comparison audioio/AudioCallbackPlaySource.cpp @ 91:9fc4b256c283

* PortAudio driver: do not specify frames per buffer, let PA decide
* Remove old non-RubberBand time stretcher -- it doesn't work with varying
  buffer sizes such as the PA driver may now be using
* Rewrite getCurrentPlayingFrame for greater correctness when using long
  buffer sizes (interpolating according to audio stream timestamp)
* Several changes to make the timestretch management RT safe(r)

 #include "data/model/DenseTimeValueModel.h"
 #include "data/model/WaveFileModel.h"
 #include "data/model/SparseOneDimensionalModel.h"
 #include "plugin/RealTimePluginInstance.h"
 
-#ifdef HAVE_RUBBERBAND
+#include "AudioCallbackPlayTarget.h"
+
 #include <rubberband/RubberBandStretcher.h>
 using namespace RubberBand;
-#else
-#include "PhaseVocoderTimeStretcher.h"
-#endif
 
 #include <iostream>
 #include <cassert>
 
 //#define DEBUG_AUDIO_PLAY_SOURCE 1

     m_sourceChannelCount(0),
     m_blockSize(1024),
     m_sourceSampleRate(0),
     m_targetSampleRate(0),
     m_playLatency(0),
+    m_target(0),
+    m_lastRetrievalTimestamp(0.0),
+    m_lastRetrievedBlockSize(0),
     m_playing(false),
     m_exiting(false),
     m_lastModelEndFrame(0),
     m_outputLeft(0.0),
     m_outputRight(0.0),
     m_auditioningPlugin(0),
     m_auditioningPluginBypassed(false),
     m_timeStretcher(0),
+    m_stretchRatio(1.0),
+    m_stretcherInputCount(0),
+    m_stretcherInputs(0),
+    m_stretcherInputSizes(0),
     m_fillThread(0),
     m_converter(0),
     m_crapConverter(0),
     m_resampleQuality(Preferences::getInstance()->getResampleQuality())
 {

 
     delete m_writeBuffers;
 
     delete m_audioGenerator;
 
+    for (size_t i = 0; i < m_stretcherInputCount; ++i) {
+        delete[] m_stretcherInputs[i];
+    }
+    delete[] m_stretcherInputSizes;
+    delete[] m_stretcherInputs;
+
     m_bufferScavenger.scavenge(true);
     m_pluginScavenger.scavenge(true);
-#ifndef HAVE_RUBBERBAND
-    m_timeStretcherScavenger.scavenge(true);
-#endif
 }
 
 void
 AudioCallbackPlaySource::addModel(Model *model)
 {

     // The fill thread will automatically empty its buffers before
     // starting again if we have not so far been playing, but not if
     // we're just re-seeking.
 
     m_mutex.lock();
+    if (m_timeStretcher) {
+        m_timeStretcher->reset();
+    }
     if (m_playing) {
 	m_readBufferFill = m_writeBufferFill = startFrame;
 	if (m_readBuffers) {
 	    for (size_t c = 0; c < getTargetChannelCount(); ++c) {
 		RingBuffer<float> *rb = getReadRingBuffer(c);

     m_mutex.unlock();
 
     m_audioGenerator->reset();
 
     bool changed = !m_playing;
+    m_lastRetrievalTimestamp = 0;
     m_playing = true;
     m_condition.wakeAll();
     if (changed) emit playStatusChanged(m_playing);
 }
 

 AudioCallbackPlaySource::stop()
 {
     bool changed = m_playing;
     m_playing = false;
     m_condition.wakeAll();
+    m_lastRetrievalTimestamp = 0;
     if (changed) emit playStatusChanged(m_playing);
 }
 
 void
 AudioCallbackPlaySource::selectionChanged()

         emit audioOverloadPluginDisabled();
     }
 }
 
 void
-AudioCallbackPlaySource::setTargetBlockSize(size_t size)
-{
+AudioCallbackPlaySource::setTarget(AudioCallbackPlayTarget *target, size_t size)
+{
+    m_target = target;
 //    std::cout << "AudioCallbackPlaySource::setTargetBlockSize() -> " << size << std::endl;
     assert(size < m_ringBufferSize);
     m_blockSize = size;
 }
 

 }
 
 size_t
 AudioCallbackPlaySource::getCurrentPlayingFrame()
 {
+    // This method attempts to estimate which audio sample frame is
+    // "currently coming through the speakers".
+
     bool resample = false;
-    double ratio = 1.0;
-
-    if (getSourceSampleRate() != getTargetSampleRate()) {
-	resample = true;
-	ratio = double(getSourceSampleRate()) / double(getTargetSampleRate());
-    }
-
-    size_t readSpace = 0;
+    double resampleRatio = 1.0;
+
+    // We resample when filling the ring buffer, and time-stretch when
+    // draining it.  The buffer contains data at the "target rate" and
+    // the latency provided by the target is also at the target rate.
+    // Because of the multiple rates involved, we do the actual
+    // calculation using RealTime instead.
+
+    size_t sourceRate = getSourceSampleRate();
+    size_t targetRate = getTargetSampleRate();
+
+    if (sourceRate == 0 || targetRate == 0) return 0;
+
+    size_t inbuffer = 0; // at target rate
+
     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
 	RingBuffer<float> *rb = getReadRingBuffer(c);
 	if (rb) {
-	    size_t spaceHere = rb->getReadSpace();
-	    if (c == 0 || spaceHere < readSpace) readSpace = spaceHere;
-	}
-    }
-
-    if (resample) {
-	readSpace = size_t(readSpace * ratio + 0.1);
-    }
-
-    size_t latency = m_playLatency;
-    if (resample) latency = size_t(m_playLatency * ratio + 0.1);
-
-#ifdef HAVE_RUBBERBAND
+	    size_t here = rb->getReadSpace();
+	    if (c == 0 || here < inbuffer) inbuffer = here;
+	}
+    }
+
+    size_t readBufferFill = m_readBufferFill;
+    size_t lastRetrievedBlockSize = m_lastRetrievedBlockSize;
+    double lastRetrievalTimestamp = m_lastRetrievalTimestamp;
+    double currentTime = 0.0;
+    if (m_target) currentTime = m_target->getCurrentTime();
+
+    RealTime inbuffer_t = RealTime::frame2RealTime(inbuffer, targetRate);
+
+    size_t latency = m_playLatency; // at target rate
+    RealTime latency_t = RealTime::frame2RealTime(latency, targetRate);
+
+    size_t stretchlat = 0;
+    double timeRatio = 1.0;
+
     if (m_timeStretcher) {
-        latency += m_timeStretcher->getLatency();
-    }
-#else
-    PhaseVocoderTimeStretcher *timeStretcher = m_timeStretcher;
-    if (timeStretcher) {
-	latency += timeStretcher->getProcessingLatency();
-    }
-#endif
-
-    latency += readSpace;
-    size_t bufferedFrame = m_readBufferFill;
+        stretchlat = m_timeStretcher->getLatency();
+        timeRatio = m_timeStretcher->getTimeRatio();
+    }
+
+    RealTime stretchlat_t = RealTime::frame2RealTime(stretchlat, targetRate);
+
+    // When the target has just requested a block from us, the last
+    // sample it obtained was our buffer fill frame count minus the
+    // amount of read space (converted back to source sample rate)
+    // remaining now.  That sample is not expected to be played until
+    // the target's play latency has elapsed.  By the time the
+    // following block is requested, that sample will be at the
+    // target's play latency minus the last requested block size away
+    // from being played.
+
+    RealTime sincerequest_t = RealTime::zeroTime;
+    RealTime lastretrieved_t = RealTime::zeroTime;
+
+    if (m_target && lastRetrievalTimestamp != 0.0) {
+
+        lastretrieved_t = RealTime::frame2RealTime
+            (lastRetrievedBlockSize, targetRate);
+
+        // calculate number of frames at target rate that have elapsed
+        // since the end of the last call to getSourceSamples
+
+        double elapsed = currentTime - lastRetrievalTimestamp;
+
+        if (elapsed > 0.0) {
+            sincerequest_t = RealTime::fromSeconds(elapsed);
+        }
+
+    } else {
+
+        lastretrieved_t = RealTime::frame2RealTime
+            (getTargetBlockSize(), targetRate);
+    }
+
+    RealTime bufferedto_t = RealTime::frame2RealTime(readBufferFill, sourceRate);
+
+    if (timeRatio != 1.0) {
+        lastretrieved_t = lastretrieved_t / timeRatio;
+        sincerequest_t = sincerequest_t / timeRatio;
+    }
 
     bool looping = m_viewManager->getPlayLoopMode();
     bool constrained = (m_viewManager->getPlaySelectionMode() &&
 			!m_viewManager->getSelections().empty());
 
-    size_t framePlaying = bufferedFrame;
-
-    if (looping && !constrained) {
-	while (framePlaying < latency) framePlaying += m_lastModelEndFrame;
-    }
-
-    if (framePlaying > latency) framePlaying -= latency;
-    else framePlaying = 0;
-
-//    std::cerr << "framePlaying = " << framePlaying << " -> reference ";
-
-    framePlaying = m_viewManager->alignPlaybackFrameToReference(framePlaying);
-
-//    std::cerr << framePlaying << std::endl;
-
-    if (!constrained) {
-	if (!looping && framePlaying > m_lastModelEndFrame) {
-	    framePlaying = m_lastModelEndFrame;
-	    stop();
-	}
-	return framePlaying;
-    }
-
-    bufferedFrame = m_viewManager->alignPlaybackFrameToReference(bufferedFrame);
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+    std::cerr << "\nbuffered to: " << bufferedto_t << ", in buffer: " << inbuffer_t << ", time ratio " << timeRatio << "\n  stretcher latency: " << stretchlat_t << ", device latency: " << latency_t << "\n  since request: " << sincerequest_t << ", last retrieved: " << lastretrieved_t << std::endl;
+#endif
+
+    RealTime end = RealTime::frame2RealTime(m_lastModelEndFrame, sourceRate);
 
     MultiSelection::SelectionList selections = m_viewManager->getSelections();
     MultiSelection::SelectionList::const_iterator i;
 
-//    i = selections.begin();
-//    size_t rangeStart = i->getStartFrame();
-
-    i = selections.end();
-    --i;
-    size_t rangeEnd = i->getEndFrame();
-
-    for (i = selections.begin(); i != selections.end(); ++i) {
-	if (i->contains(bufferedFrame)) break;
-    }
-
-    size_t f = bufferedFrame;
-
-//    std::cout << "getCurrentPlayingFrame: f=" << f << ", latency=" << latency << ", rangeEnd=" << rangeEnd << std::endl;
-
-    if (i == selections.end()) {
-	--i;
-	if (i->getEndFrame() + latency < f) {
-//    std::cout << "framePlaying = " << framePlaying << ", rangeEnd = " << rangeEnd << std::endl;
-
-	    if (!looping && (framePlaying > rangeEnd)) {
-//		std::cout << "STOPPING" << std::endl;
-		stop();
-		return rangeEnd;
-	    } else {
-		return framePlaying;
-	    }
-	} else {
-//	    std::cout << "latency <- " << latency << "-(" << f << "-" << i->getEndFrame() << ")" << std::endl;
-	    latency -= (f - i->getEndFrame());
-	    f = i->getEndFrame();
-	}
-    }
-
-//    std::cout << "i=(" << i->getStartFrame() << "," << i->getEndFrame() << ") f=" << f << ", latency=" << latency << std::endl;
-
-    while (latency > 0) {
-	size_t offset = f - i->getStartFrame();
-	if (offset >= latency) {
-	    if (f > latency) {
-		framePlaying = f - latency;
-	    } else {
-		framePlaying = 0;
-	    }
-	    break;
-	} else {
-	    if (i == selections.begin()) {
-		if (looping) {
-		    i = selections.end();
-		}
-	    }
-	    latency -= offset;
-	    --i;
-	    f = i->getEndFrame();
-	}
-    }
-
-    return framePlaying;
+    // these could be cached from one call to the next, if the
+    // selection has not changed... but some of the work would still
+    // need to be done because the playback model may have changed
+
+    std::vector<RealTime> rangeStarts;
+    std::vector<RealTime> rangeDurations;
+
+    int inRange = 0;
+    int index = 0;
+
+    if (constrained) {
+
+        for (i = selections.begin(); i != selections.end(); ++i) {
+            
+            RealTime start =
+                (RealTime::frame2RealTime
+                 (m_viewManager->alignReferenceToPlaybackFrame(i->getStartFrame()),
+                  sourceRate));
+            RealTime duration = 
+                (RealTime::frame2RealTime
+                 (m_viewManager->alignReferenceToPlaybackFrame(i->getEndFrame()) -
+                  m_viewManager->alignReferenceToPlaybackFrame(i->getStartFrame()),
+                  sourceRate));
+            
+            rangeStarts.push_back(start);
+            rangeDurations.push_back(duration);
+            
+            if (bufferedto_t >= start) {
+                inRange = index;
+            }
+            
+            ++index;
+        }
+    }
+
+    if (rangeStarts.empty()) {
+        rangeStarts.push_back(RealTime::zeroTime);
+        rangeDurations.push_back(end);
+    }
+
+    if (inRange >= rangeStarts.size()) inRange = rangeStarts.size()-1;
+
+    RealTime playing_t = bufferedto_t - rangeStarts[inRange];
+
+    playing_t = playing_t
+        - latency_t - stretchlat_t - lastretrieved_t - inbuffer_t
+        + sincerequest_t;
+ 
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+    std::cerr << "playing_t as offset into range " << inRange << " (with start = " << rangeStarts[inRange] << ") = " << playing_t << std::endl;
+#endif
+
+    while (playing_t < RealTime::zeroTime) {
+
+        if (inRange == 0) {
+            if (looping) {
+                inRange = rangeStarts.size() - 1;
+            } else {
+                break;
+            }
+        } else {
+            --inRange;
+        }
+
+        playing_t = playing_t + rangeDurations[inRange];
+    }
+
+    playing_t = playing_t + rangeStarts[inRange];
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+    std::cerr << "  playing time: " << playing_t << std::endl;
+#endif
+
+    if (!looping) {
+        if (inRange == rangeStarts.size()-1 &&
+            playing_t >= rangeStarts[inRange] + rangeDurations[inRange]) {
+            stop();
+        }
+    }
+
+    if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime;
+
+    size_t frame = RealTime::realTime2Frame(playing_t, sourceRate);
+    return m_viewManager->alignPlaybackFrameToReference(frame);
 }
 
 void
 AudioCallbackPlaySource::setOutputLevels(float left, float right)
 {

 {
     return m_sourceSampleRate;
 }
 
 void
-AudioCallbackPlaySource::setTimeStretch(float factor, bool sharpen, bool mono)
-{
-#ifdef HAVE_RUBBERBAND
-    if (m_timeStretcher) {
-        m_timeStretchRatioMutex.lock();
-        m_timeStretcher->setTimeRatio(factor);
-        m_timeStretchRatioMutex.unlock();
+AudioCallbackPlaySource::setTimeStretch(float factor)
+{
+    m_stretchRatio = factor;
+
+    if (m_timeStretcher || (factor == 1.f)) {
+        // stretch ratio will be set in next process call if appropriate
         return;
     } else {
+        m_stretcherInputCount = getTargetChannelCount();
         RubberBandStretcher *stretcher = new RubberBandStretcher
             (getTargetSampleRate(),
-             getTargetChannelCount(),
+             m_stretcherInputCount,
              RubberBandStretcher::OptionProcessRealTime,
              factor);
+        m_stretcherInputs = new float *[m_stretcherInputCount];
+        m_stretcherInputSizes = new size_t[m_stretcherInputCount];
+        for (size_t c = 0; c < m_stretcherInputCount; ++c) {
+            m_stretcherInputSizes[c] = 16384;
+            m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]];
+        }
         m_timeStretcher = stretcher;
         return;
     }
-#else
-    // Avoid locks -- create, assign, mark old one for scavenging
-    // later (as a call to getSourceSamples may still be using it)
-
-    PhaseVocoderTimeStretcher *existingStretcher = m_timeStretcher;
-
-    size_t channels = getTargetChannelCount();
-    if (mono) channels = 1;
-
-    if (existingStretcher &&
-        existingStretcher->getRatio() == factor &&
-        existingStretcher->getSharpening() == sharpen &&
-        existingStretcher->getChannelCount() == channels) {
-	return;
-    }
-
-    if (factor != 1) {
-
-        if (existingStretcher &&
-            existingStretcher->getSharpening() == sharpen &&
-            existingStretcher->getChannelCount() == channels) {
-            existingStretcher->setRatio(factor);
-            return;
-        }
-
-	PhaseVocoderTimeStretcher *newStretcher = new PhaseVocoderTimeStretcher
-	    (getTargetSampleRate(),
-             channels,
-             factor,
-             sharpen,
-             getTargetBlockSize());
-
-	m_timeStretcher = newStretcher;
-
-    } else {
-	m_timeStretcher = 0;
-    }
-
-    if (existingStretcher) {
-	m_timeStretcherScavenger.claim(existingStretcher);
-    }
-#endif
 }
 
 size_t
 AudioCallbackPlaySource::getSourceSamples(size_t count, float **buffer)
 {

         }
     }
 
     if (count == 0) return 0;
 
-#ifdef HAVE_RUBBERBAND
     RubberBandStretcher *ts = m_timeStretcher;
     float ratio = ts ? ts->getTimeRatio() : 1.f;
-#else
-    PhaseVocoderTimeStretcher *ts = m_timeStretcher;
-    float ratio = ts ? ts->getRatio() : 1.f;
-#endif
+
+    if (ratio != m_stretchRatio) {
+        if (!ts) {
+            std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: Time ratio change to " << m_stretchRatio << " is pending, but no stretcher is set" << std::endl;
+            m_stretchRatio = 1.f;
+        } else {
+            ts->setTimeRatio(m_stretchRatio);
+        }
+    }
+
+    if (m_target) {
+        m_lastRetrievedBlockSize = count;
+        m_lastRetrievalTimestamp = m_target->getCurrentTime();
+    }
 
     if (!ts || ratio == 1.f) {
 
 	size_t got = 0;
 

 	}
 
         applyAuditioningEffect(count, buffer);
 
         m_condition.wakeAll();
+
 	return got;
     }
 
     size_t channels = getTargetChannelCount();
-
-#ifdef HAVE_RUBBERBAND
-    bool mix = false;
-#else
-    bool mix = (channels > 1 && ts->getChannelCount() == 1);
-#endif
-
     size_t available;
-
     int warned = 0;
-
-    // We want output blocks of e.g. 1024 (probably fixed, certainly
-    // bounded).  We can provide input blocks of any size (unbounded)
-    // at the timestretcher's request.  The input block for a given
-    // output is approx output / ratio, but we can't predict it
-    // exactly, for an adaptive timestretcher.  The stretcher will
-    // need some additional buffer space.  See the time stretcher code
-    // and comments.
-
-#ifdef HAVE_RUBBERBAND
-    m_timeStretchRatioMutex.lock();
+    size_t fedToStretcher = 0;
+
+    // The input block for a given output is approx output / ratio,
+    // but we can't predict it exactly, for an adaptive timestretcher.
+
     while ((available = ts->available()) < count) {
-#else
-    while ((available = ts->getAvailableOutputSamples()) < count) {
-#endif
 
         size_t reqd = lrintf((count - available) / ratio);
-#ifdef HAVE_RUBBERBAND
         reqd = std::max(reqd, ts->getSamplesRequired());
-#else
-        reqd = std::max(reqd, ts->getRequiredInputSamples());
-#endif
         if (reqd == 0) reqd = 1;
                 
-        float *ib[channels];
-
         size_t got = reqd;
 
-        if (mix) {
-            for (size_t c = 0; c < channels; ++c) {
-                if (c == 0) ib[c] = new float[reqd]; //!!! fix -- this is a rt function
-                else ib[c] = 0;
-                RingBuffer<float> *rb = getReadRingBuffer(c);
-                if (rb) {
-                    size_t gotHere;
-                    if (c > 0) gotHere = rb->readAdding(ib[0], got);
-                    else gotHere = rb->read(ib[0], got);
-                    if (gotHere < got) got = gotHere;
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+        std::cerr << "reqd = " <<reqd << ", channels = " << channels << ", ic = " << m_stretcherInputCount << std::endl;
+#endif
+
+        for (size_t c = 0; c < channels; ++c) {
+            if (c >= m_stretcherInputCount) continue;
+            if (reqd > m_stretcherInputSizes[c]) {
+                if (c == 0) {
+                    std::cerr << "WARNING: resizing stretcher input buffer from " << m_stretcherInputSizes[c] << " to " << (reqd * 2) << std::endl;
                 }
+                delete[] m_stretcherInputs[c];
+                m_stretcherInputSizes[c] = reqd * 2;
+                m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]];
             }
-        } else {
-            for (size_t c = 0; c < channels; ++c) {
-                ib[c] = new float[reqd]; //!!! fix -- this is a rt function
-                RingBuffer<float> *rb = getReadRingBuffer(c);
-                if (rb) {
-                    size_t gotHere = rb->read(ib[c], got);
-                    if (gotHere < got) got = gotHere;
+        }
+
+        for (size_t c = 0; c < channels; ++c) {
+            if (c >= m_stretcherInputCount) continue;
+            RingBuffer<float> *rb = getReadRingBuffer(c);
+            if (rb) {
+                size_t gotHere = rb->read(m_stretcherInputs[c], got);
+                if (gotHere < got) got = gotHere;
+                
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+                if (c == 0) {
+                    std::cerr << "feeding stretcher: got " << gotHere
+                              << ", " << rb->getReadSpace() << " remain" << std::endl;
                 }
+#endif
+                
+            } else {
+                std::cerr << "WARNING: No ring buffer available for channel " << c << " in stretcher input block" << std::endl;
             }
         }
 
         if (got < reqd) {
             std::cerr << "WARNING: Read underrun in playback ("
                       << got << " < " << reqd << ")" << std::endl;
         }
-                
-#ifdef HAVE_RUBBERBAND
-        ts->process(ib, got, false);
-#else
-        ts->putInput(ib, got);
-#endif
-
-        for (size_t c = 0; c < channels; ++c) {
-            delete[] ib[c];
-        }
+
+        ts->process(m_stretcherInputs, got, false);
+
+        fedToStretcher += got;
 
         if (got == 0) break;
 
-#ifdef HAVE_RUBBERBAND
         if (ts->available() == available) {
-#else
-        if (ts->getAvailableOutputSamples() == available) {
-#endif
             std::cerr << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples (warned = " << warned << ")" << std::endl;
             if (++warned == 5) break;
         }
     }
 
-#ifdef HAVE_RUBBERBAND
     ts->retrieve(buffer, count);
-    m_timeStretchRatioMutex.unlock();
-#else
-    ts->getOutput(buffer, count);
-#endif
-
-    if (mix) {
-        for (size_t c = 1; c < channels; ++c) {
-            for (size_t i = 0; i < count; ++i) {
-                buffer[c][i] = buffer[0][i] / channels;
-            }
-        }
-        for (size_t i = 0; i < count; ++i) {
-            buffer[0][i] /= channels;
-        }
-    }
 
     applyAuditioningEffect(count, buffer);
 
     m_condition.wakeAll();
 

 	data.src_ratio = ratio;
 	data.end_of_input = 0;
 	
 	int err = 0;
 
-#ifdef HAVE_RUBBERBAND
         if (m_timeStretcher && m_timeStretcher->getTimeRatio() < 0.4) {
-#else
-        if (m_timeStretcher && m_timeStretcher->getRatio() < 0.4) {
-#endif
 #ifdef DEBUG_AUDIO_PLAY_SOURCE
             std::cout << "Using crappy converter" << std::endl;
 #endif
             err = src_process(m_crapConverter, &data);
         } else {

 
     } else {
 
 	// space must be a multiple of generatorBlockSize
 	space = (space / generatorBlockSize) * generatorBlockSize;
-	if (space == 0) return false;
+	if (space == 0) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+            std::cout << "requested fill is less than generator block size of "
+                      << generatorBlockSize << ", leaving it" << std::endl;
+#endif
+            return false;
+        }
 
 	if (tmpSize < channels * space) {
 	    delete[] tmp;
 	    tmp = new float[channels * space];
 	    tmpSize = channels * space;

     while (!s.m_exiting) {
 
 	s.unifyRingBuffers();
 	s.m_bufferScavenger.scavenge();
         s.m_pluginScavenger.scavenge();
-#ifndef HAVE_RUBBERBAND
-	s.m_timeStretcherScavenger.scavenge();
-#endif
 
 	if (work && s.m_playing && s.getSourceSampleRate()) {
 	    
 #ifdef DEBUG_AUDIO_PLAY_SOURCE
 	    std::cout << "AudioCallbackPlaySourceFillThread: not waiting" << std::endl;