bool EffectReverse::ProcessOneClip(int count, WaveTrack *track,
sampleCount start, sampleCount len,
sampleCount originalStart, sampleCount originalEnd)
{
bool rc = true;
// keep track of two blocks whose data we will swap
auto first = start;
auto blockSize = track->GetMaxBlockSize();
float tmp;
Floats buffer1{ blockSize };
Floats buffer2{ blockSize };
auto originalLen = originalEnd - originalStart;
while (len > 1) {
auto block =
limitSampleBufferSize( track->GetBestBlockSize(first), len / 2 );
auto second = first + (len - block);
track->Get((samplePtr)buffer1.get(), floatSample, first, block);
track->Get((samplePtr)buffer2.get(), floatSample, second, block);
for (decltype(block) i = 0; i < block; i++) {
tmp = buffer1[i];
buffer1[i] = buffer2[block-i-1];
buffer2[block-i-1] = tmp;
}
track->Set((samplePtr)buffer1.get(), floatSample, first, block);
track->Set((samplePtr)buffer2.get(), floatSample, second, block);
len -= 2 * block;
first += block;
if( TrackProgress(count, 2 * ( first - originalStart ).as_double() /
originalLen.as_double() ) ) {
rc = false;
break;
}
}
return rc;
}
size_t EffectDtmf::ProcessBlock(float **WXUNUSED(inbuf), float **outbuf, size_t size)
{
float *buffer = outbuf[0];
decltype(size) processed = 0;
// for the whole dtmf sequence, we will be generating either tone or silence
// according to a bool value, and this might be done in small chunks of size
// 'block', as a single tone might sometimes be larger than the block
// tone and silence generally have different duration, thus two generation blocks
//
// Note: to overcome a 'clicking' noise introduced by the abrupt transition from/to
// silence, I added a fade in/out of 1/250th of a second (4ms). This can still be
// tweaked but gives excellent results at 44.1kHz: I haven't tried other freqs.
// A problem might be if the tone duration is very short (<10ms)... (?)
//
// One more problem is to deal with the approximations done when calculating the duration
// of both tone and silence: in some cases the final sum might not be same as the initial
// duration. So, to overcome this, we had a redistribution block up, and now we will spread
// the remaining samples in every bin in order to achieve the full duration: test case was
// to generate an 11 tone DTMF sequence, in 4 seconds, and with DutyCycle=75%: after generation
// you ended up with 3.999s or in other units: 3 seconds and 44097 samples.
//
while (size)
{
if (numRemaining == 0)
{
isTone = !isTone;
if (isTone)
{
curSeqPos++;
numRemaining = numSamplesTone;
curTonePos = 0;
}
else
{
numRemaining = numSamplesSilence;
}
// the statement takes care of extracting one sample from the diff bin and
// adding it into the current block until depletion
numRemaining += (diff-- > 0 ? 1 : 0);
}
const auto len = limitSampleBufferSize( size, numRemaining );
if (isTone)
{
// generate the tone and append
MakeDtmfTone(buffer, len, mSampleRate, dtmfSequence[curSeqPos], curTonePos, numSamplesTone, dtmfAmplitude);
curTonePos += len;
}
else
{
memset(buffer, 0, sizeof(float) * len);
}
numRemaining -= len;
buffer += len;
size -= len;
processed += len;
}
return processed;
}
// ProcessOne() takes a track, transforms it to bunch of buffer-blocks,
// and calls libsamplerate code on these blocks.
bool EffectChangeSpeed::ProcessOne(WaveTrack * track,
sampleCount start, sampleCount end)
{
if (track == NULL)
return false;
// initialization, per examples of Mixer::Mixer and
// EffectSoundTouch::ProcessOne
auto outputTrack = mFactory->NewWaveTrack(track->GetSampleFormat(),
track->GetRate());
//Get the length of the selection (as double). len is
//used simple to calculate a progress meter, so it is easier
//to make it a double now than it is to do it later
auto len = (end - start).as_double();
// Initiate processing buffers, most likely shorter than
// the length of the selection being processed.
auto inBufferSize = track->GetMaxBlockSize();
Floats inBuffer{ inBufferSize };
// mFactor is at most 100-fold so this shouldn't overflow size_t
auto outBufferSize = size_t( mFactor * inBufferSize + 10 );
Floats outBuffer{ outBufferSize };
// Set up the resampling stuff for this track.
Resample resample(true, mFactor, mFactor); // constant rate resampling
//Go through the track one buffer at a time. samplePos counts which
//sample the current buffer starts at.
bool bResult = true;
auto samplePos = start;
while (samplePos < end) {
//Get a blockSize of samples (smaller than the size of the buffer)
auto blockSize = limitSampleBufferSize(
track->GetBestBlockSize(samplePos),
end - samplePos
);
//Get the samples from the track and put them in the buffer
track->Get((samplePtr) inBuffer.get(), floatSample, samplePos, blockSize);
const auto results = resample.Process(mFactor,
inBuffer.get(),
blockSize,
((samplePos + blockSize) >= end),
outBuffer.get(),
outBufferSize);
const auto outgen = results.second;
if (outgen > 0)
outputTrack->Append((samplePtr)outBuffer.get(), floatSample,
outgen);
// Increment samplePos
samplePos += results.first;
// Update the Progress meter
if (TrackProgress(mCurTrackNum, (samplePos - start).as_double() / len)) {
bResult = false;
break;
}
}
// Flush the output WaveTrack (since it's buffered, too)
outputTrack->Flush();
// Take the output track and insert it in place of the original
// sample data
double newLength = outputTrack->GetEndTime();
if (bResult)
{
LinearTimeWarper warper { mCurT0, mCurT0, mCurT1, mCurT0 + newLength };
bResult = track->ClearAndPaste(
mCurT0, mCurT1, outputTrack.get(), true, false, &warper);
}
if (newLength > mMaxNewLength)
mMaxNewLength = newLength;
return bResult;
}
bool EffectTruncSilence::Analyze(RegionList& silenceList,
RegionList& trackSilences,
const WaveTrack *wt,
sampleCount* silentFrame,
sampleCount* index,
int whichTrack,
double* inputLength /*= NULL*/,
double* minInputLength /*= NULL*/)
{
// Smallest silent region to detect in frames
auto minSilenceFrames = sampleCount(std::max( mInitialAllowedSilence, DEF_MinTruncMs) * wt->GetRate());
double truncDbSilenceThreshold = Enums::Db2Signal[mTruncDbChoiceIndex];
auto blockLen = wt->GetMaxBlockSize();
auto start = wt->TimeToLongSamples(mT0);
auto 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
Floats buffer{ 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).as_double() /
(end - start).as_double()) /
(double)GetNumWaveTracks());
if (cancelled)
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.
auto remainingTrackSamples = wt->TimeToLongSamples(wt->GetEndTime()) - *index;
auto 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;
auto newIndex = wt->TimeToLongSamples(rit->start);
if (inputLength) {
auto 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
auto count = limitSampleBufferSize( blockLen, end - *index );
// Fill buffer
wt->Get((samplePtr)(buffer.get()), floatSample, *index, count);
// Look for silenceList in current block
for (decltype(count) 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 += sampleCount(
allowed.as_double() +
(*silentFrame - allowed).as_double()
* mSilenceCompressPercent / 100.0
);
break;
// default: // Not currently used.
}
}
// Record the silent region
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;
}
if (inputLength) {
*inputLength = std::min<double>(*inputLength, *minInputLength);
if (outLength >= previewLen) {
*minInputLength = *inputLength;
}
}
return true;
}
//ProcessOne() takes a track, transforms it to bunch of buffer-blocks,
//and executes TwoBufferProcessPass1 or TwoBufferProcessPass2 on these blocks
bool EffectTwoPassSimpleMono::ProcessOne(WaveTrack * track,
sampleCount start, sampleCount end)
{
bool ret;
float *tmpfloat;
//Get the length of the buffer (as double). len is
//used simple to calculate a progress meter, so it is easier
//to make it a double now than it is to do it later
auto len = (end - start).as_double();
auto maxblock = track->GetMaxBlockSize();
//Initiate a processing buffer. This buffer will (most likely)
//be shorter than the length of the track being processed.
float *buffer1 = new float[maxblock];
float *buffer2 = new float[maxblock];
auto samples1 = limitSampleBufferSize(
std::min( maxblock, track->GetBestBlockSize(start) ), end - start );
//Get the samples from the track and put them in the buffer
track->Get((samplePtr) buffer1, floatSample, start, samples1);
// Process the first buffer with a NULL previous buffer
if (mPass == 0)
ret = TwoBufferProcessPass1(NULL, 0, buffer1, samples1);
else
ret = TwoBufferProcessPass2(NULL, 0, buffer1, samples1);
if (!ret) {
delete[]buffer1;
delete[]buffer2;
//Return false because the effect failed.
return false;
}
//Go through the track one buffer at a time. s counts which
//sample the current buffer starts at.
auto s = start + samples1;
while (s < end) {
//Get a block of samples (smaller than the size of the buffer)
//Adjust the block size if it is the final block in the track
auto samples2 = limitSampleBufferSize(
std::min( track->GetBestBlockSize(s), maxblock ), end - s
);
//Get the samples from the track and put them in the buffer
track->Get((samplePtr) buffer2, floatSample, s, samples2);
//Process the buffer. If it fails, clean up and exit.
if (mPass == 0)
ret = TwoBufferProcessPass1(buffer1, samples1, buffer2, samples2);
else
ret = TwoBufferProcessPass2(buffer1, samples1, buffer2, samples2);
if (!ret) {
delete[]buffer1;
delete[]buffer2;
//Return false because the effect failed.
return false;
}
//Processing succeeded. copy the newly-changed samples back
//onto the track.
track->Set((samplePtr) buffer1, floatSample, s-samples1, samples1);
//Increment s one blockfull of samples
s += samples2;
//Update the Progress meter
if (mSecondPassDisabled)
ret = TotalProgress(
(mCurTrackNum + (s-start).as_double()/len) /
GetNumWaveTracks());
else
ret = TotalProgress(
(mCurTrackNum + (s-start).as_double()/len + GetNumWaveTracks()*mPass) /
(GetNumWaveTracks()*2));
if (ret) {
delete[]buffer1;
delete[]buffer2;
//Return false because the effect failed.
return false;
}
// Rotate the buffers
tmpfloat = buffer1;
buffer1 = buffer2;
buffer2 = tmpfloat;
std::swap(samples1, samples2);
}
// Send the last buffer with a NULL pointer for the current buffer
if (mPass == 0)
ret = TwoBufferProcessPass1(buffer1, samples1, NULL, 0);
else
ret = TwoBufferProcessPass2(buffer1, samples1, NULL, 0);
if (!ret) {
delete[]buffer1;
delete[]buffer2;
//Return false because the effect failed.
return false;
}
//Processing succeeded. copy the newly-changed samples back
//onto the track.
track->Set((samplePtr) buffer1, floatSample, s-samples1, samples1);
//Clean up the buffer
delete[]buffer1;
delete[]buffer2;
//Return true because the effect processing succeeded.
return true;
}
bool VampEffect::Process()
{
if (!mPlugin)
{
return false;
}
int count = 0;
bool multiple = false;
unsigned prevTrackChannels = 0;
if (GetNumWaveGroups() > 1)
{
// 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;
}
std::vector<std::shared_ptr<Effect::AddedAnalysisTrack>> addedTracks;
for (auto leader : inputTracks()->Leaders<const WaveTrack>())
{
auto channelGroup = TrackList::Channels(leader);
auto left = *channelGroup.first++;
sampleCount lstart, rstart = 0;
sampleCount len;
GetSamples(left, &lstart, &len);
unsigned channels = 1;
// channelGroup now contains all but the first channel
const WaveTrack *right =
channelGroup.size() ? *channelGroup.first++ : nullptr;
if (right)
{
channels = 2;
GetSamples(right, &rstart, &len);
}
// TODO: more-than-two-channels
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))
{
Effect::MessageBox(_("Sorry, Vamp Plug-in failed to initialize."));
return false;
}
}
const auto effectName = GetSymbol().Translation();
addedTracks.push_back(AddAnalysisTrack(
multiple
? wxString::Format( _("%s: %s"), left->GetName(), effectName )
: effectName
));
LabelTrack *ltrack = addedTracks.back()->get();
FloatBuffers data{ channels, block };
auto originalLen = len;
auto ls = lstart;
auto rs = rstart;
while (len != 0)
{
const auto request = limitSampleBufferSize( block, len );
if (left)
{
left->Get((samplePtr)data[0].get(), floatSample, ls, request);
}
if (right)
{
right->Get((samplePtr)data[1].get(), floatSample, rs, request);
}
if (request < block)
{
for (unsigned int c = 0; c < channels; ++c)
{
for (decltype(block) i = request; i < block; ++i)
{
data[c][i] = 0.f;
}
}
}
// UNSAFE_SAMPLE_COUNT_TRUNCATION
// Truncation in case of very long tracks!
Vamp::RealTime timestamp = Vamp::RealTime::frame2RealTime(
long( ls.as_long_long() ),
(int)(mRate + 0.5)
);
Vamp::Plugin::FeatureSet features = mPlugin->process(
reinterpret_cast< float** >( data.get() ), 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).as_double() /
originalLen.as_double() ))
{
return false;
}
}
else
{
if (TrackProgress(count,
(ls - lstart).as_double() /
originalLen.as_double() ))
{
return false;
}
}
}
Vamp::Plugin::FeatureSet features = mPlugin->getRemainingFeatures();
AddFeatures(ltrack, features);
prevTrackChannels = channels;
}
// All completed without cancellation, so commit the addition of tracks now
for (auto &addedTrack : addedTracks)
addedTrack->Commit();
return true;
}
// this currently does an exponential fade
bool EffectAutoDuck::ApplyDuckFade(int trackNum, WaveTrack* t,
double t0, double t1)
{
bool cancel = false;
auto start = t->TimeToLongSamples(t0);
auto end = t->TimeToLongSamples(t1);
Floats buf{ kBufSize };
auto pos = start;
auto fadeDownSamples = t->TimeToLongSamples(
mOuterFadeDownLen + mInnerFadeDownLen);
if (fadeDownSamples < 1)
fadeDownSamples = 1;
auto fadeUpSamples = t->TimeToLongSamples(
mOuterFadeUpLen + mInnerFadeUpLen);
if (fadeUpSamples < 1)
fadeUpSamples = 1;
float fadeDownStep = mDuckAmountDb / fadeDownSamples.as_double();
float fadeUpStep = mDuckAmountDb / fadeUpSamples.as_double();
while (pos < end)
{
const auto len = limitSampleBufferSize( kBufSize, end - pos );
t->Get((samplePtr)buf.get(), floatSample, pos, len);
for (auto i = pos; i < pos + len; i++)
{
float gainDown = fadeDownStep * (i - start).as_float();
float gainUp = fadeUpStep * (end - i).as_float();
float gain;
if (gainDown > gainUp)
gain = gainDown;
else
gain = gainUp;
if (gain < mDuckAmountDb)
gain = mDuckAmountDb;
// i - pos is bounded by len:
buf[ ( i - pos ).as_size_t() ] *= DB_TO_LINEAR(gain);
}
t->Set((samplePtr)buf.get(), floatSample, pos, len);
pos += len;
float curTime = t->LongSamplesToTime(pos);
float fractionFinished = (curTime - mT0) / (mT1 - mT0);
if (TotalProgress( (trackNum + 1 + fractionFinished) /
(GetNumWaveTracks() + 1) ))
{
cancel = true;
break;
}
}
return cancel;
}
bool EffectAutoDuck::Process()
{
if (GetNumWaveTracks() == 0 || !mControlTrack)
return false;
bool cancel = false;
auto start =
mControlTrack->TimeToLongSamples(mT0 + mOuterFadeDownLen);
auto end =
mControlTrack->TimeToLongSamples(mT1 - mOuterFadeUpLen);
if (end <= start)
return false;
// the minimum number of samples we have to wait until the maximum
// pause has been exceeded
double maxPause = mMaximumPause;
// We don't fade in until we have time enough to actually fade out again
if (maxPause < mOuterFadeDownLen + mOuterFadeUpLen)
maxPause = mOuterFadeDownLen + mOuterFadeUpLen;
auto minSamplesPause =
mControlTrack->TimeToLongSamples(maxPause);
double threshold = DB_TO_LINEAR(mThresholdDb);
// adjust the threshold so we can compare it to the rmsSum value
threshold = threshold * threshold * kRMSWindowSize;
int rmsPos = 0;
float rmsSum = 0;
// to make the progress bar appear more natural, we first look for all
// duck regions and apply them all at once afterwards
std::vector<AutoDuckRegion> regions;
bool inDuckRegion = false;
{
Floats rmsWindow{ kRMSWindowSize, true };
Floats buf{ kBufSize };
// initialize the following two variables to prevent compiler warning
double duckRegionStart = 0;
sampleCount curSamplesPause = 0;
auto pos = start;
while (pos < end)
{
const auto len = limitSampleBufferSize( kBufSize, end - pos );
mControlTrack->Get((samplePtr)buf.get(), floatSample, pos, len);
for (auto i = pos; i < pos + len; i++)
{
rmsSum -= rmsWindow[rmsPos];
// i - pos is bounded by len:
auto index = ( i - pos ).as_size_t();
rmsWindow[rmsPos] = buf[ index ] * buf[ index ];
rmsSum += rmsWindow[rmsPos];
rmsPos = (rmsPos + 1) % kRMSWindowSize;
bool thresholdExceeded = rmsSum > threshold;
if (thresholdExceeded)
{
// everytime the threshold is exceeded, reset our count for
// the number of pause samples
curSamplesPause = 0;
if (!inDuckRegion)
{
// the threshold has been exceeded for the first time, so
// let the duck region begin here
inDuckRegion = true;
duckRegionStart = mControlTrack->LongSamplesToTime(i);
}
}
if (!thresholdExceeded && inDuckRegion)
{
// the threshold has not been exceeded and we are in a duck
// region, but only fade in if the maximum pause has been
// exceeded
curSamplesPause += 1;
if (curSamplesPause >= minSamplesPause)
{
// do the actual duck fade and reset all values
double duckRegionEnd =
mControlTrack->LongSamplesToTime(i - curSamplesPause);
regions.push_back(AutoDuckRegion(
duckRegionStart - mOuterFadeDownLen,
duckRegionEnd + mOuterFadeUpLen));
inDuckRegion = false;
}
}
}
pos += len;
if (TotalProgress(
(pos - start).as_double() /
(end - start).as_double() /
(GetNumWaveTracks() + 1)
))
{
cancel = true;
break;
}
}
// apply last duck fade, if any
if (inDuckRegion)
{
double duckRegionEnd =
mControlTrack->LongSamplesToTime(end - curSamplesPause);
regions.push_back(AutoDuckRegion(
duckRegionStart - mOuterFadeDownLen,
duckRegionEnd + mOuterFadeUpLen));
}
}
if (!cancel)
{
CopyInputTracks(); // Set up mOutputTracks.
SelectedTrackListOfKindIterator iter(Track::Wave, mOutputTracks.get());
Track *iterTrack = iter.First();
int trackNum = 0;
while (iterTrack)
{
WaveTrack* t = (WaveTrack*)iterTrack;
for (size_t i = 0; i < regions.size(); i++)
{
const AutoDuckRegion& region = regions[i];
if (ApplyDuckFade(trackNum, t, region.t0, region.t1))
{
cancel = true;
break;
}
}
if (cancel)
break;
iterTrack = iter.Next();
trackNum++;
}
}
ReplaceProcessedTracks(!cancel);
return !cancel;
}