.outputLeftTrack->Append((samplePtr)rb.outputLeftBuffer, floatSample, outputCount);
if(rightTrack)
rb.outputRightTrack->Append((samplePtr)rb.outputRightBuffer, floatSample, outputCount);
double frac = (double)pos/(double)samplesOut;
int nWhichTrack = mCurTrackNum;
if(rightTrack) {
nWhichTrack = 2*(mCurTrackNum/2);
if (frac < 0.5)
frac *= 2.0; // Show twice as far for each track, because we're doing 2 at once.
else {
nWhichTrack++;
frac -= 0.5;
frac *= 2.0; // Show twice as far for each track, because we're doing 2 at once.
}
}
if (TrackProgress(nWhichTrack, fracPre + (1.0-fracPre)*frac))
return false;
}
rb.outputLeftTrack->Flush();
if(rightTrack)
rb.outputRightTrack->Flush();
leftTrack->ClearAndPaste(mCurT0, mCurT1, rb.outputLeftTrack,
true, false, GetTimeWarper());
if(rightTrack) {
rightTrack->ClearAndPaste(mCurT0, mCurT1, rb.outputRightTrack,
true, false, GetTimeWarper());
}
}
mCurTrackNum++;
}
else if (mustSync && t->IsSynchroSelected())
{
t->SyncAdjust(mCurT1, mCurT0 + (mCurT1 - mCurT0) * mTotalStretch);
}
//Iterate to the next track
t = iter.Next();
}
if (bGoodResult)
ReplaceProcessedTracks(bGoodResult);
// Update selection
mT0 = mCurT0;
mT1 = mCurT0 + maxDuration;
return bGoodResult;
}
bool EffectNormalize::Process()
{
bool wasLinked = false; // set when a track has a linked (stereo) track
if (mGain == false &&
mDC == false)
return true;
//Iterate over each track
this->CopyInputTracks(); // Set up mOutputTracks.
bool bGoodResult = true;
SelectedTrackListOfKindIterator iter(Track::Wave, mOutputTracks);
WaveTrack *track = (WaveTrack *) iter.First();
mCurTrackNum = 0;
while (track) {
//Get start and end times from track
double trackStart = track->GetStartTime();
double trackEnd = track->GetEndTime();
//Set the current bounds to whichever left marker is
//greater and whichever right marker is less:
mCurT0 = mT0 < trackStart? trackStart: mT0;
mCurT1 = mT1 > trackEnd? trackEnd: mT1;
// Process only if the right marker is to the right of the left marker
if (mCurT1 > mCurT0) {
//Transform the marker timepoints to samples
sampleCount start = track->TimeToLongSamples(mCurT0);
sampleCount end = track->TimeToLongSamples(mCurT1);
//Get the track rate and samples
mCurRate = track->GetRate();
mCurChannel = track->GetChannel();
if(mStereoInd) // do stereo tracks independently (the easy way)
track->GetMinMax(&mMin, &mMax, mCurT0, mCurT1);
else
{
if(!wasLinked) // new mono track or first of a stereo pair
{
track->GetMinMax(&mMin, &mMax, mCurT0, mCurT1);
if(track->GetLinked())
{
wasLinked = true; // so we use these values for the next (linked) track
track = (WaveTrack *) iter.Next(); // get the next one for the max/min
float min, max;
track->GetMinMax(&min, &max, mCurT0, mCurT1);
mMin = min < mMin ? min : mMin;
mMax = max > mMax ? max : mMax;
track = (WaveTrack *) iter.Prev(); // back to the one we are on
}
}
else
wasLinked = false; // second of the stereo pair, next one is mono or first
}
//ProcessOne() (implemented below) processes a single track
if (!ProcessOne(track, start, end))
{
bGoodResult = false;
break;
}
}
//Iterate to the next track
track = (WaveTrack *) iter.Next();
mCurTrackNum++;
}
this->ReplaceProcessedTracks(bGoodResult);
return bGoodResult;
}