Mercurial > hg > svapp
changeset 441:aa6fb3516e28 tonioni
Merge from cxx11 branch
| author | Chris Cannam |
|---|---|
| date | Mon, 23 Mar 2015 11:26:28 +0000 |
| parents | 2185d52b4758 (current diff) 9429622647e6 (diff) |
| children | 88ae0e53a5da |
| files | audioio/AudioCallbackPlaySource.cpp audioio/AudioGenerator.cpp audioio/ContinuousSynth.cpp |
| diffstat | 26 files changed, 667 insertions(+), 367 deletions(-) [+] |
line wrap: on
line diff
--- a/acinclude.m4 Mon Mar 23 10:04:57 2015 +0000 +++ b/acinclude.m4 Mon Mar 23 11:26:28 2015 +0000 @@ -112,3 +112,146 @@ ]) +# From autoconf archive: + +# ============================================================================ +# http://www.gnu.org/software/autoconf-archive/ax_cxx_compile_stdcxx_11.html +# ============================================================================ +# +# SYNOPSIS +# +# AX_CXX_COMPILE_STDCXX_11([ext|noext],[mandatory|optional]) +# +# DESCRIPTION +# +# Check for baseline language coverage in the compiler for the C++11 +# standard; if necessary, add switches to CXXFLAGS to enable support. +# +# The first argument, if specified, indicates whether you insist on an +# extended mode (e.g. -std=gnu++11) or a strict conformance mode (e.g. +# -std=c++11). If neither is specified, you get whatever works, with +# preference for an extended mode. +# +# The second argument, if specified 'mandatory' or if left unspecified, +# indicates that baseline C++11 support is required and that the macro +# should error out if no mode with that support is found. If specified +# 'optional', then configuration proceeds regardless, after defining +# HAVE_CXX11 if and only if a supporting mode is found. +# +# LICENSE +# +# Copyright (c) 2008 Benjamin Kosnik <bkoz@redhat.com> +# Copyright (c) 2012 Zack Weinberg <zackw@panix.com> +# Copyright (c) 2013 Roy Stogner <roystgnr@ices.utexas.edu> +# Copyright (c) 2014 Alexey Sokolov <sokolov@google.com> +# +# Copying and distribution of this file, with or without modification, are +# permitted in any medium without royalty provided the copyright notice +# and this notice are preserved. This file is offered as-is, without any +# warranty. + +m4_define([_AX_CXX_COMPILE_STDCXX_11_testbody], [[ + template <typename T> + struct check + { + static_assert(sizeof(int) <= sizeof(T), "not big enough"); + }; + + struct Base { + virtual void f() {} + }; + struct Child : public Base { + virtual void f() override {} + }; + + typedef check<check<bool>> right_angle_brackets; + + int a; + decltype(a) b; + + typedef check<int> check_type; + check_type c; + check_type&& cr = static_cast<check_type&&>(c); + + auto d = a; + auto l = [](){}; +]]) + +AC_DEFUN([AX_CXX_COMPILE_STDCXX_11], [dnl + m4_if([$1], [], [], + [$1], [ext], [], + [$1], [noext], [], + [m4_fatal([invalid argument `$1' to AX_CXX_COMPILE_STDCXX_11])])dnl + m4_if([$2], [], [ax_cxx_compile_cxx11_required=true], + [$2], [mandatory], [ax_cxx_compile_cxx11_required=true], + [$2], [optional], [ax_cxx_compile_cxx11_required=false], + [m4_fatal([invalid second argument `$2' to AX_CXX_COMPILE_STDCXX_11])]) + AC_LANG_PUSH([C++])dnl + ac_success=no + AC_CACHE_CHECK(whether $CXX supports C++11 features by default, + ax_cv_cxx_compile_cxx11, + [AC_COMPILE_IFELSE([AC_LANG_SOURCE([_AX_CXX_COMPILE_STDCXX_11_testbody])], + [ax_cv_cxx_compile_cxx11=yes], + [ax_cv_cxx_compile_cxx11=no])]) + if test x$ax_cv_cxx_compile_cxx11 = xyes; then + ac_success=yes + fi + + m4_if([$1], [noext], [], [dnl + if test x$ac_success = xno; then + for switch in -std=gnu++11 -std=gnu++0x; do + cachevar=AS_TR_SH([ax_cv_cxx_compile_cxx11_$switch]) + AC_CACHE_CHECK(whether $CXX supports C++11 features with $switch, + $cachevar, + [ac_save_CXXFLAGS="$CXXFLAGS" + CXXFLAGS="$CXXFLAGS $switch" + AC_COMPILE_IFELSE([AC_LANG_SOURCE([_AX_CXX_COMPILE_STDCXX_11_testbody])], + [eval $cachevar=yes], + [eval $cachevar=no]) + CXXFLAGS="$ac_save_CXXFLAGS"]) + if eval test x\$$cachevar = xyes; then + CXXFLAGS="$CXXFLAGS $switch" + ac_success=yes + break + fi + done + fi]) + + m4_if([$1], [ext], [], [dnl + if test x$ac_success = xno; then + for switch in -std=c++11 -std=c++0x; do + cachevar=AS_TR_SH([ax_cv_cxx_compile_cxx11_$switch]) + AC_CACHE_CHECK(whether $CXX supports C++11 features with $switch, + $cachevar, + [ac_save_CXXFLAGS="$CXXFLAGS" + CXXFLAGS="$CXXFLAGS $switch" + AC_COMPILE_IFELSE([AC_LANG_SOURCE([_AX_CXX_COMPILE_STDCXX_11_testbody])], + [eval $cachevar=yes], + [eval $cachevar=no]) + CXXFLAGS="$ac_save_CXXFLAGS"]) + if eval test x\$$cachevar = xyes; then + CXXFLAGS="$CXXFLAGS $switch" + ac_success=yes + break + fi + done + fi]) + AC_LANG_POP([C++]) + if test x$ax_cxx_compile_cxx11_required = xtrue; then + if test x$ac_success = xno; then + AC_MSG_ERROR([*** A compiler with support for C++11 language features is required.]) + fi + else + if test x$ac_success = xno; then + HAVE_CXX11=0 + AC_MSG_NOTICE([No compiler with C++11 support was found]) + else + HAVE_CXX11=1 + AC_DEFINE(HAVE_CXX11,1, + [define if the compiler supports basic C++11 syntax]) + fi + + AC_SUBST(HAVE_CXX11) + fi +]) +
--- a/audioio/AudioCallbackPlaySource.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/AudioCallbackPlaySource.cpp Mon Mar 23 11:26:28 2015 +0000 @@ -256,8 +256,8 @@ emit modelReplaced(); } - connect(model, SIGNAL(modelChangedWithin(int, int)), - this, SLOT(modelChangedWithin(int, int))); + connect(model, SIGNAL(modelChangedWithin(sv_frame_t, sv_frame_t)), + this, SLOT(modelChangedWithin(sv_frame_t, sv_frame_t))); #ifdef DEBUG_AUDIO_PLAY_SOURCE cout << "AudioCallbackPlaySource::addModel: awakening thread" << endl; @@ -267,11 +267,11 @@ } void -AudioCallbackPlaySource::modelChangedWithin(int +AudioCallbackPlaySource::modelChangedWithin(sv_frame_t #ifdef DEBUG_AUDIO_PLAY_SOURCE startFrame #endif - , int endFrame) + , sv_frame_t endFrame) { #ifdef DEBUG_AUDIO_PLAY_SOURCE SVDEBUG << "AudioCallbackPlaySource::modelChangedWithin(" << startFrame << "," << endFrame << ")" << endl; @@ -291,8 +291,8 @@ cout << "AudioCallbackPlaySource::removeModel(" << model << ")" << endl; #endif - disconnect(model, SIGNAL(modelChangedWithin(int, int)), - this, SLOT(modelChangedWithin(int, int))); + disconnect(model, SIGNAL(modelChangedWithin(sv_frame_t, sv_frame_t)), + this, SLOT(modelChangedWithin(sv_frame_t, sv_frame_t))); m_models.erase(model); @@ -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; @@ -397,7 +397,7 @@ } void -AudioCallbackPlaySource::play(int startFrame) +AudioCallbackPlaySource::play(sv_frame_t startFrame) { if (!m_sourceSampleRate) { cerr << "AudioCallbackPlaySource::play: No source sample rate available, not playing" << endl; @@ -593,29 +593,29 @@ AudioCallbackPlaySource::getTargetBlockSize() const { // cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << endl; - return m_blockSize; + return int(m_blockSize); } void -AudioCallbackPlaySource::setTargetPlayLatency(int latency) +AudioCallbackPlaySource::setTargetPlayLatency(sv_frame_t latency) { m_playLatency = latency; } -int +sv_frame_t AudioCallbackPlaySource::getTargetPlayLatency() const { return m_playLatency; } -int +sv_frame_t AudioCallbackPlaySource::getCurrentPlayingFrame() { // 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) { @@ -625,13 +625,13 @@ return getCurrentFrame(latency_t); } -int +sv_frame_t AudioCallbackPlaySource::getCurrentBufferedFrame() { return getCurrentFrame(RealTime::zeroTime); } -int +sv_frame_t AudioCallbackPlaySource::getCurrentFrame(RealTime latency_t) { // We resample when filling the ring buffer, and time-stretch when @@ -640,8 +640,8 @@ // Because of the multiple rates involved, we do the actual // calculation using RealTime instead. - int sourceRate = getSourceSampleRate(); - int targetRate = getTargetSampleRate(); + sv_samplerate_t sourceRate = getSourceSampleRate(); + sv_samplerate_t targetRate = getTargetSampleRate(); if (sourceRate == 0 || targetRate == 0) return 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) { @@ -739,7 +739,7 @@ - latency_t - stretchlat_t - lastretrieved_t - inbuffer_t + sincerequest_t; if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime; - int frame = RealTime::realTime2Frame(playing_t, sourceRate); + sv_frame_t frame = RealTime::realTime2Frame(playing_t, sourceRate); return m_viewManager->alignPlaybackFrameToReference(frame); } @@ -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; } @@ -833,7 +835,7 @@ if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime; - int frame = RealTime::realTime2Frame(playing_t, sourceRate); + sv_frame_t frame = RealTime::realTime2Frame(playing_t, sourceRate); if (m_lastCurrentFrame > 0 && !looping) { if (frame < m_lastCurrentFrame) { @@ -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); @@ -916,7 +918,7 @@ } void -AudioCallbackPlaySource::setTargetSampleRate(int sr) +AudioCallbackPlaySource::setTargetSampleRate(sv_samplerate_t sr) { bool first = (m_targetSampleRate == 0); @@ -1033,7 +1035,7 @@ clearRingBuffers(); } -int +sv_samplerate_t AudioCallbackPlaySource::getTargetSampleRate() const { if (m_targetSampleRate) return m_targetSampleRate; @@ -1053,35 +1055,35 @@ return m_sourceChannelCount; } -int +sv_samplerate_t AudioCallbackPlaySource::getSourceSampleRate() const { return m_sourceSampleRate; } 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]]; @@ -1093,11 +1095,9 @@ emit activity(tr("Change time-stretch factor to %1").arg(factor)); } -int -AudioCallbackPlaySource::getSourceSamples(int ucount, float **buffer) +sv_frame_t +AudioCallbackPlaySource::getSourceSamples(sv_frame_t count, float **buffer) { - int count = ucount; - if (!m_playing) { #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING SVDEBUG << "AudioCallbackPlaySource::getSourceSamples: Not playing" << endl; @@ -1148,12 +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); @@ -1188,10 +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; @@ -1217,20 +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; @@ -1252,11 +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; @@ -1309,7 +1309,7 @@ } void -AudioCallbackPlaySource::applyAuditioningEffect(int count, float **buffers) +AudioCallbackPlaySource::applyAuditioningEffect(sv_frame_t count, float **buffers) { if (m_auditioningPluginBypassed) return; RealTimePluginInstance *plugin = m_auditioningPlugin; @@ -1343,7 +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) { @@ -1357,13 +1357,13 @@ AudioCallbackPlaySource::fillBuffers() { static float *tmp = 0; - static int tmpSize = 0; + static sv_frame_t tmpSize = 0; - int space = 0; + sv_frame_t space = 0; for (int c = 0; c < getTargetChannelCount(); ++c) { RingBuffer<float> *wb = getWriteRingBuffer(c); if (wb) { - int spaceHere = wb->getWriteSpace(); + sv_frame_t spaceHere = wb->getWriteSpace(); if (c == 0 || spaceHere < space) space = spaceHere; } } @@ -1375,7 +1375,7 @@ return false; } - int f = m_writeBufferFill; + sv_frame_t f = m_writeBufferFill; bool readWriteEqual = (m_readBuffers == m_writeBuffers); @@ -1398,8 +1398,8 @@ int channels = getTargetChannelCount(); - int orig = space; - int got = 0; + sv_frame_t orig = space; + sv_frame_t got = 0; static float **bufferPtrs = 0; static int bufferPtrCount = 0; @@ -1410,7 +1410,7 @@ bufferPtrCount = channels; } - int generatorBlockSize = m_audioGenerator->getBlockSize(); + sv_frame_t generatorBlockSize = m_audioGenerator->getBlockSize(); if (resample && !m_converter) { static bool warned = false; @@ -1424,13 +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 @@ -1492,7 +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 @@ -1512,7 +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; @@ -1521,7 +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 @@ -1547,13 +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" @@ -1580,14 +1580,14 @@ return true; } -int -AudioCallbackPlaySource::mixModels(int &frame, int count, float **buffers) +sv_frame_t +AudioCallbackPlaySource::mixModels(sv_frame_t &frame, sv_frame_t count, float **buffers) { - int processed = 0; - int chunkStart = frame; - int chunkSize = count; - int selectionSize = 0; - int nextChunkStart = chunkStart + chunkSize; + sv_frame_t processed = 0; + sv_frame_t chunkStart = frame; + sv_frame_t chunkSize = count; + sv_frame_t selectionSize = 0; + sv_frame_t nextChunkStart = chunkStart + chunkSize; bool looping = m_viewManager->getPlayLoopMode(); bool constrained = (m_viewManager->getPlaySelectionMode() && @@ -1617,11 +1617,11 @@ nextChunkStart = chunkStart + chunkSize; selectionSize = 0; - int fadeIn = 0, fadeOut = 0; + sv_frame_t fadeIn = 0, fadeOut = 0; if (constrained) { - int rChunkStart = + sv_frame_t rChunkStart = m_viewManager->alignPlaybackFrameToReference(chunkStart); Selection selection = @@ -1643,9 +1643,9 @@ } else { - int sf = m_viewManager->alignReferenceToPlaybackFrame + sv_frame_t sf = m_viewManager->alignReferenceToPlaybackFrame (selection.getStartFrame()); - int ef = m_viewManager->alignReferenceToPlaybackFrame + sv_frame_t ef = m_viewManager->alignReferenceToPlaybackFrame (selection.getEndFrame()); selectionSize = ef - sf; @@ -1761,7 +1761,7 @@ } } - int rf = m_readBufferFill; + sv_frame_t rf = m_readBufferFill; RingBuffer<float> *rb = getReadRingBuffer(0); if (rb) { int rs = rb->getReadSpace(); @@ -1775,8 +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) { @@ -1794,7 +1794,7 @@ } // cout << "skipping " << skip << endl; - wb->skip(skip); + wb->skip(int(skip)); } } @@ -1837,10 +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;
--- a/audioio/AudioCallbackPlaySource.h Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/AudioCallbackPlaySource.h Mon Mar 23 11:26:28 2015 +0000 @@ -83,7 +83,7 @@ * from the given frame. If playback is already under way, reseek * to the given frame and continue. */ - virtual void play(int startFrame); + virtual void play(sv_frame_t startFrame); /** * Stop playback and ensure that no more data is returned. @@ -99,18 +99,18 @@ * Return the frame number that is currently expected to be coming * out of the speakers. (i.e. compensating for playback latency.) */ - virtual int getCurrentPlayingFrame(); + virtual sv_frame_t getCurrentPlayingFrame(); /** * Return the last frame that would come out of the speakers if we * stopped playback right now. */ - virtual int getCurrentBufferedFrame(); + virtual sv_frame_t getCurrentBufferedFrame(); /** * Return the frame at which playback is expected to end (if not looping). */ - virtual int getPlayEndFrame() { return m_lastModelEndFrame; } + virtual sv_frame_t getPlayEndFrame() { return m_lastModelEndFrame; } /** * Set the target and the block size of the target audio device. @@ -133,12 +133,12 @@ * highest last frame across all channels) requested via * getSamples(). The default is zero. */ - void setTargetPlayLatency(int); + void setTargetPlayLatency(sv_frame_t); /** * Get the playback latency of the target audio device. */ - int getTargetPlayLatency() const; + sv_frame_t getTargetPlayLatency() const; /** * Specify that the target audio device has a fixed sample rate @@ -147,13 +147,13 @@ * source sample rate, this class will resample automatically to * fit. */ - void setTargetSampleRate(int); + void setTargetSampleRate(sv_samplerate_t); /** * Return the sample rate set by the target audio device (or the * source sample rate if the target hasn't set one). */ - virtual int getTargetSampleRate() const; + virtual sv_samplerate_t getTargetSampleRate() const; /** * Set the current output levels for metering (for call from the @@ -189,19 +189,19 @@ * safely be called from a realtime thread. Returns 0 if there is * no source yet available. */ - virtual int getSourceSampleRate() const; + virtual sv_samplerate_t getSourceSampleRate() const; /** * Get "count" samples (at the target sample rate) of the mixed * audio data, in all channels. This may safely be called from a * realtime thread. */ - int getSourceSamples(int count, float **buffer); + sv_frame_t getSourceSamples(sv_frame_t count, float **buffer); /** * 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(); @@ -260,7 +262,7 @@ void playSelectionModeChanged(); void playParametersChanged(PlayParameters *); void preferenceChanged(PropertyContainer::PropertyName); - void modelChangedWithin(int startFrame, int endFrame); + void modelChangedWithin(sv_frame_t startFrame, sv_frame_t endFrame); protected: ViewManagerBase *m_viewManager; @@ -280,29 +282,29 @@ 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; - int m_sourceSampleRate; - int m_targetSampleRate; - int m_playLatency; + sv_frame_t m_blockSize; + sv_samplerate_t m_sourceSampleRate; + sv_samplerate_t m_targetSampleRate; + sv_frame_t m_playLatency; AudioCallbackPlayTarget *m_target; double m_lastRetrievalTimestamp; - int m_lastRetrievedBlockSize; + sv_frame_t m_lastRetrievedBlockSize; bool m_trustworthyTimestamps; - int m_lastCurrentFrame; + sv_frame_t m_lastCurrentFrame; bool m_playing; bool m_exiting; - int m_lastModelEndFrame; + sv_frame_t m_lastModelEndFrame; int m_ringBufferSize; float m_outputLeft; float m_outputRight; RealTimePluginInstance *m_auditioningPlugin; bool m_auditioningPluginBypassed; Scavenger<RealTimePluginInstance> m_pluginScavenger; - int m_playStartFrame; + sv_frame_t m_playStartFrame; bool m_playStartFramePassed; RealTime m_playStartedAt; @@ -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 @@ -343,17 +345,17 @@ // which will be count or fewer. Return in the frame argument the // new buffered frame position (which may be earlier than the // frame argument passed in, in the case of looping). - int mixModels(int &frame, int count, float **buffers); + sv_frame_t mixModels(sv_frame_t &frame, sv_frame_t count, float **buffers); // Called from getSourceSamples. - void applyAuditioningEffect(int count, float **buffers); + void applyAuditioningEffect(sv_frame_t count, float **buffers); // Ranges of current selections, if play selection is active std::vector<RealTime> m_rangeStarts; std::vector<RealTime> m_rangeDurations; void rebuildRangeLists(); - int getCurrentFrame(RealTime outputLatency); + sv_frame_t getCurrentFrame(RealTime outputLatency); class FillThread : public Thread {
--- a/audioio/AudioGenerator.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/AudioGenerator.cpp Mon Mar 23 11:26:28 2015 +0000 @@ -37,7 +37,7 @@ #include <QDir> #include <QFile> -const int +const sv_frame_t AudioGenerator::m_processingBlockSize = 1024; QString @@ -228,11 +228,11 @@ 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; @@ -318,7 +318,7 @@ } } -int +sv_frame_t AudioGenerator::getBlockSize() const { return m_processingBlockSize; @@ -342,9 +342,9 @@ 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; @@ -401,13 +401,13 @@ 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(); @@ -428,7 +428,7 @@ m_channelBufSiz = maxFrames; } - int got = 0; + sv_frame_t got = 0; if (startFrame >= fadeIn/2) { got = dtvm->getData(0, modelChannels - 1, @@ -436,7 +436,7 @@ 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; @@ -470,25 +470,27 @@ 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; @@ -499,15 +501,15 @@ 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 @@ -521,7 +523,7 @@ //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 @@ -537,7 +539,7 @@ 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); @@ -552,7 +554,7 @@ 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; @@ -560,7 +562,7 @@ 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; @@ -576,7 +578,7 @@ 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 @@ -591,7 +593,7 @@ 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; @@ -617,10 +619,10 @@ 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) @@ -632,11 +634,11 @@ 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 @@ -647,7 +649,7 @@ 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;
--- a/audioio/AudioGenerator.h Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/AudioGenerator.h Mon Mar 23 11:26:28 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; };
--- a/audioio/AudioJACKTarget.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/AudioJACKTarget.cpp Mon Mar 23 11:26:28 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; } }
--- a/audioio/AudioPortAudioTarget.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/AudioPortAudioTarget.cpp Mon Mar 23 11:26:28 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;
--- a/audioio/AudioPortAudioTarget.h Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/AudioPortAudioTarget.h Mon Mar 23 11:26:28 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);
--- a/audioio/AudioPulseAudioTarget.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/AudioPulseAudioTarget.cpp Mon Mar 23 11:26:28 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(); }
--- a/audioio/AudioPulseAudioTarget.h Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/AudioPulseAudioTarget.h Mon Mar 23 11:26:28 2015 +0000 @@ -46,7 +46,7 @@ virtual void sourceModelReplaced(); protected: - void streamWrite(int); + void streamWrite(sv_frame_t); void streamStateChanged(); void contextStateChanged();
--- a/audioio/ClipMixer.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/ClipMixer.cpp Mon Mar 23 11:26:28 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); } } }
--- a/audioio/ClipMixer.h Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/ClipMixer.h Mon Mar 23 11:26:28 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); };
--- a/audioio/ContinuousSynth.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/ContinuousSynth.cpp Mon Mar 23 11:26:28 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; } - 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); } } }
--- a/audioio/ContinuousSynth.h Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/ContinuousSynth.h Mon Mar 23 11:26:28 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;
--- a/audioio/PlaySpeedRangeMapper.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/PlaySpeedRangeMapper.cpp Mon Mar 23 11:26:28 2015 +0000 @@ -25,23 +25,23 @@ } int -PlaySpeedRangeMapper::getPositionForValue(float value) const +PlaySpeedRangeMapper::getPositionForValue(double value) const { // value is percent - float factor = getFactorForValue(value); + double factor = getFactorForValue(value); int position = getPositionForFactor(factor); return position; } int -PlaySpeedRangeMapper::getPositionForValueUnclamped(float value) const +PlaySpeedRangeMapper::getPositionForValueUnclamped(double value) const { // We don't really provide this return getPositionForValue(value); } int -PlaySpeedRangeMapper::getPositionForFactor(float factor) const +PlaySpeedRangeMapper::getPositionForFactor(double factor) const { bool slow = (factor > 1.0); @@ -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; @@ -63,37 +63,37 @@ return position; } -float +double PlaySpeedRangeMapper::getValueForPosition(int position) const { - float factor = getFactorForPosition(position); - float pc = getValueForFactor(factor); + double factor = getFactorForPosition(position); + double pc = getValueForFactor(factor); return pc; } -float +double PlaySpeedRangeMapper::getValueForPositionUnclamped(int position) const { // We don't really provide this return getValueForPosition(position); } -float -PlaySpeedRangeMapper::getValueForFactor(float factor) const +double +PlaySpeedRangeMapper::getValueForFactor(double factor) const { - float pc; + double pc; if (factor < 1.0) pc = ((1.0 / factor) - 1.0) * 100.0; else pc = (1.0 - factor) * 100.0; // cerr << "position = " << position << " percent = " << pc << endl; return pc; } -float -PlaySpeedRangeMapper::getFactorForValue(float value) const +double +PlaySpeedRangeMapper::getFactorForValue(double value) const { // value is percent - float factor; + double factor; if (value <= 0) { factor = 1.0 - (value / 100.0); @@ -105,7 +105,7 @@ return factor; } -float +double PlaySpeedRangeMapper::getFactorForPosition(int position) const { bool slow = false; @@ -124,7 +124,7 @@ // position is between min and half (inclusive) - float factor; + double factor; if (position == m_minpos) { factor = 1.0;
--- a/audioio/PlaySpeedRangeMapper.h Mon Mar 23 10:04:57 2015 +0000 +++ b/audioio/PlaySpeedRangeMapper.h Mon Mar 23 11:26:28 2015 +0000 @@ -23,17 +23,17 @@ public: PlaySpeedRangeMapper(int minpos, int maxpos); - virtual int getPositionForValue(float value) const; - virtual int getPositionForValueUnclamped(float value) const; + virtual int getPositionForValue(double value) const; + virtual int getPositionForValueUnclamped(double value) const; - virtual float getValueForPosition(int position) const; - virtual float getValueForPositionUnclamped(int position) const; + virtual double getValueForPosition(int position) const; + virtual double getValueForPositionUnclamped(int position) const; - int getPositionForFactor(float factor) const; - float getValueForFactor(float factor) const; + int getPositionForFactor(double factor) const; + double getValueForFactor(double factor) const; - float getFactorForPosition(int position) const; - float getFactorForValue(float value) const; + double getFactorForPosition(int position) const; + double getFactorForValue(double value) const; virtual QString getUnit() const;
--- a/configure Mon Mar 23 10:04:57 2015 +0000 +++ b/configure Mon Mar 23 11:26:28 2015 +0000 @@ -673,6 +673,7 @@ EGREP GREP CXXCPP +HAVE_CXX11 MKDIR_P INSTALL_DATA INSTALL_SCRIPT @@ -3449,6 +3450,146 @@ $as_echo "$MKDIR_P" >&6; } +# We are daringly making use of C++11 now + + ax_cxx_compile_cxx11_required=true + ac_ext=cpp +ac_cpp='$CXXCPP $CPPFLAGS' +ac_compile='$CXX -c $CXXFLAGS $CPPFLAGS conftest.$ac_ext >&5' +ac_link='$CXX -o conftest$ac_exeext $CXXFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS >&5' +ac_compiler_gnu=$ac_cv_cxx_compiler_gnu + ac_success=no + { $as_echo "$as_me:${as_lineno-$LINENO}: checking whether $CXX supports C++11 features by default" >&5 +$as_echo_n "checking whether $CXX supports C++11 features by default... " >&6; } +if ${ax_cv_cxx_compile_cxx11+:} false; then : + $as_echo_n "(cached) " >&6 +else + cat confdefs.h - <<_ACEOF >conftest.$ac_ext +/* end confdefs.h. */ + + template <typename T> + struct check + { + static_assert(sizeof(int) <= sizeof(T), "not big enough"); + }; + + struct Base { + virtual void f() {} + }; + struct Child : public Base { + virtual void f() override {} + }; + + typedef check<check<bool>> right_angle_brackets; + + int a; + decltype(a) b; + + typedef check<int> check_type; + check_type c; + check_type&& cr = static_cast<check_type&&>(c); + + auto d = a; + auto l = [](){}; + +_ACEOF +if ac_fn_cxx_try_compile "$LINENO"; then : + ax_cv_cxx_compile_cxx11=yes +else + ax_cv_cxx_compile_cxx11=no +fi +rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext +fi +{ $as_echo "$as_me:${as_lineno-$LINENO}: result: $ax_cv_cxx_compile_cxx11" >&5 +$as_echo "$ax_cv_cxx_compile_cxx11" >&6; } + if test x$ax_cv_cxx_compile_cxx11 = xyes; then + ac_success=yes + fi + + + + if test x$ac_success = xno; then + for switch in -std=c++11 -std=c++0x; do + cachevar=`$as_echo "ax_cv_cxx_compile_cxx11_$switch" | $as_tr_sh` + { $as_echo "$as_me:${as_lineno-$LINENO}: checking whether $CXX supports C++11 features with $switch" >&5 +$as_echo_n "checking whether $CXX supports C++11 features with $switch... " >&6; } +if eval \${$cachevar+:} false; then : + $as_echo_n "(cached) " >&6 +else + ac_save_CXXFLAGS="$CXXFLAGS" + CXXFLAGS="$CXXFLAGS $switch" + cat confdefs.h - <<_ACEOF >conftest.$ac_ext +/* end confdefs.h. */ + + template <typename T> + struct check + { + static_assert(sizeof(int) <= sizeof(T), "not big enough"); + }; + + struct Base { + virtual void f() {} + }; + struct Child : public Base { + virtual void f() override {} + }; + + typedef check<check<bool>> right_angle_brackets; + + int a; + decltype(a) b; + + typedef check<int> check_type; + check_type c; + check_type&& cr = static_cast<check_type&&>(c); + + auto d = a; + auto l = [](){}; + +_ACEOF +if ac_fn_cxx_try_compile "$LINENO"; then : + eval $cachevar=yes +else + eval $cachevar=no +fi +rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext + CXXFLAGS="$ac_save_CXXFLAGS" +fi +eval ac_res=\$$cachevar + { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_res" >&5 +$as_echo "$ac_res" >&6; } + if eval test x\$$cachevar = xyes; then + CXXFLAGS="$CXXFLAGS $switch" + ac_success=yes + break + fi + done + fi + ac_ext=cpp +ac_cpp='$CXXCPP $CPPFLAGS' +ac_compile='$CXX -c $CXXFLAGS $CPPFLAGS conftest.$ac_ext >&5' +ac_link='$CXX -o conftest$ac_exeext $CXXFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS >&5' +ac_compiler_gnu=$ac_cv_cxx_compiler_gnu + + if test x$ax_cxx_compile_cxx11_required = xtrue; then + if test x$ac_success = xno; then + as_fn_error $? "*** A compiler with support for C++11 language features is required." "$LINENO" 5 + fi + else + if test x$ac_success = xno; then + HAVE_CXX11=0 + { $as_echo "$as_me:${as_lineno-$LINENO}: No compiler with C++11 support was found" >&5 +$as_echo "$as_me: No compiler with C++11 support was found" >&6;} + else + HAVE_CXX11=1 + +$as_echo "#define HAVE_CXX11 1" >>confdefs.h + + fi + + + fi + ac_ext=cpp ac_cpp='$CXXCPP $CPPFLAGS' @@ -4202,9 +4343,10 @@ CXXFLAGS_MINIMAL="$AUTOCONF_CXXFLAGS" if test "x$GCC" = "xyes"; then - CXXFLAGS_DEBUG="-Wall -Wextra -Werror -Woverloaded-virtual -Wformat-nonliteral -Wformat-security -Winit-self -Wswitch-enum -g -pipe" - CXXFLAGS_RELEASE="-g0 -O2 -Wall -pipe" - CXXFLAGS_MINIMAL="-g0 -O0" + CXXFLAGS_ANY="-Wall -Wextra -Werror -Woverloaded-virtual -Wformat-nonliteral -Wformat-security -Winit-self -Wswitch-enum -Wconversion -pipe" + CXXFLAGS_DEBUG="$CXXFLAGS_ANY -g" + CXXFLAGS_RELEASE="$CXXFLAGS_ANY -g0 -O2" + CXXFLAGS_MINIMAL="$CXXFLAGS_ANY -g0 -O0" fi CXXFLAGS_BUILD="$CXXFLAGS_RELEASE"
--- a/configure.ac Mon Mar 23 10:04:57 2015 +0000 +++ b/configure.ac Mon Mar 23 11:26:28 2015 +0000 @@ -25,6 +25,9 @@ AC_PROG_INSTALL AC_PROG_MKDIR_P +# We are daringly making use of C++11 now +AX_CXX_COMPILE_STDCXX_11(noext) + AC_HEADER_STDC # These are the flags Autoconf guesses for us; we use them later if @@ -50,9 +53,10 @@ CXXFLAGS_MINIMAL="$AUTOCONF_CXXFLAGS" if test "x$GCC" = "xyes"; then - CXXFLAGS_DEBUG="-Wall -Wextra -Werror -Woverloaded-virtual -Wformat-nonliteral -Wformat-security -Winit-self -Wswitch-enum -g -pipe" - CXXFLAGS_RELEASE="-g0 -O2 -Wall -pipe" - CXXFLAGS_MINIMAL="-g0 -O0" + CXXFLAGS_ANY="-Wall -Wextra -Werror -Woverloaded-virtual -Wformat-nonliteral -Wformat-security -Winit-self -Wswitch-enum -Wconversion -pipe" + CXXFLAGS_DEBUG="$CXXFLAGS_ANY -g" + CXXFLAGS_RELEASE="$CXXFLAGS_ANY -g0 -O2" + CXXFLAGS_MINIMAL="$CXXFLAGS_ANY -g0 -O0" fi CXXFLAGS_BUILD="$CXXFLAGS_RELEASE"
--- a/framework/Document.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/framework/Document.cpp Mon Mar 23 11:26:28 2015 +0000 @@ -734,7 +734,7 @@ applied.setPluginVersion (TransformFactory::getInstance()-> getDefaultTransformFor(applied.getIdentifier(), - lrintf(applied.getSampleRate())) + applied.getSampleRate()) .getPluginVersion()); if (!model) {
--- a/framework/MainWindowBase.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/framework/MainWindowBase.cpp Mon Mar 23 11:26:28 2015 +0000 @@ -106,9 +106,6 @@ #include <cstdio> #include <errno.h> - - - using std::vector; using std::map; using std::set; @@ -161,6 +158,9 @@ { Profiler profiler("MainWindowBase::MainWindowBase"); + qRegisterMetaType<sv_frame_t>("sv_frame_t"); + qRegisterMetaType<sv_samplerate_t>("sv_samplerate_t"); + #ifdef Q_WS_X11 XSetErrorHandler(handle_x11_error); #endif @@ -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()), @@ -228,14 +228,14 @@ connect(m_viewManager, SIGNAL(outputLevelsChanged(float, float)), this, SLOT(outputLevelsChanged(float, float))); - connect(m_viewManager, SIGNAL(playbackFrameChanged(int)), - this, SLOT(playbackFrameChanged(int))); - - connect(m_viewManager, SIGNAL(globalCentreFrameChanged(int)), - this, SLOT(globalCentreFrameChanged(int))); - - connect(m_viewManager, SIGNAL(viewCentreFrameChanged(View *, int)), - this, SLOT(viewCentreFrameChanged(View *, int))); + connect(m_viewManager, SIGNAL(playbackFrameChanged(sv_frame_t)), + this, SLOT(playbackFrameChanged(sv_frame_t))); + + connect(m_viewManager, SIGNAL(globalCentreFrameChanged(sv_frame_t)), + this, SLOT(globalCentreFrameChanged(sv_frame_t))); + + connect(m_viewManager, SIGNAL(viewCentreFrameChanged(View *, sv_frame_t)), + this, SLOT(viewCentreFrameChanged(View *, sv_frame_t))); connect(m_viewManager, SIGNAL(viewZoomLevelChanged(View *, int, bool)), this, SLOT(viewZoomLevelChanged(View *, int, bool))); @@ -691,7 +691,7 @@ // frame before switching to whichever one we decide we want to // switch to, regardless of our efforts. - int frame = m_playSource->getCurrentBufferedFrame(); + sv_frame_t frame = m_playSource->getCurrentBufferedFrame(); cerr << "currentPaneChanged: current frame (in ref model) = " << frame << endl; @@ -780,7 +780,7 @@ Pane *currentPane = m_paneStack->getCurrentPane(); if (!currentPane) return; - int startFrame, endFrame; + sv_frame_t startFrame, endFrame; if (currentPane->getStartFrame() < 0) startFrame = 0; else startFrame = currentPane->getStartFrame(); @@ -851,11 +851,11 @@ void MainWindowBase::pasteAtPlaybackPosition() { - int pos = getFrame(); + sv_frame_t pos = getFrame(); Clipboard &clipboard = m_viewManager->getClipboard(); if (!clipboard.empty()) { - int firstEventFrame = clipboard.getPoints()[0].getFrame(); - int offset = 0; + sv_frame_t firstEventFrame = clipboard.getPoints()[0].getFrame(); + sv_frame_t offset = 0; if (firstEventFrame < 0) { offset = pos - firstEventFrame; } else if (firstEventFrame < pos) { @@ -868,7 +868,7 @@ } void -MainWindowBase::pasteRelative(int offset) +MainWindowBase::pasteRelative(sv_frame_t offset) { Pane *currentPane = m_paneStack->getCurrentPane(); if (!currentPane) return; @@ -936,7 +936,7 @@ // FrameTimer method -int +sv_frame_t MainWindowBase::getFrame() const { if (m_playSource && m_playSource->isPlaying()) { @@ -958,8 +958,8 @@ MultiSelection::SelectionList selections = m_viewManager->getSelections(); for (MultiSelection::SelectionList::iterator i = selections.begin(); i != selections.end(); ++i) { - int start = i->getStartFrame(); - int end = i->getEndFrame(); + sv_frame_t start = i->getStartFrame(); + sv_frame_t end = i->getEndFrame(); if (start != end) { insertInstantAt(start); insertInstantAt(end); @@ -968,7 +968,7 @@ } void -MainWindowBase::insertInstantAt(int frame) +MainWindowBase::insertInstantAt(sv_frame_t frame) { Pane *pane = m_paneStack->getCurrentPane(); if (!pane) { @@ -1060,8 +1060,8 @@ MultiSelection::SelectionList selections = m_viewManager->getSelections(); for (MultiSelection::SelectionList::iterator i = selections.begin(); i != selections.end(); ++i) { - int start = i->getStartFrame(); - int end = i->getEndFrame(); + sv_frame_t start = i->getStartFrame(); + sv_frame_t end = i->getEndFrame(); if (start < end) { insertItemAt(start, end - start); } @@ -1069,7 +1069,7 @@ } void -MainWindowBase::insertItemAt(int frame, int duration) +MainWindowBase::insertItemAt(sv_frame_t frame, sv_frame_t duration) { Pane *pane = m_paneStack->getCurrentPane(); if (!pane) { @@ -1078,10 +1078,10 @@ // ugh! - int alignedStart = pane->alignFromReference(frame); - int alignedEnd = pane->alignFromReference(frame + duration); + sv_frame_t alignedStart = pane->alignFromReference(frame); + sv_frame_t alignedEnd = pane->alignFromReference(frame + duration); if (alignedStart >= alignedEnd) return; - int alignedDuration = alignedEnd - alignedStart; + sv_frame_t alignedDuration = alignedEnd - alignedStart; Command *c = 0; @@ -1281,7 +1281,7 @@ m_openingAudioFile = true; - int rate = 0; + sv_samplerate_t rate = 0; if (Preferences::getInstance()->getFixedSampleRate() != 0) { rate = Preferences::getInstance()->getFixedSampleRate(); @@ -1990,7 +1990,7 @@ MainWindowBase::FileOpenStatus MainWindowBase::openLayersFromRDF(FileSource source) { - int rate = 0; + sv_samplerate_t rate = 0; SVDEBUG << "MainWindowBase::openLayersFromRDF" << endl; @@ -2370,8 +2370,8 @@ Model *model = getMainModel(); if (!model) return; - int start = model->getStartFrame(); - int end = model->getEndFrame(); + sv_frame_t start = model->getStartFrame(); + sv_frame_t end = model->getEndFrame(); if (m_playSource) end = std::max(end, m_playSource->getPlayEndFrame()); int pixels = currentPane->width(); @@ -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); @@ -2609,18 +2609,18 @@ { if (!getMainModel()) return; - int frame = m_viewManager->getPlaybackFrame(); + sv_frame_t frame = m_viewManager->getPlaybackFrame(); ++frame; Pane *pane = m_paneStack->getCurrentPane(); Layer *layer = getSnapLayer(); - int sr = getMainModel()->getSampleRate(); + sv_samplerate_t sr = getMainModel()->getSampleRate(); if (!layer) { frame = RealTime::realTime2Frame (RealTime::frame2RealTime(frame, sr) + m_defaultFfwdRwdStep, sr); - if (frame > int(getMainModel()->getEndFrame())) { + if (frame > getMainModel()->getEndFrame()) { frame = getMainModel()->getEndFrame(); } @@ -2639,12 +2639,12 @@ if (frame < 0) frame = 0; if (m_viewManager->getPlaySelectionMode()) { - frame = m_viewManager->constrainFrameToSelection(int(frame)); + frame = m_viewManager->constrainFrameToSelection(frame); } m_viewManager->setPlaybackFrame(frame); - if (frame == (int)getMainModel()->getEndFrame() && + if (frame == getMainModel()->getEndFrame() && m_playSource && m_playSource->isPlaying() && !m_viewManager->getPlayLoopMode()) { @@ -2663,7 +2663,7 @@ stop(); } - int frame = getMainModel()->getEndFrame(); + sv_frame_t frame = getMainModel()->getEndFrame(); if (m_viewManager->getPlaySelectionMode()) { frame = m_viewManager->constrainFrameToSelection(frame); @@ -2681,7 +2681,7 @@ if (!layer) { ffwd(); return; } Pane *pane = m_paneStack->getCurrentPane(); - int frame = m_viewManager->getPlaybackFrame(); + sv_frame_t frame = m_viewManager->getPlaybackFrame(); int resolution = 0; if (pane) frame = pane->alignFromReference(frame); @@ -2695,12 +2695,12 @@ if (frame < 0) frame = 0; if (m_viewManager->getPlaySelectionMode()) { - frame = m_viewManager->constrainFrameToSelection(int(frame)); + frame = m_viewManager->constrainFrameToSelection(frame); } m_viewManager->setPlaybackFrame(frame); - if (frame == (int)getMainModel()->getEndFrame() && + if (frame == getMainModel()->getEndFrame() && m_playSource && m_playSource->isPlaying() && !m_viewManager->getPlayLoopMode()) { @@ -2713,12 +2713,12 @@ { if (!getMainModel()) return; - int frame = m_viewManager->getPlaybackFrame(); + sv_frame_t frame = m_viewManager->getPlaybackFrame(); if (frame > 0) --frame; Pane *pane = m_paneStack->getCurrentPane(); Layer *layer = getSnapLayer(); - int sr = getMainModel()->getSampleRate(); + sv_samplerate_t sr = getMainModel()->getSampleRate(); // when rewinding during playback, we want to allow a period // following a rewind target point at which the rewind will go to @@ -2734,7 +2734,7 @@ frame = RealTime::realTime2Frame (RealTime::frame2RealTime(frame, sr) - m_defaultFfwdRwdStep, sr); - if (frame < int(getMainModel()->getStartFrame())) { + if (frame < getMainModel()->getStartFrame()) { frame = getMainModel()->getStartFrame(); } @@ -2753,7 +2753,7 @@ if (frame < 0) frame = 0; if (m_viewManager->getPlaySelectionMode()) { - frame = m_viewManager->constrainFrameToSelection(int(frame)); + frame = m_viewManager->constrainFrameToSelection(frame); } m_viewManager->setPlaybackFrame(frame); @@ -2764,7 +2764,7 @@ { if (!getMainModel()) return; - int frame = getMainModel()->getStartFrame(); + sv_frame_t frame = getMainModel()->getStartFrame(); if (m_viewManager->getPlaySelectionMode()) { frame = m_viewManager->constrainFrameToSelection(frame); @@ -2782,7 +2782,7 @@ if (!layer) { rewind(); return; } Pane *pane = m_paneStack->getCurrentPane(); - int frame = m_viewManager->getPlaybackFrame(); + sv_frame_t frame = m_viewManager->getPlaybackFrame(); int resolution = 0; if (pane) frame = pane->alignFromReference(frame); @@ -2796,7 +2796,7 @@ if (frame < 0) frame = 0; if (m_viewManager->getPlaySelectionMode()) { - frame = m_viewManager->constrainFrameToSelection(int(frame)); + frame = m_viewManager->constrainFrameToSelection(frame); } m_viewManager->setPlaybackFrame(frame); @@ -3009,24 +3009,24 @@ MainWindowBase::connectLayerEditDialog(ModelDataTableDialog *dialog) { connect(m_viewManager, - SIGNAL(globalCentreFrameChanged(int)), + SIGNAL(globalCentreFrameChanged(sv_frame_t)), dialog, - SLOT(userScrolledToFrame(int))); + SLOT(userScrolledToFrame(sv_frame_t))); connect(m_viewManager, - SIGNAL(playbackFrameChanged(int)), + SIGNAL(playbackFrameChanged(sv_frame_t)), dialog, - SLOT(playbackScrolledToFrame(int))); + SLOT(playbackScrolledToFrame(sv_frame_t))); connect(dialog, - SIGNAL(scrollToFrame(int)), + SIGNAL(scrollToFrame(sv_frame_t)), m_viewManager, - SLOT(setGlobalCentreFrame(int))); + SLOT(setGlobalCentreFrame(sv_frame_t))); connect(dialog, - SIGNAL(scrollToFrame(int)), + SIGNAL(scrollToFrame(sv_frame_t)), m_viewManager, - SLOT(setPlaybackFrame(int))); + SLOT(setPlaybackFrame(sv_frame_t))); } void @@ -3136,7 +3136,7 @@ } void -MainWindowBase::playbackFrameChanged(int frame) +MainWindowBase::playbackFrameChanged(sv_frame_t frame) { if (!(m_playSource && m_playSource->isPlaying()) || !getMainModel()) return; @@ -3172,7 +3172,7 @@ } void -MainWindowBase::globalCentreFrameChanged(int ) +MainWindowBase::globalCentreFrameChanged(sv_frame_t ) { if ((m_playSource && m_playSource->isPlaying()) || !getMainModel()) return; Pane *p = 0; @@ -3182,7 +3182,7 @@ } void -MainWindowBase::viewCentreFrameChanged(View *v, int frame) +MainWindowBase::viewCentreFrameChanged(View *v, sv_frame_t frame) { // SVDEBUG << "MainWindowBase::viewCentreFrameChanged(" << v << "," << frame << ")" << endl;
--- a/framework/MainWindowBase.h Mon Mar 23 10:04:57 2015 +0000 +++ b/framework/MainWindowBase.h Mon Mar 23 11:26:28 2015 +0000 @@ -111,7 +111,7 @@ virtual bool saveSessionTemplate(QString path); /// Implementation of FrameTimer interface method - virtual int getFrame() const; + virtual sv_frame_t getFrame() const; void setDefaultFfwdRwdStep(RealTime step) { m_defaultFfwdRwdStep = step; @@ -213,13 +213,13 @@ 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; - virtual void playbackFrameChanged(int); - virtual void globalCentreFrameChanged(int); - virtual void viewCentreFrameChanged(View *, int); + virtual void playbackFrameChanged(sv_frame_t); + virtual void globalCentreFrameChanged(sv_frame_t); + virtual void viewCentreFrameChanged(View *, sv_frame_t); virtual void viewZoomLevelChanged(View *, int, bool); virtual void outputLevelsChanged(float, float) = 0; @@ -236,14 +236,14 @@ virtual void copy(); virtual void paste(); virtual void pasteAtPlaybackPosition(); - virtual void pasteRelative(int offset); + virtual void pasteRelative(sv_frame_t offset); virtual void deleteSelected(); virtual void insertInstant(); - virtual void insertInstantAt(int); + virtual void insertInstantAt(sv_frame_t); virtual void insertInstantsAtBoundaries(); virtual void insertItemAtSelection(); - virtual void insertItemAt(int, int); + virtual void insertItemAt(sv_frame_t, sv_frame_t); virtual void renumberInstants(); virtual void documentModified(); @@ -366,7 +366,7 @@ virtual void setWindowSize(int width, int height) { m_mw->resizeConstrained(QSize(width, height)); } - virtual void addSelection(int start, int end) { + virtual void addSelection(sv_frame_t start, sv_frame_t end) { m_mw->m_viewManager->addSelectionQuietly(Selection(start, end)); } protected:
--- a/framework/SVFileReader.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/framework/SVFileReader.cpp Mon Mar 23 11:26:28 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();
--- a/framework/SVFileReader.h Mon Mar 23 10:04:57 2015 +0000 +++ b/framework/SVFileReader.h Mon Mar 23 11:26:28 2015 +0000 @@ -34,7 +34,7 @@ virtual ~SVFileReaderPaneCallback(); virtual Pane *addPane() = 0; virtual void setWindowSize(int width, int height) = 0; - virtual void addSelection(int start, int end) = 0; + virtual void addSelection(sv_frame_t start, sv_frame_t end) = 0; }; /**
--- a/framework/TransformUserConfigurator.cpp Mon Mar 23 10:04:57 2015 +0000 +++ b/framework/TransformUserConfigurator.cpp Mon Mar 23 11:26:28 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; @@ -62,8 +62,8 @@ Vamp::PluginBase *plugin, Model *&inputModel, AudioPlaySource *source, - int startFrame, - int duration, + sv_frame_t startFrame, + sv_frame_t duration, const QMap<QString, Model *> &modelMap, QStringList candidateModelNames, QString defaultModelName)
--- a/framework/TransformUserConfigurator.h Mon Mar 23 10:04:57 2015 +0000 +++ b/framework/TransformUserConfigurator.h Mon Mar 23 11:26:28 2015 +0000 @@ -27,8 +27,8 @@ Vamp::PluginBase *plugin, Model *&inputModel, AudioPlaySource *source, - int startFrame, - int duration, + sv_frame_t startFrame, + sv_frame_t duration, const QMap<QString, Model *> &modelMap, QStringList candidateModelNames, QString defaultModelName);