bool EffectChangeSpeed::Process()
{
// Similar to EffectSoundTouch::Process()
//Iterate over each track
TrackListIterator iter(mWaveTracks);
WaveTrack *track = (WaveTrack *) iter.First();
mCurTrackNum = 0;
m_maxNewLength = 0.0;
double curT0;
double curT1;
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:
curT0 = mT0 < trackStart? trackStart: mT0;
curT1 = mT1 > trackEnd? trackEnd: mT1;
// Process only if the right marker is to the right of the left marker
if (curT1 > curT0) {
//Transform the marker timepoints to samples
longSampleCount start = track->TimeToLongSamples(curT0);
longSampleCount end = track->TimeToLongSamples(curT1);
//ProcessOne() (implemented below) processes a single track
if (!ProcessOne(track, start, end))
return false;
}
//Iterate to the next track
track = (WaveTrack *) iter.Next();
mCurTrackNum++;
}
mT1 = mT0 + m_maxNewLength; // Update selection.
return true;
}
bool EffectNoiseRemoval::Process()
{
// If we are creating a profile, we don't care whether we have
// one already. We just prepare the counters.
if (mDoProfile) {
for(int i=0; i<windowSize; i++) {
sum[i] = float(0.0);
sumsq[i] = float(0.0);
profileCount[i] = 0;
}
}
else
{
// We need a profile.
if( !mHasProfile )
{
CleanSpeechMayReadNoisegate();
}
// If we still don't have a profile we have a problem.
if( !mHasProfile)
{
wxMessageBox( _("Attempt to run Noise Removal without a noise profile\n.") );
return false;
}
}
// This same code will both remove noise and
// profile it, depending on 'mDoProfile'
TrackListIterator iter(mWaveTracks);
WaveTrack *track = (WaveTrack *) iter.First();
int count = 0;
while (track) {
double trackStart = track->GetStartTime();
double trackEnd = track->GetEndTime();
double t0 = mT0 < trackStart? trackStart: mT0;
double t1 = mT1 > trackEnd? trackEnd: mT1;
if (t1 > t0) {
longSampleCount start = track->TimeToLongSamples(t0);
longSampleCount end = track->TimeToLongSamples(t1);
sampleCount len = (sampleCount)(end - start);
if (!ProcessOne(count, track, start, len)){
return false;
}
}
track = (WaveTrack *) iter.Next();
count++;
}
if (mDoProfile) {
for(int i=0; i<=windowSize/2; i++) {
//float stddev = sqrt(sumsq[i] - (sum[i]*sum[i])/profileCount[i])
// / profileCount[i];
mNoiseGate[i] = sum[i] / profileCount[i]; // average
}
CleanSpeechMayWriteNoiseGate();
mHasProfile = true;
mDoProfile = false;
}
return true;
}
void Effect::Preview()
{
wxWindow* FocusDialog = wxWindow::FindFocus();
if (gAudioIO->IsBusy())
return;
// Mix a few seconds of audio from all of the tracks
double previewLen = 6.0;
gPrefs->Read(wxT("/AudioIO/EffectsPreviewLen"), &previewLen);
WaveTrack *mixLeft = NULL;
WaveTrack *mixRight = NULL;
double rate = mProjectRate;
double t0 = mT0;
double t1 = t0 + previewLen;
if (t1 > mT1)
t1 = mT1;
if (t1 <= t0)
return;
bool success = ::MixAndRender(mTracks, mFactory, rate, floatSample, t0, t1,
&mixLeft, &mixRight);
if (!success) {
return;
}
// Save the original track list
TrackList *saveTracks = mTracks;
// Build new tracklist from rendering tracks
mTracks = new TrackList();
mixLeft->SetSelected(true);
mTracks->Add(mixLeft);
if (mixRight) {
mixRight->SetSelected(true);
mTracks->Add(mixRight);
}
// Update track/group counts
CountWaveTracks();
// Reset times
t0 = mixLeft->GetStartTime();
t1 = mixLeft->GetEndTime();
double t0save = mT0;
double t1save = mT1;
mT0 = t0;
mT1 = t1;
// Apply effect
// Effect is already inited; we call Process, End, and then Init
// again, so the state is exactly the way it was before Preview
// was called.
mProgress = new ProgressDialog(StripAmpersand(GetEffectName()),
_("Preparing preview"),
pdlgHideCancelButton); // Have only "Stop" button.
bool bSuccess = Process();
delete mProgress;
End();
Init();
if (bSuccess)
{
mT0 = t0save;
mT1 = t1save;
WaveTrackArray playbackTracks;
WaveTrackArray recordingTracks;
// Probably not the same tracks post-processing, so can't rely on previous values of mixLeft & mixRight.
TrackListOfKindIterator iter(Track::Wave, mTracks);
mixLeft = (WaveTrack*)(iter.First());
mixRight = (WaveTrack*)(iter.Next());
playbackTracks.Add(mixLeft);
if (mixRight)
playbackTracks.Add(mixRight);
#ifdef EXPERIMENTAL_MIDI_OUT
NoteTrackArray empty;
#endif
// Start audio playing
int token =
gAudioIO->StartStream(playbackTracks, recordingTracks,
#ifdef EXPERIMENTAL_MIDI_OUT
empty,
#endif
NULL, rate, t0, t1, NULL);
if (token) {
int previewing = eProgressSuccess;
mProgress = new ProgressDialog(StripAmpersand(GetEffectName()),
_("Previewing"), pdlgHideCancelButton);
while (gAudioIO->IsStreamActive(token) && previewing == eProgressSuccess) {
::wxMilliSleep(100);
previewing = mProgress->Update(gAudioIO->GetStreamTime() - t0, t1 - t0);
}
gAudioIO->StopStream();
while (gAudioIO->IsBusy()) {
::wxMilliSleep(100);
}
delete mProgress;
}
else {
wxMessageBox(_("Error while opening sound device. Please check the output device settings and the project sample rate."),
_("Error"), wxOK | wxICON_EXCLAMATION, FocusDialog);
}
}
if (FocusDialog) {
FocusDialog->SetFocus();
}
delete mOutputTracks;
mOutputTracks = NULL;
mTracks->Clear(true); // true => delete the tracks
delete mTracks;
mTracks = saveTracks;
}
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 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;
}
bool EffectChangeSpeed::Process()
{
// Similar to EffectSoundTouch::Process()
//Iterate over each track
this->CopyInputWaveTracks(); // Set up mOutputWaveTracks.
bool bGoodResult = true;
TrackListIterator iter(mOutputWaveTracks);
WaveTrack* pOutWaveTrack = (WaveTrack*)(iter.First());
mCurTrackNum = 0;
m_maxNewLength = 0.0;
//Get start and end times from track
mCurT0 = pOutWaveTrack->GetStartTime();
mCurT1 = pOutWaveTrack->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);
double len = pOutWaveTrack->GetEndTime() - pOutWaveTrack->GetStartTime();
while (pOutWaveTrack != NULL)
{
//Get start and end times from track
mCurT0 = pOutWaveTrack->GetStartTime();
mCurT1 = pOutWaveTrack->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) {
//Transform the marker timepoints to samples
sampleCount start = pOutWaveTrack->TimeToLongSamples(mCurT0);
sampleCount end = pOutWaveTrack->TimeToLongSamples(mCurT1);
//ProcessOne() (implemented below) processes a single track
if (!ProcessOne(pOutWaveTrack, start, end))
{
bGoodResult = false;
break;
}
}
//Iterate to the next track
pOutWaveTrack = (WaveTrack*)(iter.Next());
mCurTrackNum++;
}
this->ReplaceProcessedWaveTracks(bGoodResult);
#ifdef EXPERIMENTAL_FULL_LINKING
AudacityProject *p = (AudacityProject*)mParent;
if( p && p->IsSticky() ){
pOutWaveTrack = (WaveTrack*)(iter.First());
double newLen = pOutWaveTrack->GetEndTime() - pOutWaveTrack->GetStartTime();
double timeAdded = newLen-len;
double sel = mCurT1-mCurT0;
double percent = (sel/(timeAdded+sel))*100 - 100;
if ( !(HandleGroupChangeSpeed(percent, mCurT0, mCurT1)) ) bGoodResult = false;
}
#endif
// mT1 = mT0 + m_maxNewLength; // Update selection.
return bGoodResult;
}
// Deduce m_FromFrequency from the samples at the beginning of
// the selection. Then set some other params accordingly.
void EffectChangePitch::DeduceFrequencies()
{
// As a neat trick, attempt to get the frequency of the note at the
// beginning of the selection.
SelectedTrackListOfKindIterator iter(Track::Wave, mTracks);
WaveTrack *track = (WaveTrack *) iter.First();
if (track) {
double rate = track->GetRate();
// Auto-size window -- high sample rates require larger windowSize.
// Aim for around 2048 samples at 44.1 kHz (good down to about 100 Hz).
// To detect single notes, analysis period should be about 0.2 seconds.
// windowSize must be a power of 2.
int windowSize = wxRound(pow(2.0, floor((log(rate / 20.0)/log(2.0)) + 0.5)));
// windowSize < 256 too inaccurate
windowSize = (windowSize > 256)? windowSize : 256;
// we want about 0.2 seconds to catch the first note.
// number of windows rounded to nearest integer >= 1.
int numWindows = wxRound((double)(rate / (5.0f * windowSize)));
numWindows = (numWindows > 0)? numWindows : 1;
double trackStart = track->GetStartTime();
double t0 = mT0 < trackStart? trackStart: mT0;
sampleCount start = track->TimeToLongSamples(t0);
int analyzeSize = windowSize * numWindows;
float * buffer;
buffer = new float[analyzeSize];
float * freq;
freq = new float[windowSize/2];
float * freqa;
freqa = new float[windowSize/2];
int i, j, argmax;
int lag;
for(j=0; j<windowSize/2; j++)
freqa[j] = 0;
track->Get((samplePtr) buffer, floatSample, start, analyzeSize);
for(i=0; i<numWindows; i++) {
ComputeSpectrum(buffer+i*windowSize, windowSize,
windowSize, rate, freq, true);
for(j=0; j<windowSize/2; j++)
freqa[j] += freq[j];
}
argmax=0;
for(j=1; j<windowSize/2; j++)
if (freqa[j] > freqa[argmax])
argmax = j;
delete [] freq;
delete [] freqa;
delete [] buffer;
lag = (windowSize/2 - 1) - argmax;
m_dStartFrequency = rate / lag;
}
double dFromMIDInote = FreqToMIDInote(m_dStartFrequency);
double dToMIDInote = dFromMIDInote + m_dSemitonesChange;
m_nFromPitch = PitchIndex(dFromMIDInote);
m_nFromOctave = PitchOctave(dFromMIDInote);
m_nToPitch = PitchIndex(dToMIDInote);
m_nToOctave = PitchOctave(dToMIDInote);
m_FromFrequency = m_dStartFrequency;
Calc_PercentChange();
Calc_ToFrequency();
}
bool EffectRepair::Process()
{
//v This may be too much copying for EffectRepair. To support Cancel, may be able to copy much less.
// But for now, Cancel isn't supported without this.
this->CopyInputTracks(); // Set up mOutputTracks. //v This may be too much copying for EffectRepair.
bool bGoodResult = true;
SelectedTrackListOfKindIterator iter(Track::Wave, mOutputTracks.get());
WaveTrack *track = (WaveTrack *) iter.First();
int count = 0;
while (track) {
const
double trackStart = track->GetStartTime();
const double repair_t0 = std::max(mT0, trackStart);
const
double trackEnd = track->GetEndTime();
const double repair_t1 = std::min(mT1, trackEnd);
const
double repair_deltat = repair_t1 - repair_t0;
if (repair_deltat > 0) { // selection is within track audio
const auto repair0 = track->TimeToLongSamples(repair_t0);
const auto repair1 = track->TimeToLongSamples(repair_t1);
const auto repairLen = repair1 - repair0;
if (repairLen > 128) {
::Effect::MessageBox(_("The Repair effect is intended to be used on very short sections of damaged audio (up to 128 samples).\n\nZoom in and select a tiny fraction of a second to repair."));
bGoodResult = false;
break;
}
const double rate = track->GetRate();
const double spacing = std::max(repair_deltat * 2, 128. / rate);
const double t0 = std::max(repair_t0 - spacing, trackStart);
const double t1 = std::min(repair_t1 + spacing, trackEnd);
const auto s0 = track->TimeToLongSamples(t0);
const auto s1 = track->TimeToLongSamples(t1);
// The difference is at most 2 * 128:
const auto repairStart = (repair0 - s0).as_size_t();
const auto len = s1 - s0;
if (s0 == repair0 && s1 == repair1) {
::Effect::MessageBox(_("Repair works by using audio data outside the selection region.\n\nPlease select a region that has audio touching at least one side of it.\n\nThe more surrounding audio, the better it performs."));
/// The Repair effect needs some data to go on.\n\nPlease select an area to repair with some audio on at least one side (the more the better)."));
bGoodResult = false;
break;
}
if (!ProcessOne(count, track, s0,
// len is at most 5 * 128.
len.as_size_t(),
repairStart,
// repairLen is at most 128.
repairLen.as_size_t() )) {
bGoodResult = false;
break;
}
}
track = (WaveTrack *) iter.Next();
count++;
}
this->ReplaceProcessedTracks(bGoodResult);
return bGoodResult;
}
bool EffectTruncSilence::Process()
{
TrackListIterator iter(mWaveTracks);
WaveTrack *t;
double t0 = mT0;
double t1 = mT1;
int tndx;
int tcount = 0;
// 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();
}
// Transform the marker timepoints to samples
t = (WaveTrack *) iter.First();
longSampleCount start = t->TimeToLongSamples(t0);
longSampleCount end = t->TimeToLongSamples(t1);
// Bigger buffers reduce 'reset'
blockLen *= 8;
// Allocate buffers
float **buffer = new float*[tcount];
for (tndx = 0; tndx < tcount; tndx++) {
buffer[tndx] = new float[blockLen];
}
// 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 truncLongestAllowedSilentSamples =
int((wxMax( mTruncLongestAllowedSilentMs, minTruncMs) * rate) / 1000.0);
// Figure out number of frames for ramping
int quarterSecondFrames = int((rate * QUARTER_SECOND_MS) / 1000.0);
int rampInFrames = (truncLongestAllowedSilentSamples / 4);
if (rampInFrames > quarterSecondFrames) {
rampInFrames = quarterSecondFrames;
}
// Start processing
this->CopyInputWaveTracks(); // Set up mOutputWaveTracks.
TrackListIterator iterOut(mOutputWaveTracks);
longSampleCount index = start;
longSampleCount outTrackOffset = start;
bool cancelled = false;
while (index < end) {
// Limit size of current block if we've reached the end
sampleCount limit = blockLen;
if ((index + blockLen) > end) {
limit = end - index;
}
// Fill the buffers
tndx = 0;
t = (WaveTrack *) iter.First();
while (t) {
t->Get((samplePtr)buffer[tndx++], floatSample, index, blockLen);
t = (WaveTrack *) iter.Next();
}
// Reset
bool ignoringFrames = false;
sampleCount consecutiveSilentFrames = 0;
sampleCount truncIndex = 0;
// Look for silences in current block
for (sampleCount i = 0; 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;
}
}
// 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 > truncLongestAllowedSilentSamples) {
ignoringFrames = true;
continue;
}
}
else {
if (ignoringFrames == true) {
sampleCount curOffset = i - rampInFrames;
truncIndex -= rampInFrames; // backup into ignored frames
for (tndx = 0; tndx < tcount; tndx++) {
sampleCount trunci = truncIndex;
for (int fr = 0; fr < rampInFrames; fr++) {
buffer[tndx][trunci++] = buffer[tndx][curOffset + fr];
}
if(((trunci - rampInFrames) - mBlendFrameCount) >= 0) {
BlendFrames(buffer[tndx], mBlendFrameCount,
((trunci - rampInFrames) - mBlendFrameCount),
((i - rampInFrames) - mBlendFrameCount));
}
}
truncIndex += rampInFrames;
}
consecutiveSilentFrames = 0;
ignoringFrames = 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
if (truncIndex < 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();
}
}
// Maintain output index
outTrackOffset += truncIndex;
// Update progress and bail if user cancelled
cancelled = TrackProgress(0, ((double)index / (double)end));
if (cancelled) {
break;
}
// Bump to next block
index += limit;
}
// Remove stale data at end of output tracks.
if (!cancelled && (outTrackOffset < end)) {
t = (WaveTrack *) iterOut.First();
while (t) {
t->Clear(outTrackOffset / rate, t1);
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->ReplaceProcessedWaveTracks(!cancelled);
return !cancelled;
}
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 EffectRepair::Process()
{
//v This may be too much copying for EffectRepair. To support Cancel, may be able to copy much less.
// But for now, Cancel isn't supported without this.
this->CopyInputTracks(); // Set up mOutputTracks. //v This may be too much copying for EffectRepair.
bool bGoodResult = true;
SelectedTrackListOfKindIterator iter(Track::Wave, mOutputTracks);
WaveTrack *track = (WaveTrack *) iter.First();
int count = 0;
while (track) {
double trackStart = track->GetStartTime();
double trackEnd = track->GetEndTime();
double repair_t0 = mT0;
double repair_t1 = mT1;
repair_t0 = (repair_t0 < trackStart? trackStart: repair_t0);
repair_t1 = (repair_t1 > trackEnd? trackEnd: repair_t1);
if (repair_t0 < repair_t1) { // selection is within track audio
double rate = track->GetRate();
double repair_deltat = repair_t1 - repair_t0;
double spacing = repair_deltat * 2;
if (spacing < 128. / rate)
spacing = 128. / rate;
double t0 = repair_t0 - spacing;
double t1 = repair_t1 + spacing;
t0 = t0 < trackStart? trackStart: t0;
t1 = t1 > trackEnd? trackEnd: t1;
repair_t0 = (repair_t0 < t0? t0: repair_t0);
repair_t1 = (repair_t1 > t1? t1: repair_t1);
sampleCount s0 = track->TimeToLongSamples(t0);
sampleCount repair0 = track->TimeToLongSamples(repair_t0);
sampleCount repair1 = track->TimeToLongSamples(repair_t1);
sampleCount s1 = track->TimeToLongSamples(t1);
sampleCount repairStart = (sampleCount)(repair0 - s0);
sampleCount repairLen = (sampleCount)(repair1 - repair0);
sampleCount len = (sampleCount)(s1 - s0);
if (repairLen > 128) {
::wxMessageBox(_("The Repair effect is intended to be used on very short sections of damaged audio (up to 128 samples).\n\nZoom in and select a tiny fraction of a second to repair."));
bGoodResult = false;
break;
}
if (s0 == repair0 && s1 == repair1) {
::wxMessageBox(_("Repair works by using audio data outside the selection region.\n\nPlease select a region that has audio touching at least one side of it.\n\nThe more surrounding audio, the better it performs."));
/// The Repair effect needs some data to go on.\n\nPlease select an area to repair with some audio on at least one side (the more the better)."));
bGoodResult = false;
break;
}
if (!ProcessOne(count, track,
s0, len, repairStart, repairLen)) {
bGoodResult = false;
break;
}
}
track = (WaveTrack *) iter.Next();
count++;
}
this->ReplaceProcessedTracks(bGoodResult);
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 EffectChangeSpeed::Process()
{
// Similar to EffectSoundTouch::Process()
// Iterate over each track.
// Track::All is needed because this effect needs to introduce
// silence in the sync-lock group tracks to keep sync
this->CopyInputTracks(Track::All); // Set up mOutputTracks.
bool bGoodResult = true;
TrackListIterator iter(mOutputTracks);
Track* t;
mCurTrackNum = 0;
mMaxNewLength = 0.0;
mFactor = 100.0 / (100.0 + mPercentChange);
t = iter.First();
while (t != NULL)
{
if (t->GetKind() == Track::Label) {
if (t->GetSelected() || t->IsSyncLockSelected())
{
if (!ProcessLabelTrack(t)) {
bGoodResult = false;
break;
}
}
}
else if (t->GetKind() == Track::Wave && t->GetSelected())
{
WaveTrack *pOutWaveTrack = (WaveTrack*)t;
//Get start and end times from track
mCurT0 = pOutWaveTrack->GetStartTime();
mCurT1 = pOutWaveTrack->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) {
//Transform the marker timepoints to samples
sampleCount start = pOutWaveTrack->TimeToLongSamples(mCurT0);
sampleCount end = pOutWaveTrack->TimeToLongSamples(mCurT1);
//ProcessOne() (implemented below) processes a single track
if (!ProcessOne(pOutWaveTrack, start, end))
{
bGoodResult = false;
break;
}
}
mCurTrackNum++;
}
else if (t->IsSyncLockSelected())
{
t->SyncLockAdjust(mT1, mT0 + (mT1 - mT0) * mFactor);
}
//Iterate to the next track
t=iter.Next();
}
if (bGoodResult)
ReplaceProcessedTracks(bGoodResult);
mT1 = mT0 + mMaxNewLength; // Update selection.
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 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 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;
}