# HG changeset patch
# User Chris Cannam
# Date 1426006972 0
# Node ID 72c662fe7ea311077a1a163dd3afa32581587805
# Parent  618d5816b04df9eea992a67eb092cb98288646b2
Further dedicated-types fixes

diff -r 618d5816b04d -r 72c662fe7ea3 audioio/AudioCallbackPlaySource.cpp
--- a/audioio/AudioCallbackPlaySource.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/AudioCallbackPlaySource.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -306,7 +306,7 @@
 	m_sourceSampleRate = 0;
     }
 
-    int lastEnd = 0;
+    sv_frame_t lastEnd = 0;
     for (std::set<Model *>::const_iterator i = m_models.begin();
 	 i != m_models.end(); ++i) {
 #ifdef DEBUG_AUDIO_PLAY_SOURCE
@@ -367,7 +367,7 @@
     rebuildRangeLists();
 
     if (count == 0) {
-	if (m_writeBuffers) count = m_writeBuffers->size();
+	if (m_writeBuffers) count = int(m_writeBuffers->size());
     }
 
     cerr << "current playing frame = " << getCurrentPlayingFrame() << endl;
@@ -593,7 +593,7 @@
 AudioCallbackPlaySource::getTargetBlockSize() const
 {
 //    cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << endl;
-    return m_blockSize;
+    return int(m_blockSize);
 }
 
 void
@@ -614,8 +614,8 @@
     // This method attempts to estimate which audio sample frame is
     // "currently coming through the speakers".
 
-    int targetRate = getTargetSampleRate();
-    int latency = m_playLatency; // at target rate
+    sv_samplerate_t targetRate = getTargetSampleRate();
+    sv_frame_t latency = m_playLatency; // at target rate
     RealTime latency_t = RealTime::zeroTime;
 
     if (targetRate != 0) {
@@ -655,8 +655,8 @@
 	}
     }
 
-    int readBufferFill = m_readBufferFill;
-    int lastRetrievedBlockSize = m_lastRetrievedBlockSize;
+    sv_frame_t readBufferFill = m_readBufferFill;
+    sv_frame_t lastRetrievedBlockSize = m_lastRetrievedBlockSize;
     double lastRetrievalTimestamp = m_lastRetrievalTimestamp;
     double currentTime = 0.0;
     if (m_target) currentTime = m_target->getCurrentTime();
@@ -665,7 +665,7 @@
 
     RealTime inbuffer_t = RealTime::frame2RealTime(inbuffer, targetRate);
 
-    int stretchlat = 0;
+    sv_frame_t stretchlat = 0;
     double timeRatio = 1.0;
 
     if (m_timeStretcher) {
@@ -755,7 +755,9 @@
         ++index;
     }
 
-    if (inRange >= (int)m_rangeStarts.size()) inRange = m_rangeStarts.size()-1;
+    if (inRange >= int(m_rangeStarts.size())) {
+        inRange = int(m_rangeStarts.size())-1;
+    }
 
     RealTime playing_t = bufferedto_t;
 
@@ -806,7 +808,7 @@
 
         if (inRange == 0) {
             if (looping) {
-                inRange = m_rangeStarts.size() - 1;
+                inRange = int(m_rangeStarts.size()) - 1;
             } else {
                 break;
             }
@@ -854,7 +856,7 @@
     m_rangeStarts.clear();
     m_rangeDurations.clear();
 
-    int sourceRate = getSourceSampleRate();
+    sv_samplerate_t sourceRate = getSourceSampleRate();
     if (sourceRate == 0) return;
 
     RealTime end = RealTime::frame2RealTime(m_lastModelEndFrame, sourceRate);
@@ -1060,28 +1062,28 @@
 }
 
 void
-AudioCallbackPlaySource::setTimeStretch(float factor)
+AudioCallbackPlaySource::setTimeStretch(double factor)
 {
     m_stretchRatio = factor;
 
     if (!getTargetSampleRate()) return; // have to make our stretcher later
 
-    if (m_timeStretcher || (factor == 1.f)) {
+    if (m_timeStretcher || (factor == 1.0)) {
         // stretch ratio will be set in next process call if appropriate
     } else {
         m_stretcherInputCount = getTargetChannelCount();
         RubberBandStretcher *stretcher = new RubberBandStretcher
-            (getTargetSampleRate(),
+            (int(getTargetSampleRate()),
              m_stretcherInputCount,
              RubberBandStretcher::OptionProcessRealTime,
              factor);
         RubberBandStretcher *monoStretcher = new RubberBandStretcher
-            (getTargetSampleRate(),
+            (int(getTargetSampleRate()),
              1,
              RubberBandStretcher::OptionProcessRealTime,
              factor);
         m_stretcherInputs = new float *[m_stretcherInputCount];
-        m_stretcherInputSizes = new int[m_stretcherInputCount];
+        m_stretcherInputSizes = new sv_frame_t[m_stretcherInputCount];
         for (int c = 0; c < m_stretcherInputCount; ++c) {
             m_stretcherInputSizes[c] = 16384;
             m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]];
@@ -1146,12 +1148,12 @@
     RubberBandStretcher *ts = m_timeStretcher;
     RubberBandStretcher *ms = m_monoStretcher;
 
-    float ratio = ts ? ts->getTimeRatio() : 1.f;
+    double ratio = ts ? ts->getTimeRatio() : 1.0;
 
     if (ratio != m_stretchRatio) {
         if (!ts) {
             cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: Time ratio change to " << m_stretchRatio << " is pending, but no stretcher is set" << endl;
-            m_stretchRatio = 1.f;
+            m_stretchRatio = 1.0;
         } else {
             ts->setTimeRatio(m_stretchRatio);
             if (ms) ms->setTimeRatio(m_stretchRatio);
@@ -1186,10 +1188,10 @@
 
 		// this is marginally more likely to leave our channels in
 		// sync after a processing failure than just passing "count":
-		int request = count;
+		sv_frame_t request = count;
 		if (ch > 0) request = got;
 
-		got = rb->read(buffer[ch], request);
+		got = rb->read(buffer[ch], int(request));
 	    
 #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
 		cout << "AudioCallbackPlaySource::getSamples: got " << got << " (of " << count << ") samples on channel " << ch << ", signalling for more (possibly)" << endl;
@@ -1215,20 +1217,20 @@
     }
 
     int channels = getTargetChannelCount();
-    int available;
+    sv_frame_t available;
+    sv_frame_t fedToStretcher = 0;
     int warned = 0;
-    int 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) {
 
-        int reqd = lrintf((count - available) / ratio);
-        reqd = std::max(reqd, (int)ts->getSamplesRequired());
+        sv_frame_t reqd = lrint(double(count - available) / ratio);
+        reqd = std::max(reqd, sv_frame_t(ts->getSamplesRequired()));
         if (reqd == 0) reqd = 1;
                 
-        int got = reqd;
+        sv_frame_t got = reqd;
 
 #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
         cerr << "reqd = " <<reqd << ", channels = " << channels << ", ic = " << m_stretcherInputCount << endl;
@@ -1250,11 +1252,11 @@
             if (c >= m_stretcherInputCount) continue;
             RingBuffer<float> *rb = getReadRingBuffer(c);
             if (rb) {
-                int gotHere;
+                sv_frame_t gotHere;
                 if (stretchChannels == 1 && c > 0) {
-                    gotHere = rb->readAdding(m_stretcherInputs[0], got);
+                    gotHere = rb->readAdding(m_stretcherInputs[0], int(got));
                 } else {
-                    gotHere = rb->read(m_stretcherInputs[c], got);
+                    gotHere = rb->read(m_stretcherInputs[c], int(got));
                 }
                 if (gotHere < got) got = gotHere;
                 
@@ -1341,7 +1343,7 @@
         }
     }
 
-    plugin->run(Vamp::RealTime::zeroTime, count);
+    plugin->run(Vamp::RealTime::zeroTime, int(count));
     
     for (int c = 0; c < getTargetChannelCount(); ++c) {
         for (int i = 0; i < count; ++i) {
@@ -1355,7 +1357,7 @@
 AudioCallbackPlaySource::fillBuffers()
 {
     static float *tmp = 0;
-    static int tmpSize = 0;
+    static sv_frame_t tmpSize = 0;
 
     sv_frame_t space = 0;
     for (int c = 0; c < getTargetChannelCount(); ++c) {
@@ -1408,7 +1410,7 @@
 	bufferPtrCount = channels;
     }
 
-    int generatorBlockSize = m_audioGenerator->getBlockSize();
+    sv_frame_t generatorBlockSize = m_audioGenerator->getBlockSize();
 
     if (resample && !m_converter) {
 	static bool warned = false;
@@ -1422,13 +1424,13 @@
 
 	double ratio =
 	    double(getTargetSampleRate()) / double(getSourceSampleRate());
-	orig = int(orig / ratio + 0.1);
+	orig = sv_frame_t(double(orig) / ratio + 0.1);
 
 	// orig must be a multiple of generatorBlockSize
 	orig = (orig / generatorBlockSize) * generatorBlockSize;
 	if (orig == 0) return false;
 
-	int work = std::max(orig, space);
+	sv_frame_t work = std::max(orig, space);
 
 	// We only allocate one buffer, but we use it in two halves.
 	// We place the non-interleaved values in the second half of
@@ -1490,7 +1492,7 @@
             err = src_process(m_converter, &data);
         }
 
-	int toCopy = int(got * ratio + 0.1);
+	sv_frame_t toCopy = sv_frame_t(double(got) * ratio + 0.1);
 
 	if (err) {
 	    cerr
@@ -1510,7 +1512,7 @@
 		tmp[i] = srcout[channels * i + c];
 	    }
 	    RingBuffer<float> *wb = getWriteRingBuffer(c);
-	    if (wb) wb->write(tmp, toCopy);
+	    if (wb) wb->write(tmp, int(toCopy));
 	}
 
 	m_writeBufferFill = f;
@@ -1519,7 +1521,7 @@
     } else {
 
 	// space must be a multiple of generatorBlockSize
-        int reqSpace = space;
+        sv_frame_t reqSpace = space;
 	space = (reqSpace / generatorBlockSize) * generatorBlockSize;
 	if (space == 0) {
 #ifdef DEBUG_AUDIO_PLAY_SOURCE
@@ -1545,13 +1547,13 @@
 	    }
 	}
 
-	int got = mixModels(f, space, bufferPtrs); // also modifies f
+	sv_frame_t got = mixModels(f, space, bufferPtrs); // also modifies f
 
 	for (int c = 0; c < channels; ++c) {
 
 	    RingBuffer<float> *wb = getWriteRingBuffer(c);
 	    if (wb) {
-                int actual = wb->write(bufferPtrs[c], got);
+                int actual = wb->write(bufferPtrs[c], int(got));
 #ifdef DEBUG_AUDIO_PLAY_SOURCE
 		cout << "Wrote " << actual << " samples for ch " << c << ", now "
 			  << wb->getReadSpace() << " to read" 
@@ -1759,7 +1761,7 @@
 	}
     }
 
-    int rf = m_readBufferFill;
+    sv_frame_t rf = m_readBufferFill;
     RingBuffer<float> *rb = getReadRingBuffer(0);
     if (rb) {
 	int rs = rb->getReadSpace();
@@ -1773,8 +1775,8 @@
     SVDEBUG << "AudioCallbackPlaySource::unifyRingBuffers: m_readBufferFill = " << m_readBufferFill << ", rf = " << rf << ", m_writeBufferFill = " << m_writeBufferFill << endl;
 #endif
 
-    int wf = m_writeBufferFill;
-    int skip = 0;
+    sv_frame_t wf = m_writeBufferFill;
+    sv_frame_t skip = 0;
     for (int c = 0; c < getTargetChannelCount(); ++c) {
 	RingBuffer<float> *wb = getWriteRingBuffer(c);
 	if (wb) {
@@ -1792,7 +1794,7 @@
 	    }
 
 //	    cout << "skipping " << skip << endl;
-	    wb->skip(skip);
+	    wb->skip(int(skip));
 	}
     }
 		    
@@ -1835,10 +1837,9 @@
 
 	} else {
 	    
-	    float ms = 100;
+	    double ms = 100;
 	    if (s.getSourceSampleRate() > 0) {
-		ms = float(s.m_ringBufferSize) /
-                    float(s.getSourceSampleRate()) * 1000.0;
+		ms = double(s.m_ringBufferSize) / s.getSourceSampleRate() * 1000.0;
 	    }
 	    
 	    if (s.m_playing) ms /= 10;
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/AudioCallbackPlaySource.h
--- a/audioio/AudioCallbackPlaySource.h	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/AudioCallbackPlaySource.h	Tue Mar 10 17:02:52 2015 +0000
@@ -201,7 +201,7 @@
     /**
      * Set the time stretcher factor (i.e. playback speed).
      */
-    void setTimeStretch(float factor);
+    void setTimeStretch(double factor);
 
     /**
      * Set the resampler quality, 0 - 2 where 0 is fastest and 2 is
@@ -244,7 +244,9 @@
 
     void playStatusChanged(bool isPlaying);
 
-    void sampleRateMismatch(int requested, int available, bool willResample);
+    void sampleRateMismatch(sv_samplerate_t requested,
+                            sv_samplerate_t available,
+                            bool willResample);
 
     void audioOverloadPluginDisabled();
     void audioTimeStretchMultiChannelDisabled();
@@ -280,17 +282,17 @@
     std::set<Model *>                 m_models;
     RingBufferVector                 *m_readBuffers;
     RingBufferVector                 *m_writeBuffers;
-    int                               m_readBufferFill;
-    int                               m_writeBufferFill;
+    sv_frame_t                        m_readBufferFill;
+    sv_frame_t                        m_writeBufferFill;
     Scavenger<RingBufferVector>       m_bufferScavenger;
     int                               m_sourceChannelCount;
-    int                               m_blockSize;
+    sv_frame_t                        m_blockSize;
     sv_samplerate_t                   m_sourceSampleRate;
     sv_samplerate_t                   m_targetSampleRate;
-    int                               m_playLatency;
+    sv_frame_t                        m_playLatency;
     AudioCallbackPlayTarget          *m_target;
     double                            m_lastRetrievalTimestamp;
-    int                               m_lastRetrievedBlockSize;
+    sv_frame_t                        m_lastRetrievedBlockSize;
     bool                              m_trustworthyTimestamps;
     sv_frame_t                        m_lastCurrentFrame;
     bool                              m_playing;
@@ -328,12 +330,12 @@
 
     RubberBand::RubberBandStretcher *m_timeStretcher;
     RubberBand::RubberBandStretcher *m_monoStretcher;
-    float m_stretchRatio;
+    double m_stretchRatio;
     bool m_stretchMono;
     
-    int  m_stretcherInputCount;
+    int m_stretcherInputCount;
     float **m_stretcherInputs;
-    int *m_stretcherInputSizes;
+    sv_frame_t *m_stretcherInputSizes;
 
     // Called from fill thread, m_playing true, mutex held
     // Return true if work done
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/AudioGenerator.cpp
--- a/audioio/AudioGenerator.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/AudioGenerator.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -37,7 +37,7 @@
 #include <QDir>
 #include <QFile>
 
-const int
+const sv_frame_t
 AudioGenerator::m_processingBlockSize = 1024;
 
 QString
@@ -220,11 +220,11 @@
                                      m_sourceSampleRate,
                                      m_processingBlockSize);
 
-    float clipF0 = Pitch::getFrequencyForPitch(60, 0, 440.0f); // required
+    double clipF0 = Pitch::getFrequencyForPitch(60, 0, 440.0); // required
 
     QString clipPath = QString("%1/%2.wav").arg(m_sampleDir).arg(clipId);
 
-    float level = wantsQuieterClips(model) ? 0.5 : 1.0;
+    double level = wantsQuieterClips(model) ? 0.5 : 1.0;
     if (!mixer->loadClipData(clipPath, clipF0, level)) {
         delete mixer;
         return 0;
@@ -310,7 +310,7 @@
     }
 }
 
-int
+sv_frame_t
 AudioGenerator::getBlockSize() const
 {
     return m_processingBlockSize;
@@ -334,9 +334,9 @@
     m_soloing = false;
 }
 
-int
-AudioGenerator::mixModel(Model *model, int startFrame, int frameCount,
-			 float **buffer, int fadeIn, int fadeOut)
+sv_frame_t
+AudioGenerator::mixModel(Model *model, sv_frame_t startFrame, sv_frame_t frameCount,
+			 float **buffer, sv_frame_t fadeIn, sv_frame_t fadeOut)
 {
     if (m_sourceSampleRate == 0) {
 	cerr << "WARNING: AudioGenerator::mixModel: No base source sample rate available" << endl;
@@ -393,13 +393,13 @@
     return frameCount;
 }
 
-int
+sv_frame_t
 AudioGenerator::mixDenseTimeValueModel(DenseTimeValueModel *dtvm,
-				       int startFrame, int frames,
+				       sv_frame_t startFrame, sv_frame_t frames,
 				       float **buffer, float gain, float pan,
-				       int fadeIn, int fadeOut)
+				       sv_frame_t fadeIn, sv_frame_t fadeOut)
 {
-    int maxFrames = frames + std::max(fadeIn, fadeOut);
+    sv_frame_t maxFrames = frames + std::max(fadeIn, fadeOut);
 
     int modelChannels = dtvm->getChannelCount();
 
@@ -420,7 +420,7 @@
 	m_channelBufSiz = maxFrames;
     }
 
-    int got = 0;
+    sv_frame_t got = 0;
 
     if (startFrame >= fadeIn/2) {
         got = dtvm->getData(0, modelChannels - 1,
@@ -428,7 +428,7 @@
                             frames + fadeOut/2 + fadeIn/2,
                             m_channelBuffer);
     } else {
-        int missing = fadeIn/2 - startFrame;
+        sv_frame_t missing = fadeIn/2 - startFrame;
 
         for (int c = 0; c < modelChannels; ++c) {
             m_channelBuffer[c] += missing;
@@ -462,25 +462,27 @@
 	float channelGain = gain;
 	if (pan != 0.0) {
 	    if (c == 0) {
-		if (pan > 0.0) channelGain *= 1.0 - pan;
+		if (pan > 0.0) channelGain *= 1.0f - pan;
 	    } else {
-		if (pan < 0.0) channelGain *= pan + 1.0;
+		if (pan < 0.0) channelGain *= pan + 1.0f;
 	    }
 	}
 
-	for (int i = 0; i < fadeIn/2; ++i) {
+	for (sv_frame_t i = 0; i < fadeIn/2; ++i) {
 	    float *back = buffer[c];
 	    back -= fadeIn/2;
-	    back[i] += (channelGain * m_channelBuffer[sourceChannel][i] * i) / fadeIn;
+	    back[i] +=
+                (channelGain * m_channelBuffer[sourceChannel][i] * float(i))
+                / float(fadeIn);
 	}
 
-	for (int i = 0; i < frames + fadeOut/2; ++i) {
+	for (sv_frame_t i = 0; i < frames + fadeOut/2; ++i) {
 	    float mult = channelGain;
 	    if (i < fadeIn/2) {
-		mult = (mult * i) / fadeIn;
+		mult = (mult * float(i)) / float(fadeIn);
 	    }
 	    if (i > frames - fadeOut/2) {
-		mult = (mult * ((frames + fadeOut/2) - i)) / fadeOut;
+		mult = (mult * float((frames + fadeOut/2) - i)) / float(fadeOut);
 	    }
             float val = m_channelBuffer[sourceChannel][i];
             if (i >= got) val = 0.f;
@@ -491,15 +493,15 @@
     return got;
 }
   
-int
+sv_frame_t
 AudioGenerator::mixClipModel(Model *model,
-                             int startFrame, int frames,
+                             sv_frame_t startFrame, sv_frame_t frames,
                              float **buffer, float gain, float pan)
 {
     ClipMixer *clipMixer = m_clipMixerMap[model];
     if (!clipMixer) return 0;
 
-    int blocks = frames / m_processingBlockSize;
+    int blocks = int(frames / m_processingBlockSize);
     
     //!!! todo: the below -- it matters
 
@@ -513,7 +515,7 @@
     //callback play source has to use that as a multiple for all the
     //calls to mixModel
 
-    int got = blocks * m_processingBlockSize;
+    sv_frame_t got = blocks * m_processingBlockSize;
 
 #ifdef DEBUG_AUDIO_GENERATOR
     cout << "mixModel [clip]: frames " << frames
@@ -529,7 +531,7 @@
 
     for (int i = 0; i < blocks; ++i) {
 
-	int reqStart = startFrame + i * m_processingBlockSize;
+	sv_frame_t reqStart = startFrame + i * m_processingBlockSize;
 
         NoteList notes;
         NoteExportable *exportable = dynamic_cast<NoteExportable *>(model);
@@ -544,7 +546,7 @@
 	for (NoteList::const_iterator ni = notes.begin();
              ni != notes.end(); ++ni) {
 
-	    int noteFrame = ni->start;
+	    sv_frame_t noteFrame = ni->start;
 
 	    if (noteFrame < reqStart ||
 		noteFrame >= reqStart + m_processingBlockSize) continue;
@@ -552,7 +554,7 @@
 	    while (noteOffs.begin() != noteOffs.end() &&
 		   noteOffs.begin()->frame <= noteFrame) {
 
-                int eventFrame = noteOffs.begin()->frame;
+                sv_frame_t eventFrame = noteOffs.begin()->frame;
                 if (eventFrame < reqStart) eventFrame = reqStart;
 
                 off.frameOffset = eventFrame - reqStart;
@@ -568,7 +570,7 @@
 
             on.frameOffset = noteFrame - reqStart;
             on.frequency = ni->getFrequency();
-            on.level = float(ni->velocity) / 127.0;
+            on.level = float(ni->velocity) / 127.0f;
             on.pan = pan;
 
 #ifdef DEBUG_AUDIO_GENERATOR
@@ -583,7 +585,7 @@
 	while (noteOffs.begin() != noteOffs.end() &&
 	       noteOffs.begin()->frame <= reqStart + m_processingBlockSize) {
 
-            int eventFrame = noteOffs.begin()->frame;
+            sv_frame_t eventFrame = noteOffs.begin()->frame;
             if (eventFrame < reqStart) eventFrame = reqStart;
 
             off.frameOffset = eventFrame - reqStart;
@@ -609,10 +611,10 @@
     return got;
 }
 
-int
+sv_frame_t
 AudioGenerator::mixContinuousSynthModel(Model *model,
-                                        int startFrame,
-                                        int frames,
+                                        sv_frame_t startFrame,
+                                        sv_frame_t frames,
                                         float **buffer,
                                         float gain, 
                                         float pan)
@@ -624,11 +626,11 @@
     SparseTimeValueModel *stvm = qobject_cast<SparseTimeValueModel *>(model);
     if (stvm->getScaleUnits() != "Hz") return 0;
 
-    int blocks = frames / m_processingBlockSize;
+    int blocks = int(frames / m_processingBlockSize);
 
     //!!! todo: see comment in mixClipModel
 
-    int got = blocks * m_processingBlockSize;
+    sv_frame_t got = blocks * m_processingBlockSize;
 
 #ifdef DEBUG_AUDIO_GENERATOR
     cout << "mixModel [synth]: frames " << frames
@@ -639,7 +641,7 @@
 
     for (int i = 0; i < blocks; ++i) {
 
-	int reqStart = startFrame + i * m_processingBlockSize;
+	sv_frame_t reqStart = startFrame + i * m_processingBlockSize;
 
 	for (int c = 0; c < m_targetChannelCount; ++c) {
             bufferIndexes[c] = buffer[c] + i * m_processingBlockSize;
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/AudioGenerator.h
--- a/audioio/AudioGenerator.h	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/AudioGenerator.h	Tue Mar 10 17:02:52 2015 +0000
@@ -32,6 +32,8 @@
 #include <map>
 #include <vector>
 
+#include "base/BaseTypes.h"
+
 class AudioGenerator : public QObject
 {
     Q_OBJECT
@@ -74,13 +76,13 @@
      * argument to all mixModel calls must be a multiple of this
      * value.
      */
-    virtual int getBlockSize() const;
+    virtual sv_frame_t getBlockSize() const;
 
     /**
      * Mix a single model into an output buffer.
      */
-    virtual int mixModel(Model *model, int startFrame, int frameCount,
-			    float **buffer, int fadeIn = 0, int fadeOut = 0);
+    virtual sv_frame_t mixModel(Model *model, sv_frame_t startFrame, sv_frame_t frameCount,
+			    float **buffer, sv_frame_t fadeIn = 0, sv_frame_t fadeOut = 0);
 
     /**
      * Specify that only the given set of models should be played.
@@ -97,7 +99,7 @@
     void playClipIdChanged(const Playable *, QString);
 
 protected:
-    int m_sourceSampleRate;
+    sv_samplerate_t m_sourceSampleRate;
     int m_targetChannelCount;
     int m_waveType;
 
@@ -106,10 +108,10 @@
 
     struct NoteOff {
 
-        NoteOff(float _freq, int _frame) : frequency(_freq), frame(_frame) { }
+        NoteOff(float _freq, sv_frame_t _frame) : frequency(_freq), frame(_frame) { }
 
         float frequency;
-	int frame;
+	sv_frame_t frame;
 
 	struct Comparator {
 	    bool operator()(const NoteOff &n1, const NoteOff &n2) const {
@@ -143,22 +145,22 @@
 
     static void initialiseSampleDir();
 
-    virtual int mixDenseTimeValueModel
-    (DenseTimeValueModel *model, int startFrame, int frameCount,
-     float **buffer, float gain, float pan, int fadeIn, int fadeOut);
+    virtual sv_frame_t mixDenseTimeValueModel
+    (DenseTimeValueModel *model, sv_frame_t startFrame, sv_frame_t frameCount,
+     float **buffer, float gain, float pan, sv_frame_t fadeIn, sv_frame_t fadeOut);
 
-    virtual int mixClipModel
-    (Model *model, int startFrame, int frameCount,
+    virtual sv_frame_t mixClipModel
+    (Model *model, sv_frame_t startFrame, sv_frame_t frameCount,
      float **buffer, float gain, float pan);
 
-    virtual int mixContinuousSynthModel
-    (Model *model, int startFrame, int frameCount,
+    virtual sv_frame_t mixContinuousSynthModel
+    (Model *model, sv_frame_t startFrame, sv_frame_t frameCount,
      float **buffer, float gain, float pan);
     
-    static const int m_processingBlockSize;
+    static const sv_frame_t m_processingBlockSize;
 
     float **m_channelBuffer;
-    int m_channelBufSiz;
+    sv_frame_t m_channelBufSiz;
     int m_channelBufCount;
 };
 
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/AudioJACKTarget.cpp
--- a/audioio/AudioJACKTarget.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/AudioJACKTarget.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -439,14 +439,14 @@
 	buffers[ch] = (float *)jack_port_get_buffer(m_outputs[ch], nframes);
     }
 
-    int received = 0;
+    sv_frame_t received = 0;
 
     if (m_source) {
 	received = m_source->getSourceSamples(nframes, buffers);
     }
 
     for (int ch = 0; ch < (int)m_outputs.size(); ++ch) {
-        for (int i = received; i < (int)nframes; ++i) {
+        for (sv_frame_t i = received; i < nframes; ++i) {
             buffers[ch][i] = 0.0;
         }
     }
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/AudioPortAudioTarget.cpp
--- a/audioio/AudioPortAudioTarget.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/AudioPortAudioTarget.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -52,7 +52,7 @@
     m_bufferSize = 2048;
     m_sampleRate = 44100;
     if (m_source && (m_source->getSourceSampleRate() != 0)) {
-	m_sampleRate = m_source->getSourceSampleRate();
+	m_sampleRate = int(m_source->getSourceSampleRate());
     }
 
     PaStreamParameters op;
@@ -185,7 +185,7 @@
 
 int
 AudioPortAudioTarget::process(const void *, void *outputBuffer,
-                              int nframes,
+                              sv_frame_t nframes,
                               const PaStreamCallbackTimeInfo *,
                               PaStreamCallbackFlags)
 {
@@ -244,7 +244,7 @@
 	}
     }
 	
-    int received = m_source->getSourceSamples(nframes, tmpbuf);
+    sv_frame_t received = m_source->getSourceSamples(nframes, tmpbuf);
 
     float peakLeft = 0.0, peakRight = 0.0;
 
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/AudioPortAudioTarget.h
--- a/audioio/AudioPortAudioTarget.h	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/AudioPortAudioTarget.h	Tue Mar 10 17:02:52 2015 +0000
@@ -26,6 +26,8 @@
 
 #include "AudioCallbackPlayTarget.h"
 
+#include "base/BaseTypes.h"
+
 class AudioCallbackPlaySource;
 
 class AudioPortAudioTarget : public AudioCallbackPlayTarget
@@ -46,7 +48,7 @@
     virtual void sourceModelReplaced();
 
 protected:
-    int process(const void *input, void *output, int frames,
+    int process(const void *input, void *output, sv_frame_t frames,
                 const PaStreamCallbackTimeInfo *timeInfo,
                 PaStreamCallbackFlags statusFlags);
 
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/AudioPulseAudioTarget.cpp
--- a/audioio/AudioPulseAudioTarget.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/AudioPulseAudioTarget.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -57,7 +57,7 @@
     m_bufferSize = 20480;
     m_sampleRate = 44100;
     if (m_source && (m_source->getSourceSampleRate() != 0)) {
-	m_sampleRate = m_source->getSourceSampleRate();
+	m_sampleRate = int(m_source->getSourceSampleRate());
     }
     m_spec.rate = m_sampleRate;
     m_spec.channels = 2;
@@ -141,7 +141,7 @@
     
     pa_usec_t usec = 0;
     pa_stream_get_time(m_stream, &usec);
-    return usec / 1000000.f;
+    return double(usec) / 1000000.0;
 }
 
 void
@@ -151,19 +151,19 @@
 }
 
 void
-AudioPulseAudioTarget::streamWriteStatic(pa_stream *stream,
+AudioPulseAudioTarget::streamWriteStatic(pa_stream *,
                                          size_t length,
                                          void *data)
 {
     AudioPulseAudioTarget *target = (AudioPulseAudioTarget *)data;
     
-    assert(stream == target->m_stream);
+//    assert(stream == target->m_stream);
 
     target->streamWrite(length);
 }
 
 void
-AudioPulseAudioTarget::streamWrite(int requested)
+AudioPulseAudioTarget::streamWrite(sv_frame_t requested)
 {
 #ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET_PLAY
     cout << "AudioPulseAudioTarget::streamWrite(" << requested << ")" << endl;
@@ -175,21 +175,21 @@
     pa_usec_t latency = 0;
     int negative = 0;
     if (!pa_stream_get_latency(m_stream, &latency, &negative)) {
-        int latframes = (latency / 1000000.f) * float(m_sampleRate);
+        int latframes = int(double(latency) / 1000000.0 * double(m_sampleRate));
         if (latframes > 0) m_source->setTargetPlayLatency(latframes);
     }
 
     static float *output = 0;
     static float **tmpbuf = 0;
     static int tmpbufch = 0;
-    static int tmpbufsz = 0;
+    static sv_frame_t tmpbufsz = 0;
 
     int sourceChannels = m_source->getSourceChannelCount();
 
     // Because we offer pan, we always want at least 2 channels
     if (sourceChannels < 2) sourceChannels = 2;
 
-    int nframes = requested / (sourceChannels * sizeof(float));
+    sv_frame_t nframes = requested / (sourceChannels * sizeof(float));
 
     if (nframes > m_bufferSize) {
         cerr << "WARNING: AudioPulseAudioTarget::streamWrite: nframes " << nframes << " > m_bufferSize " << m_bufferSize << endl;
@@ -223,7 +223,7 @@
         output = new float[tmpbufsz * tmpbufch];
     }
 	
-    int received = m_source->getSourceSamples(nframes, tmpbuf);
+    sv_frame_t received = m_source->getSourceSamples(nframes, tmpbuf);
 
 #ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET_PLAY
     cerr << "requested " << nframes << ", received " << received << endl;
@@ -268,12 +268,12 @@
 }
 
 void
-AudioPulseAudioTarget::streamStateChangedStatic(pa_stream *stream,
+AudioPulseAudioTarget::streamStateChangedStatic(pa_stream *,
                                                 void *data)
 {
     AudioPulseAudioTarget *target = (AudioPulseAudioTarget *)data;
     
-    assert(stream == target->m_stream);
+//    assert(stream == target->m_stream);
 
     target->streamStateChanged();
 }
@@ -303,7 +303,7 @@
             cerr << "AudioPulseAudioTarget::streamStateChanged: Failed to query latency" << endl;
         }
         cerr << "Latency = " << latency << " usec" << endl;
-        int latframes = (latency / 1000000.f) * float(m_sampleRate);
+        int latframes = int(double(latency) / 1000000.0 * m_sampleRate);
         cerr << "that's " << latframes << " frames" << endl;
 
         const pa_buffer_attr *attr;
@@ -334,12 +334,12 @@
 }
 
 void
-AudioPulseAudioTarget::contextStateChangedStatic(pa_context *context,
+AudioPulseAudioTarget::contextStateChangedStatic(pa_context *,
                                                  void *data)
 {
     AudioPulseAudioTarget *target = (AudioPulseAudioTarget *)data;
     
-    assert(context == target->m_context);
+//    assert(context == target->m_context);
 
     target->contextStateChanged();
 }
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/AudioPulseAudioTarget.h
--- a/audioio/AudioPulseAudioTarget.h	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/AudioPulseAudioTarget.h	Tue Mar 10 17:02:52 2015 +0000
@@ -46,7 +46,7 @@
     virtual void sourceModelReplaced();
 
 protected:
-    void streamWrite(int);
+    void streamWrite(sv_frame_t);
     void streamStateChanged();
     void contextStateChanged();
 
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/ClipMixer.cpp
--- a/audioio/ClipMixer.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/ClipMixer.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -20,7 +20,7 @@
 
 #include "base/Debug.h"
 
-ClipMixer::ClipMixer(int channels, int sampleRate, int blockSize) :
+ClipMixer::ClipMixer(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize) :
     m_channels(channels),
     m_sampleRate(sampleRate),
     m_blockSize(blockSize),
@@ -43,7 +43,7 @@
 }
 
 bool
-ClipMixer::loadClipData(QString path, float f0, float level)
+ClipMixer::loadClipData(QString path, double f0, double level)
 {
     if (m_clipData) {
         cerr << "ClipMixer::loadClipData: Already have clip loaded" << endl;
@@ -53,7 +53,7 @@
     SF_INFO info;
     SNDFILE *file;
     float *tmpFrames;
-    int i;
+    sv_frame_t i;
 
     info.format = 0;
     file = sf_open(path.toLocal8Bit().data(), SFM_READ, &info);
@@ -83,7 +83,7 @@
 	int j;
 	m_clipData[i] = 0.0f;
 	for (j = 0; j < info.channels; ++j) {
-	    m_clipData[i] += tmpFrames[i * info.channels + j] * level;
+	    m_clipData[i] += tmpFrames[i * info.channels + j] * float(level);
 	}
     }
 
@@ -102,19 +102,19 @@
     m_playing.clear();
 }
 
-float
-ClipMixer::getResampleRatioFor(float frequency)
+double
+ClipMixer::getResampleRatioFor(double frequency)
 {
     if (!m_clipData || !m_clipRate) return 1.0;
-    float pitchRatio = m_clipF0 / frequency;
-    float resampleRatio = m_sampleRate / m_clipRate;
+    double pitchRatio = m_clipF0 / frequency;
+    double resampleRatio = m_sampleRate / m_clipRate;
     return pitchRatio * resampleRatio;
 }
 
-int
-ClipMixer::getResampledClipDuration(float frequency)
+sv_frame_t
+ClipMixer::getResampledClipDuration(double frequency)
 {
-    return int(ceil(m_clipLength * getResampleRatioFor(frequency)));
+    return sv_frame_t(ceil(double(m_clipLength) * getResampleRatioFor(frequency)));
 }
 
 void
@@ -146,12 +146,12 @@
             levels[c] = note.level * gain;
         }
         if (note.pan != 0.0 && m_channels == 2) {
-            levels[0] *= 1.0 - note.pan;
-            levels[1] *= note.pan + 1.0;
+            levels[0] *= 1.0f - note.pan;
+            levels[1] *= note.pan + 1.0f;
         }
 
-        int start = note.frameOffset;
-        int durationHere = m_blockSize;
+        sv_frame_t start = note.frameOffset;
+        sv_frame_t durationHere = m_blockSize;
         if (start > 0) durationHere = m_blockSize - start;
 
         bool ending = false;
@@ -167,7 +167,7 @@
             }
         }
 
-        int clipDuration = getResampledClipDuration(note.frequency);
+        sv_frame_t clipDuration = getResampledClipDuration(note.frequency);
         if (start + clipDuration > 0) {
             if (start < 0 && start + clipDuration < durationHere) {
                 durationHere = start + clipDuration;
@@ -199,46 +199,46 @@
 ClipMixer::mixNote(float **toBuffers,
                    float *levels,
                    float frequency,
-                   int sourceOffset,
-                   int targetOffset,
-                   int sampleCount,
+                   sv_frame_t sourceOffset,
+                   sv_frame_t targetOffset,
+                   sv_frame_t sampleCount,
                    bool isEnd)
 {
     if (!m_clipData) return;
 
-    float ratio = getResampleRatioFor(frequency);
+    double ratio = getResampleRatioFor(frequency);
     
-    float releaseTime = 0.01;
-    int releaseSampleCount = round(releaseTime * m_sampleRate);
+    double releaseTime = 0.01;
+    sv_frame_t releaseSampleCount = sv_frame_t(round(releaseTime * m_sampleRate));
     if (releaseSampleCount > sampleCount) {
         releaseSampleCount = sampleCount;
     }
-    float releaseFraction = 1.f/releaseSampleCount;
+    double releaseFraction = 1.0/double(releaseSampleCount);
 
-    for (int i = 0; i < sampleCount; ++i) {
+    for (sv_frame_t i = 0; i < sampleCount; ++i) {
 
-        int s = sourceOffset + i;
+        sv_frame_t s = sourceOffset + i;
 
-        float os = s / ratio;
-        int osi = int(floor(os));
+        double os = double(s) / ratio;
+        sv_frame_t osi = sv_frame_t(floor(os));
 
         //!!! just linear interpolation for now (same as SV's sample
         //!!! player). a small sinc kernel would be better and
         //!!! probably "good enough"
-        float value = 0.f;
+        double value = 0.0;
         if (osi < m_clipLength) {
             value += m_clipData[osi];
         }
         if (osi + 1 < m_clipLength) {
-            value += (m_clipData[osi + 1] - m_clipData[osi]) * (os - osi);
+            value += (m_clipData[osi + 1] - m_clipData[osi]) * (os - double(osi));
         }
          
         if (isEnd && i + releaseSampleCount > sampleCount) {
-            value *= releaseFraction * (sampleCount - i); // linear ramp for release
+            value *= releaseFraction * double(sampleCount - i); // linear ramp for release
         }
 
         for (int c = 0; c < m_channels; ++c) {
-            toBuffers[c][targetOffset + i] += levels[c] * value;
+            toBuffers[c][targetOffset + i] += float(levels[c] * value);
         }
     }
 }
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/ClipMixer.h
--- a/audioio/ClipMixer.h	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/ClipMixer.h	Tue Mar 10 17:02:52 2015 +0000
@@ -19,6 +19,8 @@
 #include <QString>
 #include <vector>
 
+#include "base/BaseTypes.h"
+
 /**
  * Mix in synthetic notes produced by resampling a prerecorded
  * clip. (i.e. this is an implementation of a digital sampler in the
@@ -29,7 +31,7 @@
 class ClipMixer
 {
 public:
-    ClipMixer(int channels, int sampleRate, int blockSize);
+    ClipMixer(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize);
     ~ClipMixer();
 
     void setChannelCount(int channels);
@@ -41,19 +43,19 @@
      * and should be scaled by level (in the range 0-1) when playing
      * back.
      */
-    bool loadClipData(QString clipFilePath, float clipF0, float level);
+    bool loadClipData(QString clipFilePath, double clipF0, double level);
 
     void reset(); // discarding any playing notes
 
     struct NoteStart {
-	int frameOffset; // within current processing block
+	sv_frame_t frameOffset; // within current processing block
 	float frequency; // Hz
 	float level; // volume in range (0,1]
 	float pan; // range [-1,1]
     };
 
     struct NoteEnd {
-	int frameOffset; // in current processing block
+	sv_frame_t frameOffset; // in current processing block
         float frequency; // matching note start
     };
 
@@ -64,27 +66,27 @@
 
 private:
     int m_channels;
-    int m_sampleRate;
-    int m_blockSize;
+    sv_samplerate_t m_sampleRate;
+    sv_frame_t m_blockSize;
 
     QString m_clipPath;
 
     float *m_clipData;
-    int m_clipLength;
-    float m_clipF0;
-    float m_clipRate;
+    sv_frame_t m_clipLength;
+    double m_clipF0;
+    sv_samplerate_t m_clipRate;
 
     std::vector<NoteStart> m_playing;
 
-    float getResampleRatioFor(float frequency);
-    int getResampledClipDuration(float frequency);
+    double getResampleRatioFor(double frequency);
+    sv_frame_t getResampledClipDuration(double frequency);
 
     void mixNote(float **toBuffers, 
                  float *levels,
                  float frequency,
-                 int sourceOffset, // within resampled note
-                 int targetOffset, // within target buffer
-                 int sampleCount,
+                 sv_frame_t sourceOffset, // within resampled note
+                 sv_frame_t targetOffset, // within target buffer
+                 sv_frame_t sampleCount,
                  bool isEnd);
 };
 
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/ContinuousSynth.cpp
--- a/audioio/ContinuousSynth.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/ContinuousSynth.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -19,11 +19,11 @@
 
 #include <cmath>
 
-ContinuousSynth::ContinuousSynth(int channels, int sampleRate, int blockSize, int waveType) :
+ContinuousSynth::ContinuousSynth(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize, int waveType) :
     m_channels(channels),
     m_sampleRate(sampleRate),
     m_blockSize(blockSize),
-    m_prevF0(-1.f),
+    m_prevF0(-1.0),
     m_phase(0.0),
     m_wavetype(waveType) // 0: 3 sinusoids, 1: 1 sinusoid, 2: sawtooth, 3: square
 {
@@ -40,46 +40,47 @@
 }
 
 void
-ContinuousSynth::mix(float **toBuffers, float gain, float pan, float f0)
+ContinuousSynth::mix(float **toBuffers, float gain, float pan, float f0f)
 {
-    if (f0 == 0.f) f0 = m_prevF0;
+    double f0(f0f);
+    if (f0 == 0.0) f0 = m_prevF0;
 
-    bool wasOn = (m_prevF0 > 0.f);
-    bool nowOn = (f0 > 0.f);
+    bool wasOn = (m_prevF0 > 0.0);
+    bool nowOn = (f0 > 0.0);
 
     if (!nowOn && !wasOn) {
-    m_phase = 0;
-    return;
+        m_phase = 0;
+        return;
     }
 
-    int fadeLength = 100; // samples
+    sv_frame_t fadeLength = 100;
 
     float *levels = new float[m_channels];
     
     for (int c = 0; c < m_channels; ++c) {
-    levels[c] = gain * 0.5; // scale gain otherwise too loud compared to source
+        levels[c] = gain * 0.5f; // scale gain otherwise too loud compared to source
     }
     if (pan != 0.0 && m_channels == 2) {
-    levels[0] *= 1.0 - pan;
-    levels[1] *= pan + 1.0;
+        levels[0] *= 1.0f - pan;
+        levels[1] *= pan + 1.0f;
     }
 
 //    cerr << "ContinuousSynth::mix: f0 = " << f0 << " (from " << m_prevF0 << "), phase = " << m_phase << endl;
 
-    for (int i = 0; i < m_blockSize; ++i) {
+    for (sv_frame_t i = 0; i < m_blockSize; ++i) {
 
         double fHere = (nowOn ? f0 : m_prevF0);
 
         if (wasOn && nowOn && (f0 != m_prevF0) && (i < fadeLength)) {
             // interpolate the frequency shift
-            fHere = m_prevF0 + ((f0 - m_prevF0) * i) / fadeLength;
+            fHere = m_prevF0 + ((f0 - m_prevF0) * double(i)) / double(fadeLength);
         }
 
         double phasor = (fHere * 2 * M_PI) / m_sampleRate;
     
         m_phase = m_phase + phasor;
 
-        int harmonics = (m_sampleRate / 4) / fHere - 1;
+        int harmonics = int((m_sampleRate / 4) / fHere - 1);
         if (harmonics < 1) harmonics = 1;
 
         switch (m_wavetype) {
@@ -95,7 +96,6 @@
             break;
         }
 
-
         for (int h = 0; h < harmonics; ++h) {
 
             double v = 0;
@@ -129,15 +129,15 @@
 
             if (!wasOn && i < fadeLength) {
                 // fade in
-                v = v * (i / double(fadeLength));
+                v = v * (double(i) / double(fadeLength));
             } else if (!nowOn) {
                 // fade out
                 if (i > fadeLength) v = 0;
-                else v = v * (1.0 - (i / double(fadeLength)));
+                else v = v * (1.0 - (double(i) / double(fadeLength)));
             }
 
             for (int c = 0; c < m_channels; ++c) {
-                toBuffers[c][i] += levels[c] * v;
+                toBuffers[c][i] += float(levels[c] * v);
             }
         }
     }    
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/ContinuousSynth.h
--- a/audioio/ContinuousSynth.h	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/ContinuousSynth.h	Tue Mar 10 17:02:52 2015 +0000
@@ -15,6 +15,8 @@
 #ifndef CONTINUOUS_SYNTH_H
 #define CONTINUOUS_SYNTH_H
 
+#include "base/BaseTypes.h"
+
 /**
  * Mix into a target buffer a signal synthesised so as to sound at a
  * specific frequency. The frequency may change with each processing
@@ -24,7 +26,7 @@
 class ContinuousSynth
 {
 public:
-    ContinuousSynth(int channels, int sampleRate, int blockSize, int waveType);
+    ContinuousSynth(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize, int waveType);
     ~ContinuousSynth();
     
     void setChannelCount(int channels);
@@ -45,14 +47,14 @@
      * sound switches on and off cleanly.
      */
     void mix(float **toBuffers,
-         float gain,
-         float pan,
-         float f0);
+             float gain,
+             float pan,
+             float f0);
 
 private:
     int m_channels;
-    int m_sampleRate;
-    int m_blockSize;
+    sv_samplerate_t m_sampleRate;
+    sv_frame_t m_blockSize;
 
     double m_prevF0;
     double m_phase;
diff -r 618d5816b04d -r 72c662fe7ea3 audioio/PlaySpeedRangeMapper.cpp
--- a/audioio/PlaySpeedRangeMapper.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/audioio/PlaySpeedRangeMapper.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -49,8 +49,8 @@
     
     int half = (m_maxpos + m_minpos) / 2;
 
-    factor = sqrtf((factor - 1.0) * 1000.f);
-    int position = lrintf(((factor * (half - m_minpos)) / 100.0) + m_minpos);
+    factor = sqrt((factor - 1.0) * 1000.0);
+    int position = int(lrint(((factor * (half - m_minpos)) / 100.0) + m_minpos));
 
     if (slow) {
         position = half - position;
diff -r 618d5816b04d -r 72c662fe7ea3 configure
--- a/configure	Tue Mar 10 13:22:10 2015 +0000
+++ b/configure	Tue Mar 10 17:02:52 2015 +0000
@@ -4343,9 +4343,10 @@
 CXXFLAGS_MINIMAL="$AUTOCONF_CXXFLAGS"
 
 if test "x$GCC" = "xyes"; then
-        CXXFLAGS_DEBUG="-std=c++11 -Wall -Wextra -Werror -Woverloaded-virtual -Wformat-nonliteral -Wformat-security -Winit-self -Wswitch-enum -g -pipe"
-   	CXXFLAGS_RELEASE="-std=c++11 -g0 -O2 -Wall -pipe"
-   	CXXFLAGS_MINIMAL="-std=c++11 -g0 -O0"
+   	CXXFLAGS_ANY="-std=c++11 -Wall -Wextra -Werror -Woverloaded-virtual -Wformat-nonliteral -Wformat-security -Winit-self -Wswitch-enum -Wconversion -pipe"
+        CXXFLAGS_DEBUG="$CXXFLAGS_ANY -g"
+   	CXXFLAGS_RELEASE="$CXXFLAGS_ANY -g0 -O2"
+   	CXXFLAGS_MINIMAL="$CXXFLAGS_ANY -g0 -O0"
 fi
 
 CXXFLAGS_BUILD="$CXXFLAGS_RELEASE"
diff -r 618d5816b04d -r 72c662fe7ea3 configure.ac
--- a/configure.ac	Tue Mar 10 13:22:10 2015 +0000
+++ b/configure.ac	Tue Mar 10 17:02:52 2015 +0000
@@ -53,9 +53,10 @@
 CXXFLAGS_MINIMAL="$AUTOCONF_CXXFLAGS"
 
 if test "x$GCC" = "xyes"; then
-        CXXFLAGS_DEBUG="-std=c++11 -Wall -Wextra -Werror -Woverloaded-virtual -Wformat-nonliteral -Wformat-security -Winit-self -Wswitch-enum -g -pipe"
-   	CXXFLAGS_RELEASE="-std=c++11 -g0 -O2 -Wall -pipe"
-   	CXXFLAGS_MINIMAL="-std=c++11 -g0 -O0"
+   	CXXFLAGS_ANY="-std=c++11 -Wall -Wextra -Werror -Woverloaded-virtual -Wformat-nonliteral -Wformat-security -Winit-self -Wswitch-enum -Wconversion -pipe"
+        CXXFLAGS_DEBUG="$CXXFLAGS_ANY -g"
+   	CXXFLAGS_RELEASE="$CXXFLAGS_ANY -g0 -O2"
+   	CXXFLAGS_MINIMAL="$CXXFLAGS_ANY -g0 -O0"
 fi
 
 CXXFLAGS_BUILD="$CXXFLAGS_RELEASE"
diff -r 618d5816b04d -r 72c662fe7ea3 framework/Document.cpp
--- a/framework/Document.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/framework/Document.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -734,7 +734,7 @@
         applied.setPluginVersion
             (TransformFactory::getInstance()->
              getDefaultTransformFor(applied.getIdentifier(),
-                                    lrintf(applied.getSampleRate()))
+                                    applied.getSampleRate())
              .getPluginVersion());
 
         if (!model) {
diff -r 618d5816b04d -r 72c662fe7ea3 framework/MainWindowBase.cpp
--- a/framework/MainWindowBase.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/framework/MainWindowBase.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -178,7 +178,7 @@
 
     // set a sensible default font size for views -- cannot do this
     // in Preferences, which is in base and not supposed to use QtGui
-    int viewFontSize = QApplication::font().pointSize() * 0.9;
+    int viewFontSize = int(QApplication::font().pointSize() * 0.9);
     QSettings settings;
     settings.beginGroup("Preferences");
     viewFontSize = settings.value("view-font-size", viewFontSize).toInt();
@@ -218,8 +218,8 @@
     m_playSource = new AudioCallbackPlaySource(m_viewManager,
                                                QApplication::applicationName());
 
-    connect(m_playSource, SIGNAL(sampleRateMismatch(int, int, bool)),
-	    this,           SLOT(sampleRateMismatch(int, int, bool)));
+    connect(m_playSource, SIGNAL(sampleRateMismatch(sv_samplerate_t, sv_samplerate_t, bool)),
+	    this,           SLOT(sampleRateMismatch(sv_samplerate_t, sv_samplerate_t, bool)));
     connect(m_playSource, SIGNAL(audioOverloadPluginDisabled()),
             this,           SLOT(audioOverloadPluginDisabled()));
     connect(m_playSource, SIGNAL(audioTimeStretchMultiChannelDisabled()),
@@ -2380,7 +2380,7 @@
     else pixels = 1;
     if (pixels > 4) pixels -= 4;
 
-    int zoomLevel = (end - start) / pixels;
+    int zoomLevel = int((end - start) / pixels);
     if (zoomLevel < 1) zoomLevel = 1;
 
     currentPane->setZoomLevel(zoomLevel);
diff -r 618d5816b04d -r 72c662fe7ea3 framework/MainWindowBase.h
--- a/framework/MainWindowBase.h	Tue Mar 10 13:22:10 2015 +0000
+++ b/framework/MainWindowBase.h	Tue Mar 10 17:02:52 2015 +0000
@@ -213,7 +213,7 @@
     virtual void playSelectionToggled();
     virtual void playSoloToggled();
 
-    virtual void sampleRateMismatch(int, int, bool) = 0;
+    virtual void sampleRateMismatch(sv_samplerate_t, sv_samplerate_t, bool) = 0;
     virtual void audioOverloadPluginDisabled() = 0;
     virtual void audioTimeStretchMultiChannelDisabled() = 0;
 
diff -r 618d5816b04d -r 72c662fe7ea3 framework/SVFileReader.cpp
--- a/framework/SVFileReader.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/framework/SVFileReader.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -450,7 +450,7 @@
 
     SVDEBUG << "SVFileReader::readModel: model name \"" << name << "\"" << endl;
 
-    READ_MANDATORY(int, sampleRate, toInt);
+    READ_MANDATORY(double, sampleRate, toDouble);
 
     QString type = attributes.value("type").trimmed();
     bool isMainModel = (attributes.value("mainModel").trimmed() == "true");
@@ -478,7 +478,7 @@
 
             file.waitForData();
 
-            int rate = sampleRate;
+            sv_samplerate_t rate = sampleRate;
 
             if (Preferences::getInstance()->getFixedSampleRate() != 0) {
                 rate = Preferences::getInstance()->getFixedSampleRate();
diff -r 618d5816b04d -r 72c662fe7ea3 framework/TransformUserConfigurator.cpp
--- a/framework/TransformUserConfigurator.cpp	Tue Mar 10 13:22:10 2015 +0000
+++ b/framework/TransformUserConfigurator.cpp	Tue Mar 10 17:02:52 2015 +0000
@@ -46,8 +46,8 @@
     if (plugin && plugin->getType() == "Feature Extraction Plugin") {
 	Vamp::Plugin *vp = static_cast<Vamp::Plugin *>(plugin);
 	SVDEBUG << "TransformUserConfigurator::getChannelRange: is a VP" << endl;
-        minChannels = vp->getMinChannelCount();
-        maxChannels = vp->getMaxChannelCount();
+        minChannels = int(vp->getMinChannelCount());
+        maxChannels = int(vp->getMaxChannelCount());
         return true;
     } else {
 	SVDEBUG << "TransformUserConfigurator::getChannelRange: is not a VP" << endl;