bool EffectTruncSilence::FindSilences
(RegionList &silences, Track *firstTrack, Track *lastTrack)
{
// Start with the whole selection silent
silences.push_back(Region(mT0, mT1));
// Remove non-silent regions in each track
SelectedTrackListOfKindIterator iter(Track::Wave, mTracks);
int whichTrack = 0;
bool lastSeen = false;
for (Track *t = iter.StartWith(firstTrack); !lastSeen && t; t = iter.Next())
{
lastSeen = (t == lastTrack);
WaveTrack *const wt = static_cast<WaveTrack *>(t);
// Smallest silent region to detect in frames
sampleCount minSilenceFrames =
sampleCount(std::max(mInitialAllowedSilence, DEF_MinTruncMs) * wt->GetRate());
//
// Scan the track for silences
//
RegionList trackSilences;
sampleCount index = wt->TimeToLongSamples(mT0);
sampleCount silentFrame = 0;
// Detect silences
bool cancelled = !(Analyze(silences, trackSilences, wt, &silentFrame, &index, whichTrack));
// Buffer has been freed, so we're OK to return if cancelled
if (cancelled)
{
ReplaceProcessedTracks(false);
return false;
}
if (silentFrame >= minSilenceFrames)
{
// Track ended in silence -- record region
trackSilences.push_back(Region(
wt->LongSamplesToTime(index - silentFrame),
wt->LongSamplesToTime(index)
));
}
// Intersect with the overall silent region list
Intersect(silences, trackSilences);
whichTrack++;
}
return true;
}
void TranscriptionToolBar::GetSamples(
const WaveTrack *t, sampleCount *s0, sampleCount *slen)
{
// GetSamples attempts to translate the start and end selection markers into sample indices
// These selection numbers are doubles.
AudacityProject *p = GetActiveProject();
if (!p) {
return;
}
//First, get the current selection. It is part of the mViewInfo, which is
//part of the project
const auto &selectedRegion = p->GetViewInfo().selectedRegion;
double start = selectedRegion.t0();
double end = selectedRegion.t1();
auto ss0 = sampleCount( (start - t->GetOffset()) * t->GetRate() );
auto ss1 = sampleCount( (end - t->GetOffset()) * t->GetRate() );
if (start < t->GetOffset()) {
ss0 = 0;
}
#if 0
//This adjusts the right samplecount to the maximum sample.
if (ss1 >= t->GetNumSamples()) {
ss1 = t->GetNumSamples();
}
#endif
if (ss1 < ss0) {
ss1 = ss0;
}
*s0 = ss0;
*slen = ss1 - ss0;
}
//! Compute the mean, maximum amplitude in DC, maximum amplitude in AC (mean substracted). NULL can be passed if value has to be ignored
void SignalDisplayData::channelAmplitude(unsigned channel, int *mean, int *maxAmplitudeDC, int *maxAmplitudeAC) const
{
Q_ASSERT(channel < channelCount);
const signed short *channelSamples = &data[channel * sampleCount()];
int minValue, maxValue;
int meanValue = 0;
minValue = maxValue = *channelSamples++;
for (unsigned i = 1; i < sampleCount(); i++)
{
int v = *channelSamples++;
minValue = std::min(v, minValue);
maxValue = std::max(v, maxValue);
meanValue += v;
}
meanValue /= sampleCount();
if (mean)
*mean = meanValue;
if (maxAmplitudeDC)
*maxAmplitudeDC = std::max(maxValue, abs(minValue));
if (maxAmplitudeAC)
*maxAmplitudeAC = std::max(maxValue - meanValue, abs(minValue - meanValue));
}
/**
* Construct a tile handler. This will determine a good chunk size to put
* into the output cube.
*
* @param dataFile The file with cube DN data in it
* @param virtualBandList The mapping from virtual band to physical band, see
* CubeIoHandler's description.
* @param labels The Pvl labels for the cube
* @param alreadyOnDisk True if the cube is allocated on the disk, false
* otherwise
*/
CubeTileHandler::CubeTileHandler(QFile * dataFile,
const QList<int> *virtualBandList, const Pvl &labels, bool alreadyOnDisk)
: CubeIoHandler(dataFile, virtualBandList, labels, alreadyOnDisk) {
const PvlObject &core = labels.findObject("IsisCube").findObject("Core");
if(core.hasKeyword("Format")) {
setChunkSizes(core["TileSamples"], core["TileLines"], 1);
}
else {
// up to 1MB chunks
int sampleChunkSize =
findGoodSize(512 * 4 / SizeOf(pixelType()), sampleCount());
int lineChunkSize =
findGoodSize(512 * 4 / SizeOf(pixelType()), lineCount());
setChunkSizes(sampleChunkSize, lineChunkSize, 1);
}
}
bool EffectTruncSilence::Analyze(RegionList& silenceList,
RegionList& trackSilences,
WaveTrack* wt,
sampleCount* silentFrame,
sampleCount* index,
int whichTrack,
double* inputLength /*= NULL*/,
double* minInputLength /*= NULL*/)
{
// Smallest silent region to detect in frames
sampleCount minSilenceFrames = sampleCount(std::max( mInitialAllowedSilence, DEF_MinTruncMs) * wt->GetRate());
double truncDbSilenceThreshold = Enums::Db2Signal[mTruncDbChoiceIndex];
sampleCount blockLen = wt->GetMaxBlockSize();
sampleCount start = wt->TimeToLongSamples(mT0);
sampleCount end = wt->TimeToLongSamples(mT1);
sampleCount outLength = 0;
double previewLength;
gPrefs->Read(wxT("/AudioIO/EffectsPreviewLen"), &previewLength, 6.0);
// Minimum required length in samples.
const sampleCount previewLen = previewLength * wt->GetRate();
// Keep position in overall silences list for optimization
RegionList::iterator rit(silenceList.begin());
// Allocate buffer
float *buffer = new float[blockLen];
// Loop through current track
while (*index < end) {
if (inputLength && ((outLength >= previewLen) || (*index - start > wt->TimeToLongSamples(*minInputLength)))) {
*inputLength = std::min<double>(*inputLength, *minInputLength);
if (outLength >= previewLen) {
*minInputLength = *inputLength;
}
return true;
}
if (!inputLength) {
// Show progress dialog, test for cancellation
bool cancelled = TotalProgress(
detectFrac * (whichTrack + (*index - start) / (double)(end - start)) /
(double)GetNumWaveTracks());
if (cancelled) {
delete [] buffer;
return false;
}
}
// Optimization: if not in a silent region skip ahead to the next one
double curTime = wt->LongSamplesToTime(*index);
for ( ; rit != silenceList.end(); ++rit) {
// Find the first silent region ending after current time
if (rit->end >= curTime) {
break;
}
}
if (rit == silenceList.end()) {
// No more regions -- no need to process the rest of the track
if (inputLength) {
// Add available samples up to previewLength.
sampleCount remainingTrackSamples = wt->TimeToLongSamples(wt->GetEndTime()) - *index;
sampleCount requiredTrackSamples = previewLen - outLength;
outLength += (remainingTrackSamples > requiredTrackSamples)? requiredTrackSamples : remainingTrackSamples;
}
break;
}
else if (rit->start > curTime) {
// End current silent region, skip ahead
if (*silentFrame >= minSilenceFrames) {
trackSilences.push_back(Region(
wt->LongSamplesToTime(*index - *silentFrame),
wt->LongSamplesToTime(*index)
));
}
*silentFrame = 0;
sampleCount newIndex = wt->TimeToLongSamples(rit->start);
if (inputLength) {
sampleCount requiredTrackSamples = previewLen - outLength;
// Add non-silent sample to outLength
outLength += ((newIndex - *index) > requiredTrackSamples)? requiredTrackSamples : newIndex - *index;
}
*index = newIndex;
}
// End of optimization
// Limit size of current block if we've reached the end
sampleCount count = blockLen;
if ((*index + count) > end) {
count = end - *index;
}
// Fill buffer
wt->Get((samplePtr)(buffer), floatSample, *index, count);
// Look for silenceList in current block
for (sampleCount i = 0; i < count; ++i) {
if (inputLength && ((outLength >= previewLen) || (outLength > wt->TimeToLongSamples(*minInputLength)))) {
*inputLength = wt->LongSamplesToTime(*index + i) - wt->LongSamplesToTime(start);
break;
}
if (fabs(buffer[i]) < truncDbSilenceThreshold) {
(*silentFrame)++;
}
else {
sampleCount allowed = 0;
if (*silentFrame >= minSilenceFrames) {
if (inputLength) {
switch (mActionIndex) {
case kTruncate:
outLength += wt->TimeToLongSamples(mTruncLongestAllowedSilence);
break;
case kCompress:
allowed = wt->TimeToLongSamples(mInitialAllowedSilence);
outLength += allowed +
(*silentFrame - allowed) * mSilenceCompressPercent / 100.0;
break;
// default: // Not currently used.
}
}
// Record the silent region
Region *r = new Region;
r->start = wt->LongSamplesToTime(*index + i - *silentFrame);
r->end = wt->LongSamplesToTime(*index + i);
trackSilences.push_back(Region(
wt->LongSamplesToTime(*index + i - *silentFrame),
wt->LongSamplesToTime(*index + i)
));
}
else if (inputLength) { // included as part of non-silence
outLength += *silentFrame;
}
*silentFrame = 0;
if (inputLength) {
++outLength; // Add non-silent sample to outLength
}
}
}
// Next block
*index += count;
}
delete [] buffer;
if (inputLength) {
*inputLength = std::min<double>(*inputLength, *minInputLength);
if (outLength >= previewLen) {
*minInputLength = *inputLength;
}
}
return true;
}
// Base code for adding, removing, moving and duplicating samples. Returns new number of samples on success, SAMPLEINDEX_INVALID otherwise.
// The new sample vector can contain SAMPLEINDEX_INVALID for adding new (empty) samples.
// newOrder indices are zero-based, i.e. newOrder[0] will define the contents of the first sample slot.
SAMPLEINDEX CModDoc::ReArrangeSamples(const std::vector<SAMPLEINDEX> &newOrder)
//-----------------------------------------------------------------------------
{
if(newOrder.size() > m_SndFile.GetModSpecifications().samplesMax)
{
return SAMPLEINDEX_INVALID;
}
CriticalSection cs;
const SAMPLEINDEX oldNumSamples = m_SndFile.GetNumSamples(), newNumSamples = static_cast<SAMPLEINDEX>(newOrder.size());
std::vector<int> sampleCount(oldNumSamples + 1, 0);
std::vector<ModSample> sampleHeaders(oldNumSamples + 1);
std::vector<SAMPLEINDEX> newIndex(oldNumSamples + 1, 0); // One of the new indexes for the old sample
std::vector<std::string> sampleNames(oldNumSamples + 1);
std::vector<mpt::PathString> samplePaths(oldNumSamples + 1);
for(SAMPLEINDEX i = 0; i < newNumSamples; i++)
{
const SAMPLEINDEX origSlot = newOrder[i];
if(origSlot > 0 && origSlot <= oldNumSamples)
{
sampleCount[origSlot]++;
sampleHeaders[origSlot] = m_SndFile.GetSample(origSlot);
if(!newIndex[origSlot]) newIndex[origSlot] = i + 1;
}
}
// First, delete all samples that will be removed anyway.
for(SAMPLEINDEX i = 1; i < sampleCount.size(); i++)
{
if(sampleCount[i] == 0)
{
m_SndFile.DestroySample(i);
GetSampleUndo().ClearUndo(i);
}
sampleNames[i] = m_SndFile.m_szNames[i];
samplePaths[i] = m_SndFile.GetSamplePath(i);
}
// Remove sample data references from now unused slots.
for(SAMPLEINDEX i = newNumSamples + 1; i <= oldNumSamples; i++)
{
m_SndFile.GetSample(i).pSample = nullptr;
m_SndFile.GetSample(i).nLength = 0;
strcpy(m_SndFile.m_szNames[i], "");
}
// Now, create new sample list.
m_SndFile.m_nSamples = std::max(m_SndFile.m_nSamples, newNumSamples); // Avoid assertions when using GetSample()...
for(SAMPLEINDEX i = 0; i < newNumSamples; i++)
{
const SAMPLEINDEX origSlot = newOrder[i];
ModSample &target = m_SndFile.GetSample(i + 1);
if(origSlot > 0 && origSlot <= oldNumSamples)
{
// Copy an original sample.
target = sampleHeaders[origSlot];
if(--sampleCount[origSlot] > 0 && sampleHeaders[origSlot].pSample != nullptr)
{
// This sample slot is referenced multiple times, so we have to copy the actual sample.
target.pSample = ModSample::AllocateSample(target.nLength, target.GetBytesPerSample());
if(target.pSample != nullptr)
{
memcpy(target.pSample, sampleHeaders[origSlot].pSample, target.GetSampleSizeInBytes());
target.PrecomputeLoops(m_SndFile, false);
} else
{
Reporting::Error("Cannot duplicate sample - out of memory!");
}
}
strcpy(m_SndFile.m_szNames[i + 1], sampleNames[origSlot].c_str());
m_SndFile.SetSamplePath(i + 1, samplePaths[origSlot]);
} else
{
// Invalid sample reference.
target.Initialize(m_SndFile.GetType());
target.pSample = nullptr;
strcpy(m_SndFile.m_szNames[i + 1], "");
m_SndFile.ResetSamplePath(i + 1);
}
}
GetSampleUndo().RearrangeSamples(newIndex);
for(CHANNELINDEX c = 0; c < CountOf(m_SndFile.m_PlayState.Chn); c++)
{
ModChannel &chn = m_SndFile.m_PlayState.Chn[c];
for(SAMPLEINDEX i = 1; i <= oldNumSamples; i++)
{
if(chn.pModSample == &m_SndFile.GetSample(i))
{
chn.pModSample = &m_SndFile.GetSample(newIndex[i]);
if(i == 0 || i > newNumSamples)
{
chn.Reset(ModChannel::resetTotal, m_SndFile, c);
}
break;
}
}
}
m_SndFile.m_nSamples = newNumSamples;
if(m_SndFile.GetNumInstruments())
{
// Instrument mode: Update sample maps.
for(INSTRUMENTINDEX i = 0; i <= m_SndFile.GetNumInstruments(); i++)
{
ModInstrument *ins = m_SndFile.Instruments[i];
if(ins == nullptr)
{
continue;
}
GetInstrumentUndo().RearrangeSamples(i, newIndex);
for(auto &sample : ins->Keyboard)
{
if(sample < newIndex.size())
sample = newIndex[sample];
else
sample = 0;
}
}
} else
{
PrepareUndoForAllPatterns(false, "Rearrange Samples");
std::vector<ModCommand::INSTR> indices(newIndex.size(), 0);
for(size_t i = 0; i < newIndex.size(); i++)
{
indices[i] = newIndex[i];
}
m_SndFile.Patterns.ForEachModCommand(RewriteInstrumentReferencesInPatterns(indices));
}
return GetNumSamples();
}
sampleCount Mixer::MixVariableRates(int *channelFlags, WaveTrackCache &cache,
sampleCount *pos, float *queue,
int *queueStart, int *queueLen,
Resample * pResample)
{
const WaveTrack *const track = cache.GetTrack();
const double trackRate = track->GetRate();
const double initialWarp = mRate / mSpeed / trackRate;
const double tstep = 1.0 / trackRate;
int sampleSize = SAMPLE_SIZE(floatSample);
sampleCount out = 0;
/* time is floating point. Sample rate is integer. The number of samples
* has to be integer, but the multiplication gives a float result, which we
* round to get an integer result. TODO: is this always right or can it be
* off by one sometimes? Can we not get this information directly from the
* clip (which must know) rather than convert the time?
*
* LLL: Not at this time. While WaveClips provide methods to retrieve the
* start and end sample, they do the same float->sampleCount conversion
* to calculate the position.
*/
// Find the last sample
double endTime = track->GetEndTime();
double startTime = track->GetStartTime();
const bool backwards = (mT1 < mT0);
const double tEnd = backwards
? std::max(startTime, mT1)
: std::min(endTime, mT1);
const sampleCount endPos = track->TimeToLongSamples(tEnd);
// Find the time corresponding to the start of the queue, for use with time track
double t = (*pos + (backwards ? *queueLen : - *queueLen)) / trackRate;
while (out < mMaxOut) {
if (*queueLen < mProcessLen) {
// Shift pending portion to start of the buffer
memmove(queue, &queue[*queueStart], (*queueLen) * sampleSize);
*queueStart = 0;
int getLen =
std::min((backwards ? *pos - endPos : endPos - *pos),
sampleCount(mQueueMaxLen - *queueLen));
// Nothing to do if past end of play interval
if (getLen > 0) {
if (backwards) {
auto results = cache.Get(floatSample, *pos - (getLen - 1), getLen);
memcpy(&queue[*queueLen], results, sizeof(float) * getLen);
track->GetEnvelopeValues(mEnvValues,
getLen,
(*pos - (getLen- 1)) / trackRate,
tstep);
*pos -= getLen;
}
else {
auto results = cache.Get(floatSample, *pos, getLen);
memcpy(&queue[*queueLen], results, sizeof(float) * getLen);
track->GetEnvelopeValues(mEnvValues,
getLen,
(*pos) / trackRate,
tstep);
*pos += getLen;
}
for (int i = 0; i < getLen; i++) {
queue[(*queueLen) + i] *= mEnvValues[i];
}
if (backwards)
ReverseSamples((samplePtr)&queue[0], floatSample,
*queueLen, getLen);
*queueLen += getLen;
}
}
sampleCount thisProcessLen = mProcessLen;
bool last = (*queueLen < mProcessLen);
if (last) {
thisProcessLen = *queueLen;
}
double factor = initialWarp;
if (mTimeTrack)
{
//TODO-MB: The end time is wrong when the resampler doesn't use all input samples,
// as a result of this the warp factor may be slightly wrong, so AudioIO will stop too soon
// or too late (resulting in missing sound or inserted silence). This can't be fixed
// without changing the way the resampler works, because the number of input samples that will be used
// is unpredictable. Maybe it can be compensated later though.
if (backwards)
factor *= mTimeTrack->ComputeWarpFactor
(t - (double)thisProcessLen / trackRate + tstep, t + tstep);
else
factor *= mTimeTrack->ComputeWarpFactor
(t, t + (double)thisProcessLen / trackRate);
}
int input_used;
int outgen = pResample->Process(factor,
&queue[*queueStart],
thisProcessLen,
last,
&input_used,
&mFloatBuffer[out],
mMaxOut - out);
if (outgen < 0) {
return 0;
}
*queueStart += input_used;
*queueLen -= input_used;
out += outgen;
t += ((backwards ? -input_used : input_used) / trackRate);
if (last) {
break;
}
}
for (int c = 0; c < mNumChannels; c++) {
if (mApplyTrackGains) {
mGains[c] = track->GetChannelGain(c);
}
else {
mGains[c] = 1.0;
}
}
MixBuffers(mNumChannels,
channelFlags,
mGains,
(samplePtr)mFloatBuffer,
mTemp,
out,
mInterleaved);
return out;
}
double Curve::minX() const {
if (hasBars() && sampleCount() > 0) {
return MinX - (MaxX - MinX)/(2*(sampleCount()-1));
}
return MinX;
}
void reconfigure()
{
if(!self.isReady() || size == Size()) return;
LOGDEV_GL_VERBOSE("Reconfiguring framebuffer: %s ms:%i")
<< size.asText() << sampleCount();
// Configure textures for the framebuffer.
color.setUndefinedImage(size, colorFormat);
color.setWrap(gl::ClampToEdge, gl::ClampToEdge);
color.setFilter(gl::Nearest, gl::Linear, gl::MipNone);
DENG2_ASSERT(color.isReady());
depthStencil.setDepthStencilContent(size);
depthStencil.setWrap(gl::ClampToEdge, gl::ClampToEdge);
depthStencil.setFilter(gl::Nearest, gl::Nearest, gl::MipNone);
DENG2_ASSERT(depthStencil.isReady());
try
{
// We'd like to use texture attachments for both color and depth/stencil.
target.configure(&color, &depthStencil);
}
catch(GLTarget::ConfigError const &er)
{
// Alternatively try without depth/stencil texture (some renderer features
// will not be available!).
LOG_GL_WARNING("Texture-based framebuffer failed: %s\n"
"Trying fallback without depth/stencil texture")
<< er.asText();
target.configure(GLTarget::Color, color, GLTarget::DepthStencil);
}
target.clear(GLTarget::ColorDepthStencil);
if(isMultisampled())
{
try
{
// Set up the multisampled target with suitable renderbuffers.
multisampleTarget.configure(size, GLTarget::ColorDepthStencil, sampleCount());
multisampleTarget.clear(GLTarget::ColorDepthStencil);
// Actual drawing occurs in the multisampled target that is then
// blitted to the main target.
target.setProxy(&multisampleTarget);
}
catch(GLTarget::ConfigError const &er)
{
LOG_GL_WARNING("Multisampling not supported: %s") << er.asText();
_samples = 1;
goto noMultisampling;
}
}
else
{
noMultisampling:
multisampleTarget.configure();
}
}
bool EffectTruncSilence::Process()
{
// Typical fraction of total time taken by detection (better to guess low)
const double detectFrac = .4;
// Copy tracks
this->CopyInputTracks(Track::All);
// Lower bound on the amount of silence to find at a time -- this avoids
// detecting silence repeatedly in low-frequency sounds.
const double minTruncMs = 0.001;
double truncDbSilenceThreshold = Enums::Db2Signal[mTruncDbChoiceIndex];
// Master list of silent regions; it is responsible for deleting them.
// This list should always be kept in order.
RegionList silences;
silences.DeleteContents(true);
// Start with the whole selection silent
Region *sel = new Region;
sel->start = mT0;
sel->end = mT1;
silences.push_back(sel);
// Remove non-silent regions in each track
SelectedTrackListOfKindIterator iter(Track::Wave, mTracks);
int whichTrack = 0;
for (Track *t = iter.First(); t; t = iter.Next())
{
WaveTrack *wt = (WaveTrack *)t;
// Smallest silent region to detect in frames
sampleCount minSilenceFrames =
sampleCount(wxMax( mInitialAllowedSilence, minTruncMs) *
wt->GetRate());
//
// Scan the track for silences
//
RegionList trackSilences;
trackSilences.DeleteContents(true);
sampleCount blockLen = wt->GetMaxBlockSize();
sampleCount start = wt->TimeToLongSamples(mT0);
sampleCount end = wt->TimeToLongSamples(mT1);
// Allocate buffer
float *buffer = new float[blockLen];
sampleCount index = start;
sampleCount silentFrames = 0;
bool cancelled = false;
// Keep position in overall silences list for optimization
RegionList::iterator rit(silences.begin());
while (index < end) {
// Show progress dialog, test for cancellation
cancelled = TotalProgress(
detectFrac * (whichTrack + index / (double)end) /
(double)GetNumWaveTracks());
if (cancelled)
break;
//
// Optimization: if not in a silent region skip ahead to the next one
//
double curTime = wt->LongSamplesToTime(index);
for ( ; rit != silences.end(); ++rit)
{
// Find the first silent region ending after current time
if ((*rit)->end >= curTime)
break;
}
if (rit == silences.end()) {
// No more regions -- no need to process the rest of the track
break;
}
else if ((*rit)->start > curTime) {
// End current silent region, skip ahead
if (silentFrames >= minSilenceFrames) {
Region *r = new Region;
r->start = wt->LongSamplesToTime(index - silentFrames);
r->end = wt->LongSamplesToTime(index);
trackSilences.push_back(r);
}
silentFrames = 0;
index = wt->TimeToLongSamples((*rit)->start);
}
//
// End of optimization
//
// Limit size of current block if we've reached the end
sampleCount count = blockLen;
if ((index + count) > end) {
count = end - index;
}
// Fill buffer
wt->Get((samplePtr)(buffer), floatSample, index, count);
// Look for silences in current block
for (sampleCount i = 0; i < count; ++i) {
if (fabs(buffer[i]) < truncDbSilenceThreshold) {
++silentFrames;
}
else {
if (silentFrames >= minSilenceFrames)
{
// Record the silent region
Region *r = new Region;
r->start = wt->LongSamplesToTime(index + i - silentFrames);
r->end = wt->LongSamplesToTime(index + i);
trackSilences.push_back(r);
}
silentFrames = 0;
}
}
// Next block
index += count;
}
delete [] buffer;
// Buffer has been freed, so we're OK to return if cancelled
if (cancelled)
{
ReplaceProcessedTracks(false);
return false;
}
if (silentFrames >= minSilenceFrames)
{
// Track ended in silence -- record region
Region *r = new Region;
r->start = wt->LongSamplesToTime(index - silentFrames);
r->end = wt->LongSamplesToTime(index);
trackSilences.push_back(r);
}
// Intersect with the overall silent region list
Intersect(silences, trackSilences);
whichTrack++;
}
//
// Now remove the silent regions from all selected / sync-lock selected tracks.
//
// Loop over detected regions in reverse (so cuts don't change time values
// down the line)
int whichReg = 0;
RegionList::reverse_iterator rit;
double totalCutLen = 0.0; // For cutting selection at the end
for (rit = silences.rbegin(); rit != silences.rend(); ++rit) {
Region *r = *rit;
// Progress dialog and cancellation. Do additional cleanup before return.
if (TotalProgress(detectFrac + (1 - detectFrac) * whichReg / (double)silences.size()))
{
ReplaceProcessedTracks(false);
return false;
}
// Intersection may create regions smaller than allowed; ignore them.
// Allow one nanosecond extra for consistent results with exact milliseconds of allowed silence.
if ((r->end - r->start) < (mInitialAllowedSilence - 0.000000001))
continue;
// Find new silence length as requested
double inLength = r->end - r->start;
double outLength;
switch (mProcessIndex) {
case 0:
outLength = wxMin(mTruncLongestAllowedSilence, inLength);
break;
case 1:
outLength = mInitialAllowedSilence +
(inLength - mInitialAllowedSilence) * mSilenceCompressPercent / 100.0;
break;
default: // Not currently used.
outLength = wxMin(mInitialAllowedSilence +
(inLength - mInitialAllowedSilence) * mSilenceCompressPercent / 100.0,
mTruncLongestAllowedSilence);
}
double cutLen = inLength - outLength;
totalCutLen += cutLen;
TrackListIterator iterOut(mOutputTracks);
for (Track *t = iterOut.First(); t; t = iterOut.Next())
{
// Don't waste time past the end of a track
if (t->GetEndTime() < r->start)
continue;
if (t->GetKind() == Track::Wave && (
t->GetSelected() || t->IsSyncLockSelected()))
{
// In WaveTracks, clear with a cross-fade
WaveTrack *wt = (WaveTrack *)t;
sampleCount blendFrames = mBlendFrameCount;
double cutStart = (r->start + r->end - cutLen) / 2;
double cutEnd = cutStart + cutLen;
// Round start/end times to frame boundaries
cutStart = wt->LongSamplesToTime(wt->TimeToLongSamples(cutStart));
cutEnd = wt->LongSamplesToTime(wt->TimeToLongSamples(cutEnd));
// Make sure the cross-fade does not affect non-silent frames
if (wt->LongSamplesToTime(blendFrames) > inLength) {
blendFrames = wt->TimeToLongSamples(inLength);
}
// Perform cross-fade in memory
float *buf1 = new float[blendFrames];
float *buf2 = new float[blendFrames];
sampleCount t1 = wt->TimeToLongSamples(cutStart) - blendFrames / 2;
sampleCount t2 = wt->TimeToLongSamples(cutEnd) - blendFrames / 2;
wt->Get((samplePtr)buf1, floatSample, t1, blendFrames);
wt->Get((samplePtr)buf2, floatSample, t2, blendFrames);
for (sampleCount i = 0; i < blendFrames; ++i) {
buf1[i] = ((blendFrames-i) * buf1[i] + i * buf2[i]) /
(double)blendFrames;
}
// Perform the cut
wt->Clear(cutStart, cutEnd);
// Write cross-faded data
wt->Set((samplePtr)buf1, floatSample, t1, blendFrames);
delete [] buf1;
delete [] buf2;
}
else if (t->GetSelected() || t->IsSyncLockSelected())
{
// Non-wave tracks: just do a sync-lock adjust
double cutStart = (r->start + r->end - cutLen) / 2;
double cutEnd = cutStart + cutLen;
t->SyncLockAdjust(cutEnd, cutStart);
}
}
++whichReg;
}
mT1 -= totalCutLen;
ReplaceProcessedTracks(true);
return true;
}
//! Return a duration in ms for a given sample value. Floating point version
double SignalDisplayData::sampleToTime(double sample) const
{
return (sample * (double)duration) / (double)sampleCount();
}
//! Return a duration in ms for a given sample value. Integer version
unsigned SignalDisplayData::sampleToTime(unsigned sample) const
{
return (sample * duration) / sampleCount();
}
//! Return a duration in sample for the given time in ms. Floating point version
double SignalDisplayData::timeToSample(double time) const
{
return (time * (double)sampleCount()) / (double)duration;
}
//! Return a duration in sample for the given time in ms. Integer version
unsigned SignalDisplayData::timeToSample(unsigned time) const
{
return (time * sampleCount()) / duration;
}
