Mercurial > hg > svapp
changeset 468:56acd9368532 bqaudioio
Initial work toward switching to bqaudioio library (so as to get I/O, not just O)
| author | Chris Cannam |
|---|---|
| date | Tue, 04 Aug 2015 13:27:42 +0100 |
| parents | 45054b36ddbf |
| children | 4480b031fe38 c6094bca34f4 |
| files | audio/AudioCallbackPlaySource.cpp audio/AudioCallbackPlaySource.h audio/AudioGenerator.cpp audio/AudioGenerator.h audio/ClipMixer.cpp audio/ClipMixer.h audio/ContinuousSynth.cpp audio/ContinuousSynth.h audio/PlaySpeedRangeMapper.cpp audio/PlaySpeedRangeMapper.h audioio/AudioCallbackPlaySource.cpp audioio/AudioCallbackPlaySource.h audioio/AudioCallbackPlayTarget.cpp audioio/AudioCallbackPlayTarget.h audioio/AudioGenerator.cpp audioio/AudioGenerator.h audioio/AudioJACKTarget.cpp audioio/AudioJACKTarget.h audioio/AudioPortAudioTarget.cpp audioio/AudioPortAudioTarget.h audioio/AudioPulseAudioTarget.cpp audioio/AudioPulseAudioTarget.h audioio/AudioTargetFactory.cpp audioio/AudioTargetFactory.h audioio/ClipMixer.cpp audioio/ClipMixer.h audioio/ContinuousSynth.cpp audioio/ContinuousSynth.h audioio/PlaySpeedRangeMapper.cpp audioio/PlaySpeedRangeMapper.h configure.ac framework/MainWindowBase.cpp framework/MainWindowBase.h svapp.pro |
| diffstat | 34 files changed, 3929 insertions(+), 5644 deletions(-) [+] |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/AudioCallbackPlaySource.cpp Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,1902 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Sonic Visualiser + An audio file viewer and annotation editor. + Centre for Digital Music, Queen Mary, University of London. + This file copyright 2006 Chris Cannam and QMUL. + + This program is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public License as + published by the Free Software Foundation; either version 2 of the + License, or (at your option) any later version. See the file + COPYING included with this distribution for more information. +*/ + +#include "AudioCallbackPlaySource.h" + +#include "AudioGenerator.h" + +#include "data/model/Model.h" +#include "base/ViewManagerBase.h" +#include "base/PlayParameterRepository.h" +#include "base/Preferences.h" +#include "data/model/DenseTimeValueModel.h" +#include "data/model/WaveFileModel.h" +#include "data/model/SparseOneDimensionalModel.h" +#include "plugin/RealTimePluginInstance.h" + +#include "bqaudioio/SystemPlaybackTarget.h" + +#include <rubberband/RubberBandStretcher.h> +using namespace RubberBand; + +#include <iostream> +#include <cassert> + +//#define DEBUG_AUDIO_PLAY_SOURCE 1 +//#define DEBUG_AUDIO_PLAY_SOURCE_PLAYING 1 + +static const int DEFAULT_RING_BUFFER_SIZE = 131071; + +AudioCallbackPlaySource::AudioCallbackPlaySource(ViewManagerBase *manager, + QString clientName) : + m_viewManager(manager), + m_audioGenerator(new AudioGenerator()), + m_clientName(clientName.toUtf8().data()), + m_readBuffers(0), + m_writeBuffers(0), + m_readBufferFill(0), + m_writeBufferFill(0), + m_bufferScavenger(1), + m_sourceChannelCount(0), + m_blockSize(1024), + m_sourceSampleRate(0), + m_targetSampleRate(0), + m_playLatency(0), + m_target(0), + m_lastRetrievalTimestamp(0.0), + m_lastRetrievedBlockSize(0), + m_trustworthyTimestamps(true), + m_lastCurrentFrame(0), + m_playing(false), + m_exiting(false), + m_lastModelEndFrame(0), + m_ringBufferSize(DEFAULT_RING_BUFFER_SIZE), + m_outputLeft(0.0), + m_outputRight(0.0), + m_auditioningPlugin(0), + m_auditioningPluginBypassed(false), + m_playStartFrame(0), + m_playStartFramePassed(false), + m_timeStretcher(0), + m_monoStretcher(0), + m_stretchRatio(1.0), + m_stretchMono(false), + m_stretcherInputCount(0), + m_stretcherInputs(0), + m_stretcherInputSizes(0), + m_fillThread(0), + m_converter(0), + m_crapConverter(0), + m_resampleQuality(Preferences::getInstance()->getResampleQuality()) +{ + m_viewManager->setAudioPlaySource(this); + + connect(m_viewManager, SIGNAL(selectionChanged()), + this, SLOT(selectionChanged())); + connect(m_viewManager, SIGNAL(playLoopModeChanged()), + this, SLOT(playLoopModeChanged())); + connect(m_viewManager, SIGNAL(playSelectionModeChanged()), + this, SLOT(playSelectionModeChanged())); + + connect(this, SIGNAL(playStatusChanged(bool)), + m_viewManager, SLOT(playStatusChanged(bool))); + + connect(PlayParameterRepository::getInstance(), + SIGNAL(playParametersChanged(PlayParameters *)), + this, SLOT(playParametersChanged(PlayParameters *))); + + connect(Preferences::getInstance(), + SIGNAL(propertyChanged(PropertyContainer::PropertyName)), + this, SLOT(preferenceChanged(PropertyContainer::PropertyName))); +} + +AudioCallbackPlaySource::~AudioCallbackPlaySource() +{ +#ifdef DEBUG_AUDIO_PLAY_SOURCE + SVDEBUG << "AudioCallbackPlaySource::~AudioCallbackPlaySource entering" << endl; +#endif + m_exiting = true; + + if (m_fillThread) { +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySource dtor: awakening thread" << endl; +#endif + m_condition.wakeAll(); + m_fillThread->wait(); + delete m_fillThread; + } + + clearModels(); + + if (m_readBuffers != m_writeBuffers) { + delete m_readBuffers; + } + + delete m_writeBuffers; + + delete m_audioGenerator; + + for (int i = 0; i < m_stretcherInputCount; ++i) { + delete[] m_stretcherInputs[i]; + } + delete[] m_stretcherInputSizes; + delete[] m_stretcherInputs; + + delete m_timeStretcher; + delete m_monoStretcher; + + m_bufferScavenger.scavenge(true); + m_pluginScavenger.scavenge(true); +#ifdef DEBUG_AUDIO_PLAY_SOURCE + SVDEBUG << "AudioCallbackPlaySource::~AudioCallbackPlaySource finishing" << endl; +#endif +} + +void +AudioCallbackPlaySource::addModel(Model *model) +{ + if (m_models.find(model) != m_models.end()) return; + + bool willPlay = m_audioGenerator->addModel(model); + + m_mutex.lock(); + + m_models.insert(model); + if (model->getEndFrame() > m_lastModelEndFrame) { + m_lastModelEndFrame = model->getEndFrame(); + } + + bool buffersChanged = false, srChanged = false; + + int modelChannels = 1; + DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model); + if (dtvm) modelChannels = dtvm->getChannelCount(); + if (modelChannels > m_sourceChannelCount) { + m_sourceChannelCount = modelChannels; + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySource: Adding model with " << modelChannels << " channels at rate " << model->getSampleRate() << endl; +#endif + + if (m_sourceSampleRate == 0) { + + m_sourceSampleRate = model->getSampleRate(); + srChanged = true; + + } else if (model->getSampleRate() != m_sourceSampleRate) { + + // If this is a dense time-value model and we have no other, we + // can just switch to this model's sample rate + + if (dtvm) { + + bool conflicting = false; + + for (std::set<Model *>::const_iterator i = m_models.begin(); + i != m_models.end(); ++i) { + // Only wave file models can be considered conflicting -- + // writable wave file models are derived and we shouldn't + // take their rates into account. Also, don't give any + // particular weight to a file that's already playing at + // the wrong rate anyway + WaveFileModel *wfm = dynamic_cast<WaveFileModel *>(*i); + if (wfm && wfm != dtvm && + wfm->getSampleRate() != model->getSampleRate() && + wfm->getSampleRate() == m_sourceSampleRate) { + SVDEBUG << "AudioCallbackPlaySource::addModel: Conflicting wave file model " << *i << " found" << endl; + conflicting = true; + break; + } + } + + if (conflicting) { + + SVDEBUG << "AudioCallbackPlaySource::addModel: ERROR: " + << "New model sample rate does not match" << endl + << "existing model(s) (new " << model->getSampleRate() + << " vs " << m_sourceSampleRate + << "), playback will be wrong" + << endl; + + emit sampleRateMismatch(model->getSampleRate(), + m_sourceSampleRate, + false); + } else { + m_sourceSampleRate = model->getSampleRate(); + srChanged = true; + } + } + } + + if (!m_writeBuffers || (int)m_writeBuffers->size() < getTargetChannelCount()) { + clearRingBuffers(true, getTargetChannelCount()); + buffersChanged = true; + } else { + if (willPlay) clearRingBuffers(true); + } + + if (buffersChanged || srChanged) { + if (m_converter) { + src_delete(m_converter); + src_delete(m_crapConverter); + m_converter = 0; + m_crapConverter = 0; + } + } + + rebuildRangeLists(); + + m_mutex.unlock(); + + m_audioGenerator->setTargetChannelCount(getTargetChannelCount()); + + if (!m_fillThread) { + m_fillThread = new FillThread(*this); + m_fillThread->start(); + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySource::addModel: now have " << m_models.size() << " model(s) -- emitting modelReplaced" << endl; +#endif + + if (buffersChanged || srChanged) { + emit modelReplaced(); + } + + 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; +#endif + + m_condition.wakeAll(); +} + +void +AudioCallbackPlaySource::modelChangedWithin(sv_frame_t +#ifdef DEBUG_AUDIO_PLAY_SOURCE + startFrame +#endif + , sv_frame_t endFrame) +{ +#ifdef DEBUG_AUDIO_PLAY_SOURCE + SVDEBUG << "AudioCallbackPlaySource::modelChangedWithin(" << startFrame << "," << endFrame << ")" << endl; +#endif + if (endFrame > m_lastModelEndFrame) { + m_lastModelEndFrame = endFrame; + rebuildRangeLists(); + } +} + +void +AudioCallbackPlaySource::removeModel(Model *model) +{ + m_mutex.lock(); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySource::removeModel(" << model << ")" << endl; +#endif + + disconnect(model, SIGNAL(modelChangedWithin(sv_frame_t, sv_frame_t)), + this, SLOT(modelChangedWithin(sv_frame_t, sv_frame_t))); + + m_models.erase(model); + + if (m_models.empty()) { + if (m_converter) { + src_delete(m_converter); + src_delete(m_crapConverter); + m_converter = 0; + m_crapConverter = 0; + } + m_sourceSampleRate = 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 + cout << "AudioCallbackPlaySource::removeModel(" << model << "): checking end frame on model " << *i << endl; +#endif + if ((*i)->getEndFrame() > lastEnd) { + lastEnd = (*i)->getEndFrame(); + } +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "(done, lastEnd now " << lastEnd << ")" << endl; +#endif + } + m_lastModelEndFrame = lastEnd; + + m_audioGenerator->removeModel(model); + + m_mutex.unlock(); + + clearRingBuffers(); +} + +void +AudioCallbackPlaySource::clearModels() +{ + m_mutex.lock(); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySource::clearModels()" << endl; +#endif + + m_models.clear(); + + if (m_converter) { + src_delete(m_converter); + src_delete(m_crapConverter); + m_converter = 0; + m_crapConverter = 0; + } + + m_lastModelEndFrame = 0; + + m_sourceSampleRate = 0; + + m_mutex.unlock(); + + m_audioGenerator->clearModels(); + + clearRingBuffers(); +} + +void +AudioCallbackPlaySource::clearRingBuffers(bool haveLock, int count) +{ + if (!haveLock) m_mutex.lock(); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cerr << "clearRingBuffers" << endl; +#endif + + rebuildRangeLists(); + + if (count == 0) { + if (m_writeBuffers) count = int(m_writeBuffers->size()); + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cerr << "current playing frame = " << getCurrentPlayingFrame() << endl; + + cerr << "write buffer fill (before) = " << m_writeBufferFill << endl; +#endif + + m_writeBufferFill = getCurrentBufferedFrame(); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cerr << "current buffered frame = " << m_writeBufferFill << endl; +#endif + + if (m_readBuffers != m_writeBuffers) { + delete m_writeBuffers; + } + + m_writeBuffers = new RingBufferVector; + + for (int i = 0; i < count; ++i) { + m_writeBuffers->push_back(new RingBuffer<float>(m_ringBufferSize)); + } + + m_audioGenerator->reset(); + +// cout << "AudioCallbackPlaySource::clearRingBuffers: Created " +// << count << " write buffers" << endl; + + if (!haveLock) { + m_mutex.unlock(); + } +} + +void +AudioCallbackPlaySource::play(sv_frame_t startFrame) +{ + if (!m_sourceSampleRate) { + cerr << "AudioCallbackPlaySource::play: No source sample rate available, not playing" << endl; + return; + } + + if (m_viewManager->getPlaySelectionMode() && + !m_viewManager->getSelections().empty()) { + + SVDEBUG << "AudioCallbackPlaySource::play: constraining frame " << startFrame << " to selection = "; + + startFrame = m_viewManager->constrainFrameToSelection(startFrame); + + SVDEBUG << startFrame << endl; + + } else { + if (startFrame < 0) { + startFrame = 0; + } + if (startFrame >= m_lastModelEndFrame) { + startFrame = 0; + } + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cerr << "play(" << startFrame << ") -> playback model "; +#endif + + startFrame = m_viewManager->alignReferenceToPlaybackFrame(startFrame); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cerr << startFrame << endl; +#endif + + // The fill thread will automatically empty its buffers before + // starting again if we have not so far been playing, but not if + // we're just re-seeking. + // NO -- we can end up playing some first -- always reset here + + m_mutex.lock(); + + if (m_timeStretcher) { + m_timeStretcher->reset(); + } + if (m_monoStretcher) { + m_monoStretcher->reset(); + } + + m_readBufferFill = m_writeBufferFill = startFrame; + if (m_readBuffers) { + for (int c = 0; c < getTargetChannelCount(); ++c) { + RingBuffer<float> *rb = getReadRingBuffer(c); +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cerr << "reset ring buffer for channel " << c << endl; +#endif + if (rb) rb->reset(); + } + } + if (m_converter) src_reset(m_converter); + if (m_crapConverter) src_reset(m_crapConverter); + + m_mutex.unlock(); + + m_audioGenerator->reset(); + + m_playStartFrame = startFrame; + m_playStartFramePassed = false; + m_playStartedAt = RealTime::zeroTime; + if (m_target) { + m_playStartedAt = RealTime::fromSeconds(m_target->getCurrentTime()); + } + + bool changed = !m_playing; + m_lastRetrievalTimestamp = 0; + m_lastCurrentFrame = 0; + m_playing = true; + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySource::play: awakening thread" << endl; +#endif + + m_condition.wakeAll(); + if (changed) { + emit playStatusChanged(m_playing); + emit activity(tr("Play from %1").arg + (RealTime::frame2RealTime + (m_playStartFrame, m_sourceSampleRate).toText().c_str())); + } +} + +void +AudioCallbackPlaySource::stop() +{ +#ifdef DEBUG_AUDIO_PLAY_SOURCE + SVDEBUG << "AudioCallbackPlaySource::stop()" << endl; +#endif + bool changed = m_playing; + m_playing = false; + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySource::stop: awakening thread" << endl; +#endif + + m_condition.wakeAll(); + m_lastRetrievalTimestamp = 0; + if (changed) { + emit playStatusChanged(m_playing); + emit activity(tr("Stop at %1").arg + (RealTime::frame2RealTime + (m_lastCurrentFrame, m_sourceSampleRate).toText().c_str())); + } + m_lastCurrentFrame = 0; +} + +void +AudioCallbackPlaySource::selectionChanged() +{ + if (m_viewManager->getPlaySelectionMode()) { + clearRingBuffers(); + } +} + +void +AudioCallbackPlaySource::playLoopModeChanged() +{ + clearRingBuffers(); +} + +void +AudioCallbackPlaySource::playSelectionModeChanged() +{ + if (!m_viewManager->getSelections().empty()) { + clearRingBuffers(); + } +} + +void +AudioCallbackPlaySource::playParametersChanged(PlayParameters *) +{ + clearRingBuffers(); +} + +void +AudioCallbackPlaySource::preferenceChanged(PropertyContainer::PropertyName n) +{ + if (n == "Resample Quality") { + setResampleQuality(Preferences::getInstance()->getResampleQuality()); + } +} + +void +AudioCallbackPlaySource::audioProcessingOverload() +{ + cerr << "Audio processing overload!" << endl; + + if (!m_playing) return; + + RealTimePluginInstance *ap = m_auditioningPlugin; + if (ap && !m_auditioningPluginBypassed) { + m_auditioningPluginBypassed = true; + emit audioOverloadPluginDisabled(); + return; + } + + if (m_timeStretcher && + m_timeStretcher->getTimeRatio() < 1.0 && + m_stretcherInputCount > 1 && + m_monoStretcher && !m_stretchMono) { + m_stretchMono = true; + emit audioTimeStretchMultiChannelDisabled(); + return; + } +} + +void +AudioCallbackPlaySource::setSystemPlaybackTarget(breakfastquay::SystemPlaybackTarget *target) +{ + m_target = target; +} + +void +AudioCallbackPlaySource::setSystemPlaybackBlockSize(int size) +{ + cout << "AudioCallbackPlaySource::setTarget: Block size -> " << size << endl; + if (size != 0) { + m_blockSize = size; + } + if (size * 4 > m_ringBufferSize) { + SVDEBUG << "AudioCallbackPlaySource::setTarget: Buffer size " + << size << " > a quarter of ring buffer size " + << m_ringBufferSize << ", calling for more ring buffer" + << endl; + m_ringBufferSize = size * 4; + if (m_writeBuffers && !m_writeBuffers->empty()) { + clearRingBuffers(); + } + } +} + +int +AudioCallbackPlaySource::getTargetBlockSize() const +{ +// cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << endl; + return int(m_blockSize); +} + +void +AudioCallbackPlaySource::setSystemPlaybackLatency(int latency) +{ + m_playLatency = latency; +} + +sv_frame_t +AudioCallbackPlaySource::getTargetPlayLatency() const +{ + return m_playLatency; +} + +sv_frame_t +AudioCallbackPlaySource::getCurrentPlayingFrame() +{ + // This method attempts to estimate which audio sample frame is + // "currently coming through the speakers". + + sv_samplerate_t targetRate = getTargetSampleRate(); + sv_frame_t latency = m_playLatency; // at target rate + RealTime latency_t = RealTime::zeroTime; + + if (targetRate != 0) { + latency_t = RealTime::frame2RealTime(latency, targetRate); + } + + return getCurrentFrame(latency_t); +} + +sv_frame_t +AudioCallbackPlaySource::getCurrentBufferedFrame() +{ + return getCurrentFrame(RealTime::zeroTime); +} + +sv_frame_t +AudioCallbackPlaySource::getCurrentFrame(RealTime latency_t) +{ + // We resample when filling the ring buffer, and time-stretch when + // draining it. The buffer contains data at the "target rate" and + // the latency provided by the target is also at the target rate. + // Because of the multiple rates involved, we do the actual + // calculation using RealTime instead. + + sv_samplerate_t sourceRate = getSourceSampleRate(); + sv_samplerate_t targetRate = getTargetSampleRate(); + + if (sourceRate == 0 || targetRate == 0) return 0; + + int inbuffer = 0; // at target rate + + for (int c = 0; c < getTargetChannelCount(); ++c) { + RingBuffer<float> *rb = getReadRingBuffer(c); + if (rb) { + int here = rb->getReadSpace(); + if (c == 0 || here < inbuffer) inbuffer = here; + } + } + + 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(); + + bool looping = m_viewManager->getPlayLoopMode(); + + RealTime inbuffer_t = RealTime::frame2RealTime(inbuffer, targetRate); + + sv_frame_t stretchlat = 0; + double timeRatio = 1.0; + + if (m_timeStretcher) { + stretchlat = m_timeStretcher->getLatency(); + timeRatio = m_timeStretcher->getTimeRatio(); + } + + RealTime stretchlat_t = RealTime::frame2RealTime(stretchlat, targetRate); + + // When the target has just requested a block from us, the last + // sample it obtained was our buffer fill frame count minus the + // amount of read space (converted back to source sample rate) + // remaining now. That sample is not expected to be played until + // the target's play latency has elapsed. By the time the + // following block is requested, that sample will be at the + // target's play latency minus the last requested block size away + // from being played. + + RealTime sincerequest_t = RealTime::zeroTime; + RealTime lastretrieved_t = RealTime::zeroTime; + + if (m_target && + m_trustworthyTimestamps && + lastRetrievalTimestamp != 0.0) { + + lastretrieved_t = RealTime::frame2RealTime + (lastRetrievedBlockSize, targetRate); + + // calculate number of frames at target rate that have elapsed + // since the end of the last call to getSourceSamples + + if (m_trustworthyTimestamps && !looping) { + + // this adjustment seems to cause more problems when looping + double elapsed = currentTime - lastRetrievalTimestamp; + + if (elapsed > 0.0) { + sincerequest_t = RealTime::fromSeconds(elapsed); + } + } + + } else { + + lastretrieved_t = RealTime::frame2RealTime + (getTargetBlockSize(), targetRate); + } + + RealTime bufferedto_t = RealTime::frame2RealTime(readBufferFill, sourceRate); + + if (timeRatio != 1.0) { + lastretrieved_t = lastretrieved_t / timeRatio; + sincerequest_t = sincerequest_t / timeRatio; + latency_t = latency_t / timeRatio; + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + cerr << "\nbuffered to: " << bufferedto_t << ", in buffer: " << inbuffer_t << ", time ratio " << timeRatio << "\n stretcher latency: " << stretchlat_t << ", device latency: " << latency_t << "\n since request: " << sincerequest_t << ", last retrieved quantity: " << lastretrieved_t << endl; +#endif + + // Normally the range lists should contain at least one item each + // -- if playback is unconstrained, that item should report the + // entire source audio duration. + + if (m_rangeStarts.empty()) { + rebuildRangeLists(); + } + + if (m_rangeStarts.empty()) { + // this code is only used in case of error in rebuildRangeLists + RealTime playing_t = bufferedto_t + - latency_t - stretchlat_t - lastretrieved_t - inbuffer_t + + sincerequest_t; + if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime; + sv_frame_t frame = RealTime::realTime2Frame(playing_t, sourceRate); + return m_viewManager->alignPlaybackFrameToReference(frame); + } + + int inRange = 0; + int index = 0; + + for (int i = 0; i < (int)m_rangeStarts.size(); ++i) { + if (bufferedto_t >= m_rangeStarts[i]) { + inRange = index; + } else { + break; + } + ++index; + } + + if (inRange >= int(m_rangeStarts.size())) { + inRange = int(m_rangeStarts.size())-1; + } + + RealTime playing_t = bufferedto_t; + + playing_t = playing_t + - latency_t - stretchlat_t - lastretrieved_t - inbuffer_t + + sincerequest_t; + + // This rather gross little hack is used to ensure that latency + // compensation doesn't result in the playback pointer appearing + // to start earlier than the actual playback does. It doesn't + // work properly (hence the bail-out in the middle) because if we + // are playing a relatively short looped region, the playing time + // estimated from the buffer fill frame may have wrapped around + // the region boundary and end up being much smaller than the + // theoretical play start frame, perhaps even for the entire + // duration of playback! + + if (!m_playStartFramePassed) { + RealTime playstart_t = RealTime::frame2RealTime(m_playStartFrame, + sourceRate); + if (playing_t < playstart_t) { +// cerr << "playing_t " << playing_t << " < playstart_t " +// << playstart_t << endl; + if (/*!!! sincerequest_t > RealTime::zeroTime && */ + m_playStartedAt + latency_t + stretchlat_t < + RealTime::fromSeconds(currentTime)) { +// cerr << "but we've been playing for long enough that I think we should disregard it (it probably results from loop wrapping)" << endl; + m_playStartFramePassed = true; + } else { + playing_t = playstart_t; + } + } else { + m_playStartFramePassed = true; + } + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + cerr << "playing_t " << playing_t; +#endif + + playing_t = playing_t - m_rangeStarts[inRange]; + +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + cerr << " as offset into range " << inRange << " (start =" << m_rangeStarts[inRange] << " duration =" << m_rangeDurations[inRange] << ") = " << playing_t << endl; +#endif + + while (playing_t < RealTime::zeroTime) { + + if (inRange == 0) { + if (looping) { + inRange = int(m_rangeStarts.size()) - 1; + } else { + break; + } + } else { + --inRange; + } + + playing_t = playing_t + m_rangeDurations[inRange]; + } + + playing_t = playing_t + m_rangeStarts[inRange]; + +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + cerr << " playing time: " << playing_t << endl; +#endif + + if (!looping) { + if (inRange == (int)m_rangeStarts.size()-1 && + playing_t >= m_rangeStarts[inRange] + m_rangeDurations[inRange]) { +cerr << "Not looping, inRange " << inRange << " == rangeStarts.size()-1, playing_t " << playing_t << " >= m_rangeStarts[inRange] " << m_rangeStarts[inRange] << " + m_rangeDurations[inRange] " << m_rangeDurations[inRange] << " -- stopping" << endl; + stop(); + } + } + + if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime; + + sv_frame_t frame = RealTime::realTime2Frame(playing_t, sourceRate); + + if (m_lastCurrentFrame > 0 && !looping) { + if (frame < m_lastCurrentFrame) { + frame = m_lastCurrentFrame; + } + } + + m_lastCurrentFrame = frame; + + return m_viewManager->alignPlaybackFrameToReference(frame); +} + +void +AudioCallbackPlaySource::rebuildRangeLists() +{ + bool constrained = (m_viewManager->getPlaySelectionMode()); + + m_rangeStarts.clear(); + m_rangeDurations.clear(); + + sv_samplerate_t sourceRate = getSourceSampleRate(); + if (sourceRate == 0) return; + + RealTime end = RealTime::frame2RealTime(m_lastModelEndFrame, sourceRate); + if (end == RealTime::zeroTime) return; + + if (!constrained) { + m_rangeStarts.push_back(RealTime::zeroTime); + m_rangeDurations.push_back(end); + return; + } + + MultiSelection::SelectionList selections = m_viewManager->getSelections(); + MultiSelection::SelectionList::const_iterator i; + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + SVDEBUG << "AudioCallbackPlaySource::rebuildRangeLists" << endl; +#endif + + if (!selections.empty()) { + + for (i = selections.begin(); i != selections.end(); ++i) { + + RealTime start = + (RealTime::frame2RealTime + (m_viewManager->alignReferenceToPlaybackFrame(i->getStartFrame()), + sourceRate)); + RealTime duration = + (RealTime::frame2RealTime + (m_viewManager->alignReferenceToPlaybackFrame(i->getEndFrame()) - + m_viewManager->alignReferenceToPlaybackFrame(i->getStartFrame()), + sourceRate)); + + m_rangeStarts.push_back(start); + m_rangeDurations.push_back(duration); + } + } else { + m_rangeStarts.push_back(RealTime::zeroTime); + m_rangeDurations.push_back(end); + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cerr << "Now have " << m_rangeStarts.size() << " play ranges" << endl; +#endif +} + +void +AudioCallbackPlaySource::setOutputLevels(float left, float right) +{ + m_outputLeft = left; + m_outputRight = right; +} + +bool +AudioCallbackPlaySource::getOutputLevels(float &left, float &right) +{ + left = m_outputLeft; + right = m_outputRight; + return true; +} + +void +AudioCallbackPlaySource::setSystemPlaybackSampleRate(int sr) +{ + bool first = (m_targetSampleRate == 0); + + m_targetSampleRate = sr; + initialiseConverter(); + + if (first && (m_stretchRatio != 1.f)) { + // couldn't create a stretcher before because we had no sample + // rate: make one now + setTimeStretch(m_stretchRatio); + } +} + +void +AudioCallbackPlaySource::initialiseConverter() +{ + m_mutex.lock(); + + if (m_converter) { + src_delete(m_converter); + src_delete(m_crapConverter); + m_converter = 0; + m_crapConverter = 0; + } + + if (getSourceSampleRate() != getTargetSampleRate()) { + + int err = 0; + + m_converter = src_new(m_resampleQuality == 2 ? SRC_SINC_BEST_QUALITY : + m_resampleQuality == 1 ? SRC_SINC_MEDIUM_QUALITY : + m_resampleQuality == 0 ? SRC_SINC_FASTEST : + SRC_SINC_MEDIUM_QUALITY, + getTargetChannelCount(), &err); + + if (m_converter) { + m_crapConverter = src_new(SRC_LINEAR, + getTargetChannelCount(), + &err); + } + + if (!m_converter || !m_crapConverter) { + cerr + << "AudioCallbackPlaySource::setModel: ERROR in creating samplerate converter: " + << src_strerror(err) << endl; + + if (m_converter) { + src_delete(m_converter); + m_converter = 0; + } + + if (m_crapConverter) { + src_delete(m_crapConverter); + m_crapConverter = 0; + } + + m_mutex.unlock(); + + emit sampleRateMismatch(getSourceSampleRate(), + getTargetSampleRate(), + false); + } else { + + m_mutex.unlock(); + + emit sampleRateMismatch(getSourceSampleRate(), + getTargetSampleRate(), + true); + } + } else { + m_mutex.unlock(); + } +} + +void +AudioCallbackPlaySource::setResampleQuality(int q) +{ + if (q == m_resampleQuality) return; + m_resampleQuality = q; + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + SVDEBUG << "AudioCallbackPlaySource::setResampleQuality: setting to " + << m_resampleQuality << endl; +#endif + + initialiseConverter(); +} + +void +AudioCallbackPlaySource::setAuditioningEffect(Auditionable *a) +{ + RealTimePluginInstance *plugin = dynamic_cast<RealTimePluginInstance *>(a); + if (a && !plugin) { + cerr << "WARNING: AudioCallbackPlaySource::setAuditioningEffect: auditionable object " << a << " is not a real-time plugin instance" << endl; + } + + m_mutex.lock(); + m_auditioningPlugin = plugin; + m_auditioningPluginBypassed = false; + m_mutex.unlock(); +} + +void +AudioCallbackPlaySource::setSoloModelSet(std::set<Model *> s) +{ + m_audioGenerator->setSoloModelSet(s); + clearRingBuffers(); +} + +void +AudioCallbackPlaySource::clearSoloModelSet() +{ + m_audioGenerator->clearSoloModelSet(); + clearRingBuffers(); +} + +sv_samplerate_t +AudioCallbackPlaySource::getTargetSampleRate() const +{ + if (m_targetSampleRate) return m_targetSampleRate; + else return getSourceSampleRate(); +} + +int +AudioCallbackPlaySource::getSourceChannelCount() const +{ + return m_sourceChannelCount; +} + +int +AudioCallbackPlaySource::getTargetChannelCount() const +{ + if (m_sourceChannelCount < 2) return 2; + return m_sourceChannelCount; +} + +sv_samplerate_t +AudioCallbackPlaySource::getSourceSampleRate() const +{ + return m_sourceSampleRate; +} + +void +AudioCallbackPlaySource::setTimeStretch(double factor) +{ + m_stretchRatio = factor; + + if (!getTargetSampleRate()) return; // have to make our stretcher later + + 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 + (int(getTargetSampleRate()), + m_stretcherInputCount, + RubberBandStretcher::OptionProcessRealTime, + factor); + RubberBandStretcher *monoStretcher = new RubberBandStretcher + (int(getTargetSampleRate()), + 1, + RubberBandStretcher::OptionProcessRealTime, + factor); + m_stretcherInputs = new float *[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]]; + } + m_monoStretcher = monoStretcher; + m_timeStretcher = stretcher; + } + + emit activity(tr("Change time-stretch factor to %1").arg(factor)); +} + +void +AudioCallbackPlaySource::getSourceSamples(int count, float **buffer) +{ + if (!m_playing) { +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + SVDEBUG << "AudioCallbackPlaySource::getSourceSamples: Not playing" << endl; +#endif + for (int ch = 0; ch < getTargetChannelCount(); ++ch) { + for (int i = 0; i < count; ++i) { + buffer[ch][i] = 0.0; + } + } + return; + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + SVDEBUG << "AudioCallbackPlaySource::getSourceSamples: Playing" << endl; +#endif + + // Ensure that all buffers have at least the amount of data we + // need -- else reduce the size of our requests correspondingly + + for (int ch = 0; ch < getTargetChannelCount(); ++ch) { + + RingBuffer<float> *rb = getReadRingBuffer(ch); + + if (!rb) { + cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: " + << "No ring buffer available for channel " << ch + << ", returning no data here" << endl; + count = 0; + break; + } + + int rs = rb->getReadSpace(); + if (rs < count) { +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: " + << "Ring buffer for channel " << ch << " has only " + << rs << " (of " << count << ") samples available (" + << "ring buffer size is " << rb->getSize() << ", write " + << "space " << rb->getWriteSpace() << "), " + << "reducing request size" << endl; +#endif + count = rs; + } + } + + if (count == 0) return; + + RubberBandStretcher *ts = m_timeStretcher; + RubberBandStretcher *ms = m_monoStretcher; + + 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 = 1.0; + } else { + ts->setTimeRatio(m_stretchRatio); + if (ms) ms->setTimeRatio(m_stretchRatio); + if (m_stretchRatio >= 1.0) m_stretchMono = false; + } + } + + int stretchChannels = m_stretcherInputCount; + if (m_stretchMono) { + if (ms) { + ts = ms; + stretchChannels = 1; + } else { + m_stretchMono = false; + } + } + + if (m_target) { + m_lastRetrievedBlockSize = count; + m_lastRetrievalTimestamp = m_target->getCurrentTime(); + } + + if (!ts || ratio == 1.f) { + + int got = 0; + + for (int ch = 0; ch < getTargetChannelCount(); ++ch) { + + RingBuffer<float> *rb = getReadRingBuffer(ch); + + if (rb) { + + // this is marginally more likely to leave our channels in + // sync after a processing failure than just passing "count": + sv_frame_t request = count; + if (ch > 0) request = got; + + 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; +#endif + } + + for (int ch = 0; ch < getTargetChannelCount(); ++ch) { + for (int i = got; i < count; ++i) { + buffer[ch][i] = 0.0; + } + } + } + + applyAuditioningEffect(count, buffer); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySource::getSamples: awakening thread" << endl; +#endif + + m_condition.wakeAll(); + + return; + } + + int channels = getTargetChannelCount(); + sv_frame_t available; + sv_frame_t fedToStretcher = 0; + int warned = 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) { + + sv_frame_t reqd = lrint(double(count - available) / ratio); + reqd = std::max(reqd, sv_frame_t(ts->getSamplesRequired())); + if (reqd == 0) reqd = 1; + + sv_frame_t got = reqd; + +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + cerr << "reqd = " <<reqd << ", channels = " << channels << ", ic = " << m_stretcherInputCount << endl; +#endif + + for (int c = 0; c < channels; ++c) { + if (c >= m_stretcherInputCount) continue; + if (reqd > m_stretcherInputSizes[c]) { + if (c == 0) { + cerr << "WARNING: resizing stretcher input buffer from " << m_stretcherInputSizes[c] << " to " << (reqd * 2) << endl; + } + delete[] m_stretcherInputs[c]; + m_stretcherInputSizes[c] = reqd * 2; + m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]]; + } + } + + for (int c = 0; c < channels; ++c) { + if (c >= m_stretcherInputCount) continue; + RingBuffer<float> *rb = getReadRingBuffer(c); + if (rb) { + sv_frame_t gotHere; + if (stretchChannels == 1 && c > 0) { + gotHere = rb->readAdding(m_stretcherInputs[0], int(got)); + } else { + gotHere = rb->read(m_stretcherInputs[c], int(got)); + } + if (gotHere < got) got = gotHere; + +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + if (c == 0) { + SVDEBUG << "feeding stretcher: got " << gotHere + << ", " << rb->getReadSpace() << " remain" << endl; + } +#endif + + } else { + cerr << "WARNING: No ring buffer available for channel " << c << " in stretcher input block" << endl; + } + } + + if (got < reqd) { + cerr << "WARNING: Read underrun in playback (" + << got << " < " << reqd << ")" << endl; + } + + ts->process(m_stretcherInputs, size_t(got), false); + + fedToStretcher += got; + + if (got == 0) break; + + if (ts->available() == available) { + cerr << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples (warned = " << warned << ")" << endl; + if (++warned == 5) break; + } + } + + ts->retrieve(buffer, size_t(count)); + + for (int c = stretchChannels; c < getTargetChannelCount(); ++c) { + for (int i = 0; i < count; ++i) { + buffer[c][i] = buffer[0][i]; + } + } + + applyAuditioningEffect(count, buffer); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySource::getSamples [stretched]: awakening thread" << endl; +#endif + + m_condition.wakeAll(); + + return; +} + +void +AudioCallbackPlaySource::applyAuditioningEffect(sv_frame_t count, float **buffers) +{ + if (m_auditioningPluginBypassed) return; + RealTimePluginInstance *plugin = m_auditioningPlugin; + if (!plugin) return; + + if ((int)plugin->getAudioInputCount() != getTargetChannelCount()) { +// cerr << "plugin input count " << plugin->getAudioInputCount() +// << " != our channel count " << getTargetChannelCount() +// << endl; + return; + } + if ((int)plugin->getAudioOutputCount() != getTargetChannelCount()) { +// cerr << "plugin output count " << plugin->getAudioOutputCount() +// << " != our channel count " << getTargetChannelCount() +// << endl; + return; + } + if ((int)plugin->getBufferSize() < count) { +// cerr << "plugin buffer size " << plugin->getBufferSize() +// << " < our block size " << count +// << endl; + return; + } + + float **ib = plugin->getAudioInputBuffers(); + float **ob = plugin->getAudioOutputBuffers(); + + for (int c = 0; c < getTargetChannelCount(); ++c) { + for (int i = 0; i < count; ++i) { + ib[c][i] = buffers[c][i]; + } + } + + plugin->run(Vamp::RealTime::zeroTime, int(count)); + + for (int c = 0; c < getTargetChannelCount(); ++c) { + for (int i = 0; i < count; ++i) { + buffers[c][i] = ob[c][i]; + } + } +} + +// Called from fill thread, m_playing true, mutex held +bool +AudioCallbackPlaySource::fillBuffers() +{ + static float *tmp = 0; + static sv_frame_t tmpSize = 0; + + sv_frame_t space = 0; + for (int c = 0; c < getTargetChannelCount(); ++c) { + RingBuffer<float> *wb = getWriteRingBuffer(c); + if (wb) { + sv_frame_t spaceHere = wb->getWriteSpace(); + if (c == 0 || spaceHere < space) space = spaceHere; + } + } + + if (space == 0) { +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySourceFillThread: no space to fill" << endl; +#endif + return false; + } + + sv_frame_t f = m_writeBufferFill; + + bool readWriteEqual = (m_readBuffers == m_writeBuffers); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + if (!readWriteEqual) { + cout << "AudioCallbackPlaySourceFillThread: note read buffers != write buffers" << endl; + } + cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << endl; +#endif + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "buffered to " << f << " already" << endl; +#endif + + bool resample = (getSourceSampleRate() != getTargetSampleRate()); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << (resample ? "" : "not ") << "resampling (source " << getSourceSampleRate() << ", target " << getTargetSampleRate() << ")" << endl; +#endif + + int channels = getTargetChannelCount(); + + sv_frame_t orig = space; + sv_frame_t got = 0; + + static float **bufferPtrs = 0; + static int bufferPtrCount = 0; + + if (bufferPtrCount < channels) { + if (bufferPtrs) delete[] bufferPtrs; + bufferPtrs = new float *[channels]; + bufferPtrCount = channels; + } + + sv_frame_t generatorBlockSize = m_audioGenerator->getBlockSize(); + + if (resample && !m_converter) { + static bool warned = false; + if (!warned) { + cerr << "WARNING: sample rates differ, but no converter available!" << endl; + warned = true; + } + } + + if (resample && m_converter) { + + double ratio = + double(getTargetSampleRate()) / double(getSourceSampleRate()); + 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; + + 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 + // the buffer (orig samples for channel 0, orig samples for + // channel 1 etc), and then interleave them into the first + // half of the buffer. Then we resample back into the second + // half (interleaved) and de-interleave the results back to + // the start of the buffer for insertion into the ringbuffers. + // What a faff -- especially as we've already de-interleaved + // the audio data from the source file elsewhere before we + // even reach this point. + + if (tmpSize < channels * work * 2) { + delete[] tmp; + tmp = new float[channels * work * 2]; + tmpSize = channels * work * 2; + } + + float *nonintlv = tmp + channels * work; + float *intlv = tmp; + float *srcout = tmp + channels * work; + + for (int c = 0; c < channels; ++c) { + for (int i = 0; i < orig; ++i) { + nonintlv[channels * i + c] = 0.0f; + } + } + + for (int c = 0; c < channels; ++c) { + bufferPtrs[c] = nonintlv + c * orig; + } + + got = mixModels(f, orig, bufferPtrs); // also modifies f + + // and interleave into first half + for (int c = 0; c < channels; ++c) { + for (int i = 0; i < got; ++i) { + float sample = nonintlv[c * got + i]; + intlv[channels * i + c] = sample; + } + } + + SRC_DATA data; + data.data_in = intlv; + data.data_out = srcout; + data.input_frames = long(got); + data.output_frames = long(work); + data.src_ratio = ratio; + data.end_of_input = 0; + + int err = 0; + + if (m_timeStretcher && m_timeStretcher->getTimeRatio() < 0.4) { +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "Using crappy converter" << endl; +#endif + err = src_process(m_crapConverter, &data); + } else { + err = src_process(m_converter, &data); + } + + sv_frame_t toCopy = sv_frame_t(double(got) * ratio + 0.1); + + if (err) { + cerr + << "AudioCallbackPlaySourceFillThread: ERROR in samplerate conversion: " + << src_strerror(err) << endl; + //!!! Then what? + } else { + got = data.input_frames_used; + toCopy = data.output_frames_gen; +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "Resampled " << got << " frames to " << toCopy << " frames" << endl; +#endif + } + + for (int c = 0; c < channels; ++c) { + for (int i = 0; i < toCopy; ++i) { + tmp[i] = srcout[channels * i + c]; + } + RingBuffer<float> *wb = getWriteRingBuffer(c); + if (wb) wb->write(tmp, int(toCopy)); + } + + m_writeBufferFill = f; + if (readWriteEqual) m_readBufferFill = f; + + } else { + + // space must be a multiple of generatorBlockSize + sv_frame_t reqSpace = space; + space = (reqSpace / generatorBlockSize) * generatorBlockSize; + if (space == 0) { +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "requested fill of " << reqSpace + << " is less than generator block size of " + << generatorBlockSize << ", leaving it" << endl; +#endif + return false; + } + + if (tmpSize < channels * space) { + delete[] tmp; + tmp = new float[channels * space]; + tmpSize = channels * space; + } + + for (int c = 0; c < channels; ++c) { + + bufferPtrs[c] = tmp + c * space; + + for (int i = 0; i < space; ++i) { + tmp[c * space + i] = 0.0f; + } + } + + 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], int(got)); +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "Wrote " << actual << " samples for ch " << c << ", now " + << wb->getReadSpace() << " to read" + << endl; +#endif + if (actual < got) { + cerr << "WARNING: Buffer overrun in channel " << c + << ": wrote " << actual << " of " << got + << " samples" << endl; + } + } + } + + m_writeBufferFill = f; + if (readWriteEqual) m_readBufferFill = f; + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "Read buffer fill is now " << m_readBufferFill << endl; +#endif + + //!!! how do we know when ended? need to mark up a fully-buffered flag and check this if we find the buffers empty in getSourceSamples + } + + return true; +} + +sv_frame_t +AudioCallbackPlaySource::mixModels(sv_frame_t &frame, sv_frame_t count, float **buffers) +{ + 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() && + !m_viewManager->getSelections().empty()); + + static float **chunkBufferPtrs = 0; + static int chunkBufferPtrCount = 0; + int channels = getTargetChannelCount(); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "Selection playback: start " << frame << ", size " << count <<", channels " << channels << endl; +#endif + + if (chunkBufferPtrCount < channels) { + if (chunkBufferPtrs) delete[] chunkBufferPtrs; + chunkBufferPtrs = new float *[channels]; + chunkBufferPtrCount = channels; + } + + for (int c = 0; c < channels; ++c) { + chunkBufferPtrs[c] = buffers[c]; + } + + while (processed < count) { + + chunkSize = count - processed; + nextChunkStart = chunkStart + chunkSize; + selectionSize = 0; + + sv_frame_t fadeIn = 0, fadeOut = 0; + + if (constrained) { + + sv_frame_t rChunkStart = + m_viewManager->alignPlaybackFrameToReference(chunkStart); + + Selection selection = + m_viewManager->getContainingSelection(rChunkStart, true); + + if (selection.isEmpty()) { + if (looping) { + selection = *m_viewManager->getSelections().begin(); + chunkStart = m_viewManager->alignReferenceToPlaybackFrame + (selection.getStartFrame()); + fadeIn = 50; + } + } + + if (selection.isEmpty()) { + + chunkSize = 0; + nextChunkStart = chunkStart; + + } else { + + sv_frame_t sf = m_viewManager->alignReferenceToPlaybackFrame + (selection.getStartFrame()); + sv_frame_t ef = m_viewManager->alignReferenceToPlaybackFrame + (selection.getEndFrame()); + + selectionSize = ef - sf; + + if (chunkStart < sf) { + chunkStart = sf; + fadeIn = 50; + } + + nextChunkStart = chunkStart + chunkSize; + + if (nextChunkStart >= ef) { + nextChunkStart = ef; + fadeOut = 50; + } + + chunkSize = nextChunkStart - chunkStart; + } + + } else if (looping && m_lastModelEndFrame > 0) { + + if (chunkStart >= m_lastModelEndFrame) { + chunkStart = 0; + } + if (chunkSize > m_lastModelEndFrame - chunkStart) { + chunkSize = m_lastModelEndFrame - chunkStart; + } + nextChunkStart = chunkStart + chunkSize; + } + +// cout << "chunkStart " << chunkStart << ", chunkSize " << chunkSize << ", nextChunkStart " << nextChunkStart << ", frame " << frame << ", count " << count << ", processed " << processed << endl; + + if (!chunkSize) { +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "Ending selection playback at " << nextChunkStart << endl; +#endif + // We need to maintain full buffers so that the other + // thread can tell where it's got to in the playback -- so + // return the full amount here + frame = frame + count; + return count; + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "Selection playback: chunk at " << chunkStart << " -> " << nextChunkStart << " (size " << chunkSize << ")" << endl; +#endif + + if (selectionSize < 100) { + fadeIn = 0; + fadeOut = 0; + } else if (selectionSize < 300) { + if (fadeIn > 0) fadeIn = 10; + if (fadeOut > 0) fadeOut = 10; + } + + if (fadeIn > 0) { + if (processed * 2 < fadeIn) { + fadeIn = processed * 2; + } + } + + if (fadeOut > 0) { + if ((count - processed - chunkSize) * 2 < fadeOut) { + fadeOut = (count - processed - chunkSize) * 2; + } + } + + for (std::set<Model *>::iterator mi = m_models.begin(); + mi != m_models.end(); ++mi) { + + (void) m_audioGenerator->mixModel(*mi, chunkStart, + chunkSize, chunkBufferPtrs, + fadeIn, fadeOut); + } + + for (int c = 0; c < channels; ++c) { + chunkBufferPtrs[c] += chunkSize; + } + + processed += chunkSize; + chunkStart = nextChunkStart; + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "Returning selection playback " << processed << " frames to " << nextChunkStart << endl; +#endif + + frame = nextChunkStart; + return processed; +} + +void +AudioCallbackPlaySource::unifyRingBuffers() +{ + if (m_readBuffers == m_writeBuffers) return; + + // only unify if there will be something to read + for (int c = 0; c < getTargetChannelCount(); ++c) { + RingBuffer<float> *wb = getWriteRingBuffer(c); + if (wb) { + if (wb->getReadSpace() < m_blockSize * 2) { + if ((m_writeBufferFill + m_blockSize * 2) < + m_lastModelEndFrame) { + // OK, we don't have enough and there's more to + // read -- don't unify until we can do better +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + SVDEBUG << "AudioCallbackPlaySource::unifyRingBuffers: Not unifying: write buffer has less (" << wb->getReadSpace() << ") than " << m_blockSize*2 << " to read and write buffer fill (" << m_writeBufferFill << ") is not close to end frame (" << m_lastModelEndFrame << ")" << endl; +#endif + return; + } + } + break; + } + } + + sv_frame_t rf = m_readBufferFill; + RingBuffer<float> *rb = getReadRingBuffer(0); + if (rb) { + int rs = rb->getReadSpace(); + //!!! incorrect when in non-contiguous selection, see comments elsewhere +// cout << "rs = " << rs << endl; + if (rs < rf) rf -= rs; + else rf = 0; + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + SVDEBUG << "AudioCallbackPlaySource::unifyRingBuffers: m_readBufferFill = " << m_readBufferFill << ", rf = " << rf << ", m_writeBufferFill = " << m_writeBufferFill << endl; +#endif + + 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) { + if (c == 0) { + + int wrs = wb->getReadSpace(); +// cout << "wrs = " << wrs << endl; + + if (wrs < wf) wf -= wrs; + else wf = 0; +// cout << "wf = " << wf << endl; + + if (wf < rf) skip = rf - wf; + if (skip == 0) break; + } + +// cout << "skipping " << skip << endl; + wb->skip(int(skip)); + } + } + + m_bufferScavenger.claim(m_readBuffers); + m_readBuffers = m_writeBuffers; + m_readBufferFill = m_writeBufferFill; +#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING + cerr << "unified" << endl; +#endif +} + +void +AudioCallbackPlaySource::FillThread::run() +{ + AudioCallbackPlaySource &s(m_source); + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySourceFillThread starting" << endl; +#endif + + s.m_mutex.lock(); + + bool previouslyPlaying = s.m_playing; + bool work = false; + + while (!s.m_exiting) { + + s.unifyRingBuffers(); + s.m_bufferScavenger.scavenge(); + s.m_pluginScavenger.scavenge(); + + if (work && s.m_playing && s.getSourceSampleRate()) { + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySourceFillThread: not waiting" << endl; +#endif + + s.m_mutex.unlock(); + s.m_mutex.lock(); + + } else { + + double ms = 100; + if (s.getSourceSampleRate() > 0) { + ms = double(s.m_ringBufferSize) / s.getSourceSampleRate() * 1000.0; + } + + if (s.m_playing) ms /= 10; + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + if (!s.m_playing) cout << endl; + cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms << "ms..." << endl; +#endif + + s.m_condition.wait(&s.m_mutex, int(ms)); + } + +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySourceFillThread: awoken" << endl; +#endif + + work = false; + + if (!s.getSourceSampleRate()) { +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySourceFillThread: source sample rate is zero" << endl; +#endif + continue; + } + + bool playing = s.m_playing; + + if (playing && !previouslyPlaying) { +#ifdef DEBUG_AUDIO_PLAY_SOURCE + cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << endl; +#endif + for (int c = 0; c < s.getTargetChannelCount(); ++c) { + RingBuffer<float> *rb = s.getReadRingBuffer(c); + if (rb) rb->reset(); + } + } + previouslyPlaying = playing; + + work = s.fillBuffers(); + } + + s.m_mutex.unlock(); +} +
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/AudioCallbackPlaySource.h Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,407 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Sonic Visualiser + An audio file viewer and annotation editor. + Centre for Digital Music, Queen Mary, University of London. + This file copyright 2006 Chris Cannam and QMUL. + + This program is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public License as + published by the Free Software Foundation; either version 2 of the + License, or (at your option) any later version. See the file + COPYING included with this distribution for more information. +*/ + +#ifndef _AUDIO_CALLBACK_PLAY_SOURCE_H_ +#define _AUDIO_CALLBACK_PLAY_SOURCE_H_ + +#include "base/RingBuffer.h" +#include "base/AudioPlaySource.h" +#include "base/PropertyContainer.h" +#include "base/Scavenger.h" + +#include <bqaudioio/ApplicationPlaybackSource.h> + +#include <QObject> +#include <QMutex> +#include <QWaitCondition> + +#include "base/Thread.h" +#include "base/RealTime.h" + +#include <samplerate.h> + +#include <set> +#include <map> + +namespace RubberBand { + class RubberBandStretcher; +} + +class Model; +class ViewManagerBase; +class AudioGenerator; +class PlayParameters; +class RealTimePluginInstance; +class AudioCallbackPlayTarget; + +/** + * AudioCallbackPlaySource manages audio data supply to callback-based + * audio APIs such as JACK or CoreAudio. It maintains one ring buffer + * per channel, filled during playback by a non-realtime thread, and + * provides a method for a realtime thread to pick up the latest + * available sample data from these buffers. + */ +class AudioCallbackPlaySource : public QObject, + public AudioPlaySource, + public breakfastquay::ApplicationPlaybackSource +{ + Q_OBJECT + +public: + AudioCallbackPlaySource(ViewManagerBase *, QString clientName); + virtual ~AudioCallbackPlaySource(); + + /** + * Add a data model to be played from. The source can mix + * playback from a number of sources including dense and sparse + * models. The models must match in sample rate, but they don't + * have to have identical numbers of channels. + */ + virtual void addModel(Model *model); + + /** + * Remove a model. + */ + virtual void removeModel(Model *model); + + /** + * Remove all models. (Silence will ensue.) + */ + virtual void clearModels(); + + /** + * Start making data available in the ring buffers for playback, + * from the given frame. If playback is already under way, reseek + * to the given frame and continue. + */ + virtual void play(sv_frame_t startFrame); + + /** + * Stop playback and ensure that no more data is returned. + */ + virtual void stop(); + + /** + * Return whether playback is currently supposed to be happening. + */ + virtual bool isPlaying() const { return m_playing; } + + /** + * Return the frame number that is currently expected to be coming + * out of the speakers. (i.e. compensating for playback latency.) + */ + virtual sv_frame_t getCurrentPlayingFrame(); + + /** + * Return the last frame that would come out of the speakers if we + * stopped playback right now. + */ + virtual sv_frame_t getCurrentBufferedFrame(); + + /** + * Return the frame at which playback is expected to end (if not looping). + */ + virtual sv_frame_t getPlayEndFrame() { return m_lastModelEndFrame; } + + /** + * Set the playback target. This should be called by the target + * class. + */ + virtual void setSystemPlaybackTarget(breakfastquay::SystemPlaybackTarget *); + + /** + * Set the block size of the target audio device. This should be + * called by the target class. + */ + virtual void setSystemPlaybackBlockSize(int blockSize); + + /** + * Get the block size of the target audio device. This may be an + * estimate or upper bound, if the target has a variable block + * size; the source should behave itself even if this value turns + * out to be inaccurate. + */ + int getTargetBlockSize() const; + + /** + * Set the playback latency of the target audio device, in frames + * at the target sample rate. This is the difference between the + * frame currently "leaving the speakers" and the last frame (or + * highest last frame across all channels) requested via + * getSamples(). The default is zero. + */ + void setSystemPlaybackLatency(int); + + /** + * Get the playback latency of the target audio device. + */ + sv_frame_t getTargetPlayLatency() const; + + /** + * Specify that the target audio device has a fixed sample rate + * (i.e. cannot accommodate arbitrary sample rates based on the + * source). If the target sets this to something other than the + * source sample rate, this class will resample automatically to + * fit. + */ + void setSystemPlaybackSampleRate(int); + + /** + * Return the sample rate set by the target audio device (or the + * source sample rate if the target hasn't set one). + */ + virtual sv_samplerate_t getTargetSampleRate() const; + + /** + * Set the current output levels for metering (for call from the + * target) + */ + void setOutputLevels(float left, float right); + + /** + * Return the current (or thereabouts) output levels in the range + * 0.0 -> 1.0, for metering purposes. + */ + virtual bool getOutputLevels(float &left, float &right); + + /** + * Get the number of channels of audio that in the source models. + * This may safely be called from a realtime thread. Returns 0 if + * there is no source yet available. + */ + int getSourceChannelCount() const; + + /** + * Get the number of channels of audio that will be provided + * to the play target. This may be more than the source channel + * count: for example, a mono source will provide 2 channels + * after pan. + * This may safely be called from a realtime thread. Returns 0 if + * there is no source yet available. + */ + int getTargetChannelCount() const; + + /** + * ApplicationPlaybackSource equivalent of the above. + */ + virtual int getApplicationChannelCount() const { + return getTargetChannelCount(); + } + + /** + * Get the actual sample rate of the source material. This may + * safely be called from a realtime thread. Returns 0 if there is + * no source yet available. + */ + virtual sv_samplerate_t getSourceSampleRate() const; + + /** + * ApplicationPlaybackSource equivalent of the above. + */ + virtual int getApplicationSampleRate() const { + return int(round(getSourceSampleRate())); + } + + /** + * 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. + */ + virtual void getSourceSamples(int count, float **buffer); + + /** + * Set the time stretcher factor (i.e. playback speed). + */ + void setTimeStretch(double factor); + + /** + * Set the resampler quality, 0 - 2 where 0 is fastest and 2 is + * highest quality. + */ + void setResampleQuality(int q); + + /** + * Set a single real-time plugin as a processing effect for + * auditioning during playback. + * + * The plugin must have been initialised with + * getTargetChannelCount() channels and a getTargetBlockSize() + * sample frame processing block size. + * + * This playback source takes ownership of the plugin, which will + * be deleted at some point after the following call to + * setAuditioningEffect (depending on real-time constraints). + * + * Pass a null pointer to remove the current auditioning plugin, + * if any. + */ + void setAuditioningEffect(Auditionable *plugin); + + /** + * Specify that only the given set of models should be played. + */ + void setSoloModelSet(std::set<Model *>s); + + /** + * Specify that all models should be played as normal (if not + * muted). + */ + void clearSoloModelSet(); + + std::string getClientName() const { return m_clientName; } + +signals: + void modelReplaced(); + + void playStatusChanged(bool isPlaying); + + void sampleRateMismatch(sv_samplerate_t requested, + sv_samplerate_t available, + bool willResample); + + void audioOverloadPluginDisabled(); + void audioTimeStretchMultiChannelDisabled(); + + void activity(QString); + +public slots: + void audioProcessingOverload(); + +protected slots: + void selectionChanged(); + void playLoopModeChanged(); + void playSelectionModeChanged(); + void playParametersChanged(PlayParameters *); + void preferenceChanged(PropertyContainer::PropertyName); + void modelChangedWithin(sv_frame_t startFrame, sv_frame_t endFrame); + +protected: + ViewManagerBase *m_viewManager; + AudioGenerator *m_audioGenerator; + std::string m_clientName; + + class RingBufferVector : public std::vector<RingBuffer<float> *> { + public: + virtual ~RingBufferVector() { + while (!empty()) { + delete *begin(); + erase(begin()); + } + } + }; + + std::set<Model *> m_models; + RingBufferVector *m_readBuffers; + RingBufferVector *m_writeBuffers; + sv_frame_t m_readBufferFill; + sv_frame_t m_writeBufferFill; + Scavenger<RingBufferVector> m_bufferScavenger; + int m_sourceChannelCount; + sv_frame_t m_blockSize; + sv_samplerate_t m_sourceSampleRate; + sv_samplerate_t m_targetSampleRate; + sv_frame_t m_playLatency; + breakfastquay::SystemPlaybackTarget *m_target; + double m_lastRetrievalTimestamp; + sv_frame_t m_lastRetrievedBlockSize; + bool m_trustworthyTimestamps; + sv_frame_t m_lastCurrentFrame; + bool m_playing; + bool m_exiting; + sv_frame_t m_lastModelEndFrame; + int m_ringBufferSize; + float m_outputLeft; + float m_outputRight; + RealTimePluginInstance *m_auditioningPlugin; + bool m_auditioningPluginBypassed; + Scavenger<RealTimePluginInstance> m_pluginScavenger; + sv_frame_t m_playStartFrame; + bool m_playStartFramePassed; + RealTime m_playStartedAt; + + RingBuffer<float> *getWriteRingBuffer(int c) { + if (m_writeBuffers && c < (int)m_writeBuffers->size()) { + return (*m_writeBuffers)[c]; + } else { + return 0; + } + } + + RingBuffer<float> *getReadRingBuffer(int c) { + RingBufferVector *rb = m_readBuffers; + if (rb && c < (int)rb->size()) { + return (*rb)[c]; + } else { + return 0; + } + } + + void clearRingBuffers(bool haveLock = false, int count = 0); + void unifyRingBuffers(); + + RubberBand::RubberBandStretcher *m_timeStretcher; + RubberBand::RubberBandStretcher *m_monoStretcher; + double m_stretchRatio; + bool m_stretchMono; + + int m_stretcherInputCount; + float **m_stretcherInputs; + sv_frame_t *m_stretcherInputSizes; + + // Called from fill thread, m_playing true, mutex held + // Return true if work done + bool fillBuffers(); + + // Called from fillBuffers. Return the number of frames written, + // 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). + sv_frame_t mixModels(sv_frame_t &frame, sv_frame_t count, float **buffers); + + // Called from getSourceSamples. + 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(); + + sv_frame_t getCurrentFrame(RealTime outputLatency); + + class FillThread : public Thread + { + public: + FillThread(AudioCallbackPlaySource &source) : + Thread(Thread::NonRTThread), + m_source(source) { } + + virtual void run(); + + protected: + AudioCallbackPlaySource &m_source; + }; + + QMutex m_mutex; + QWaitCondition m_condition; + FillThread *m_fillThread; + SRC_STATE *m_converter; + SRC_STATE *m_crapConverter; // for use when playing very fast + int m_resampleQuality; + void initialiseConverter(); +}; + +#endif + +
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/AudioGenerator.cpp Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,709 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Sonic Visualiser + An audio file viewer and annotation editor. + Centre for Digital Music, Queen Mary, University of London. + This file copyright 2006 Chris Cannam. + + This program is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public License as + published by the Free Software Foundation; either version 2 of the + License, or (at your option) any later version. See the file + COPYING included with this distribution for more information. +*/ + +#include "AudioGenerator.h" + +#include "base/TempDirectory.h" +#include "base/PlayParameters.h" +#include "base/PlayParameterRepository.h" +#include "base/Pitch.h" +#include "base/Exceptions.h" + +#include "data/model/NoteModel.h" +#include "data/model/FlexiNoteModel.h" +#include "data/model/DenseTimeValueModel.h" +#include "data/model/SparseTimeValueModel.h" +#include "data/model/SparseOneDimensionalModel.h" +#include "data/model/NoteData.h" + +#include "ClipMixer.h" +#include "ContinuousSynth.h" + +#include <iostream> +#include <cmath> + +#include <QDir> +#include <QFile> + +const sv_frame_t +AudioGenerator::m_processingBlockSize = 1024; + +QString +AudioGenerator::m_sampleDir = ""; + +//#define DEBUG_AUDIO_GENERATOR 1 + +AudioGenerator::AudioGenerator() : + m_sourceSampleRate(0), + m_targetChannelCount(1), + m_waveType(0), + m_soloing(false), + m_channelBuffer(0), + m_channelBufSiz(0), + m_channelBufCount(0) +{ + initialiseSampleDir(); + + connect(PlayParameterRepository::getInstance(), + SIGNAL(playClipIdChanged(const Playable *, QString)), + this, + SLOT(playClipIdChanged(const Playable *, QString))); +} + +AudioGenerator::~AudioGenerator() +{ +#ifdef DEBUG_AUDIO_GENERATOR + SVDEBUG << "AudioGenerator::~AudioGenerator" << endl; +#endif +} + +void +AudioGenerator::initialiseSampleDir() +{ + if (m_sampleDir != "") return; + + try { + m_sampleDir = TempDirectory::getInstance()->getSubDirectoryPath("samples"); + } catch (DirectoryCreationFailed f) { + cerr << "WARNING: AudioGenerator::initialiseSampleDir:" + << " Failed to create temporary sample directory" + << endl; + m_sampleDir = ""; + return; + } + + QDir sampleResourceDir(":/samples", "*.wav"); + + for (unsigned int i = 0; i < sampleResourceDir.count(); ++i) { + + QString fileName(sampleResourceDir[i]); + QFile file(sampleResourceDir.filePath(fileName)); + QString target = QDir(m_sampleDir).filePath(fileName); + + if (!file.copy(target)) { + cerr << "WARNING: AudioGenerator::getSampleDir: " + << "Unable to copy " << fileName + << " into temporary directory \"" + << m_sampleDir << "\"" << endl; + } else { + QFile tf(target); + tf.setPermissions(tf.permissions() | + QFile::WriteOwner | + QFile::WriteUser); + } + } +} + +bool +AudioGenerator::addModel(Model *model) +{ + if (m_sourceSampleRate == 0) { + + m_sourceSampleRate = model->getSampleRate(); + + } else { + + DenseTimeValueModel *dtvm = + dynamic_cast<DenseTimeValueModel *>(model); + + if (dtvm) { + m_sourceSampleRate = model->getSampleRate(); + return true; + } + } + + const Playable *playable = model; + if (!playable || !playable->canPlay()) return 0; + + PlayParameters *parameters = + PlayParameterRepository::getInstance()->getPlayParameters(playable); + + bool willPlay = !parameters->isPlayMuted(); + + if (usesClipMixer(model)) { + ClipMixer *mixer = makeClipMixerFor(model); + if (mixer) { + QMutexLocker locker(&m_mutex); + m_clipMixerMap[model] = mixer; + return willPlay; + } + } + + if (usesContinuousSynth(model)) { + ContinuousSynth *synth = makeSynthFor(model); + if (synth) { + QMutexLocker locker(&m_mutex); + m_continuousSynthMap[model] = synth; + return willPlay; + } + } + + return false; +} + +void +AudioGenerator::playClipIdChanged(const Playable *playable, QString) +{ + const Model *model = dynamic_cast<const Model *>(playable); + if (!model) { + cerr << "WARNING: AudioGenerator::playClipIdChanged: playable " + << playable << " is not a supported model type" + << endl; + return; + } + + if (m_clipMixerMap.find(model) == m_clipMixerMap.end()) return; + + ClipMixer *mixer = makeClipMixerFor(model); + if (mixer) { + QMutexLocker locker(&m_mutex); + m_clipMixerMap[model] = mixer; + } +} + +bool +AudioGenerator::usesClipMixer(const Model *model) +{ + bool clip = + (qobject_cast<const SparseOneDimensionalModel *>(model) || + qobject_cast<const NoteModel *>(model) || + qobject_cast<const FlexiNoteModel *>(model)); + return clip; +} + +bool +AudioGenerator::wantsQuieterClips(const Model *model) +{ + // basically, anything that usually has sustain (like notes) or + // often has multiple sounds at once (like notes) wants to use a + // quieter level than simple click tracks + bool does = + (qobject_cast<const NoteModel *>(model) || + qobject_cast<const FlexiNoteModel *>(model)); + return does; +} + +bool +AudioGenerator::usesContinuousSynth(const Model *model) +{ + bool cont = + (qobject_cast<const SparseTimeValueModel *>(model)); + return cont; +} + +ClipMixer * +AudioGenerator::makeClipMixerFor(const Model *model) +{ + QString clipId; + + const Playable *playable = model; + if (!playable || !playable->canPlay()) return 0; + + PlayParameters *parameters = + PlayParameterRepository::getInstance()->getPlayParameters(playable); + if (parameters) { + clipId = parameters->getPlayClipId(); + } + +#ifdef DEBUG_AUDIO_GENERATOR + std::cerr << "AudioGenerator::makeClipMixerFor(" << model << "): sample id = " << clipId << std::endl; +#endif + + if (clipId == "") { + SVDEBUG << "AudioGenerator::makeClipMixerFor(" << model << "): no sample, skipping" << endl; + return 0; + } + + ClipMixer *mixer = new ClipMixer(m_targetChannelCount, + m_sourceSampleRate, + m_processingBlockSize); + + double clipF0 = Pitch::getFrequencyForPitch(60, 0, 440.0); // required + + QString clipPath = QString("%1/%2.wav").arg(m_sampleDir).arg(clipId); + + double level = wantsQuieterClips(model) ? 0.5 : 1.0; + if (!mixer->loadClipData(clipPath, clipF0, level)) { + delete mixer; + return 0; + } + +#ifdef DEBUG_AUDIO_GENERATOR + std::cerr << "AudioGenerator::makeClipMixerFor(" << model << "): loaded clip " << clipId << std::endl; +#endif + + return mixer; +} + +ContinuousSynth * +AudioGenerator::makeSynthFor(const Model *model) +{ + const Playable *playable = model; + if (!playable || !playable->canPlay()) return 0; + + ContinuousSynth *synth = new ContinuousSynth(m_targetChannelCount, + m_sourceSampleRate, + m_processingBlockSize, + m_waveType); + +#ifdef DEBUG_AUDIO_GENERATOR + std::cerr << "AudioGenerator::makeSynthFor(" << model << "): created synth" << std::endl; +#endif + + return synth; +} + +void +AudioGenerator::removeModel(Model *model) +{ + SparseOneDimensionalModel *sodm = + dynamic_cast<SparseOneDimensionalModel *>(model); + if (!sodm) return; // nothing to do + + QMutexLocker locker(&m_mutex); + + if (m_clipMixerMap.find(sodm) == m_clipMixerMap.end()) return; + + ClipMixer *mixer = m_clipMixerMap[sodm]; + m_clipMixerMap.erase(sodm); + delete mixer; +} + +void +AudioGenerator::clearModels() +{ + QMutexLocker locker(&m_mutex); + + while (!m_clipMixerMap.empty()) { + ClipMixer *mixer = m_clipMixerMap.begin()->second; + m_clipMixerMap.erase(m_clipMixerMap.begin()); + delete mixer; + } +} + +void +AudioGenerator::reset() +{ + QMutexLocker locker(&m_mutex); + +#ifdef DEBUG_AUDIO_GENERATOR + cerr << "AudioGenerator::reset()" << endl; +#endif + + for (ClipMixerMap::iterator i = m_clipMixerMap.begin(); i != m_clipMixerMap.end(); ++i) { + if (i->second) { + i->second->reset(); + } + } + + m_noteOffs.clear(); +} + +void +AudioGenerator::setTargetChannelCount(int targetChannelCount) +{ + if (m_targetChannelCount == targetChannelCount) return; + +// SVDEBUG << "AudioGenerator::setTargetChannelCount(" << targetChannelCount << ")" << endl; + + QMutexLocker locker(&m_mutex); + m_targetChannelCount = targetChannelCount; + + for (ClipMixerMap::iterator i = m_clipMixerMap.begin(); i != m_clipMixerMap.end(); ++i) { + if (i->second) i->second->setChannelCount(targetChannelCount); + } +} + +sv_frame_t +AudioGenerator::getBlockSize() const +{ + return m_processingBlockSize; +} + +void +AudioGenerator::setSoloModelSet(std::set<Model *> s) +{ + QMutexLocker locker(&m_mutex); + + m_soloModelSet = s; + m_soloing = true; +} + +void +AudioGenerator::clearSoloModelSet() +{ + QMutexLocker locker(&m_mutex); + + m_soloModelSet.clear(); + m_soloing = false; +} + +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; + return frameCount; + } + + QMutexLocker locker(&m_mutex); + + Playable *playable = model; + if (!playable || !playable->canPlay()) return frameCount; + + PlayParameters *parameters = + PlayParameterRepository::getInstance()->getPlayParameters(playable); + if (!parameters) return frameCount; + + bool playing = !parameters->isPlayMuted(); + if (!playing) { +#ifdef DEBUG_AUDIO_GENERATOR + cout << "AudioGenerator::mixModel(" << model << "): muted" << endl; +#endif + return frameCount; + } + + if (m_soloing) { + if (m_soloModelSet.find(model) == m_soloModelSet.end()) { +#ifdef DEBUG_AUDIO_GENERATOR + cout << "AudioGenerator::mixModel(" << model << "): not one of the solo'd models" << endl; +#endif + return frameCount; + } + } + + float gain = parameters->getPlayGain(); + float pan = parameters->getPlayPan(); + + DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model); + if (dtvm) { + return mixDenseTimeValueModel(dtvm, startFrame, frameCount, + buffer, gain, pan, fadeIn, fadeOut); + } + + if (usesClipMixer(model)) { + return mixClipModel(model, startFrame, frameCount, + buffer, gain, pan); + } + + if (usesContinuousSynth(model)) { + return mixContinuousSynthModel(model, startFrame, frameCount, + buffer, gain, pan); + } + + std::cerr << "AudioGenerator::mixModel: WARNING: Model " << model << " of type " << model->getTypeName() << " is marked as playable, but I have no mechanism to play it" << std::endl; + + return frameCount; +} + +sv_frame_t +AudioGenerator::mixDenseTimeValueModel(DenseTimeValueModel *dtvm, + sv_frame_t startFrame, sv_frame_t frames, + float **buffer, float gain, float pan, + sv_frame_t fadeIn, sv_frame_t fadeOut) +{ + sv_frame_t maxFrames = frames + std::max(fadeIn, fadeOut); + + int modelChannels = dtvm->getChannelCount(); + + if (m_channelBufSiz < maxFrames || m_channelBufCount < modelChannels) { + + for (int c = 0; c < m_channelBufCount; ++c) { + delete[] m_channelBuffer[c]; + } + + delete[] m_channelBuffer; + m_channelBuffer = new float *[modelChannels]; + + for (int c = 0; c < modelChannels; ++c) { + m_channelBuffer[c] = new float[maxFrames]; + } + + m_channelBufCount = modelChannels; + m_channelBufSiz = maxFrames; + } + + sv_frame_t got = 0; + + if (startFrame >= fadeIn/2) { + got = dtvm->getMultiChannelData(0, modelChannels - 1, + startFrame - fadeIn/2, + frames + fadeOut/2 + fadeIn/2, + m_channelBuffer); + } else { + sv_frame_t missing = fadeIn/2 - startFrame; + + for (int c = 0; c < modelChannels; ++c) { + m_channelBuffer[c] += missing; + } + + if (missing > 0) { + cerr << "note: channelBufSiz = " << m_channelBufSiz + << ", frames + fadeOut/2 = " << frames + fadeOut/2 + << ", startFrame = " << startFrame + << ", missing = " << missing << endl; + } + + got = dtvm->getMultiChannelData(0, modelChannels - 1, + startFrame, + frames + fadeOut/2, + m_channelBuffer); + + for (int c = 0; c < modelChannels; ++c) { + m_channelBuffer[c] -= missing; + } + + got += missing; + } + + for (int c = 0; c < m_targetChannelCount; ++c) { + + int sourceChannel = (c % modelChannels); + +// SVDEBUG << "mixing channel " << c << " from source channel " << sourceChannel << endl; + + float channelGain = gain; + if (pan != 0.0) { + if (c == 0) { + if (pan > 0.0) channelGain *= 1.0f - pan; + } else { + if (pan < 0.0) channelGain *= pan + 1.0f; + } + } + + for (sv_frame_t i = 0; i < fadeIn/2; ++i) { + float *back = buffer[c]; + back -= fadeIn/2; + back[i] += + (channelGain * m_channelBuffer[sourceChannel][i] * float(i)) + / float(fadeIn); + } + + for (sv_frame_t i = 0; i < frames + fadeOut/2; ++i) { + float mult = channelGain; + if (i < fadeIn/2) { + mult = (mult * float(i)) / float(fadeIn); + } + if (i > frames - fadeOut/2) { + mult = (mult * float((frames + fadeOut/2) - i)) / float(fadeOut); + } + float val = m_channelBuffer[sourceChannel][i]; + if (i >= got) val = 0.f; + buffer[c][i] += mult * val; + } + } + + return got; +} + +sv_frame_t +AudioGenerator::mixClipModel(Model *model, + 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 = int(frames / m_processingBlockSize); + + //!!! todo: the below -- it matters + + //!!! hang on -- the fact that the audio callback play source's + //buffer is a multiple of the plugin's buffer size doesn't mean + //that we always get called for a multiple of it here (because it + //also depends on the JACK block size). how should we ensure that + //all models write the same amount in to the mix, and that we + //always have a multiple of the plugin buffer size? I guess this + //class has to be queryable for the plugin buffer size & the + //callback play source has to use that as a multiple for all the + //calls to mixModel + + sv_frame_t got = blocks * m_processingBlockSize; + +#ifdef DEBUG_AUDIO_GENERATOR + cout << "mixModel [clip]: start " << startFrame << ", frames " << frames + << ", blocks " << blocks << ", have " << m_noteOffs.size() + << " note-offs" << endl; +#endif + + ClipMixer::NoteStart on; + ClipMixer::NoteEnd off; + + NoteOffSet ¬eOffs = m_noteOffs[model]; + + float **bufferIndexes = new float *[m_targetChannelCount]; + + for (int i = 0; i < blocks; ++i) { + + sv_frame_t reqStart = startFrame + i * m_processingBlockSize; + + NoteList notes; + NoteExportable *exportable = dynamic_cast<NoteExportable *>(model); + if (exportable) { + notes = exportable->getNotesWithin(reqStart, + reqStart + m_processingBlockSize); + } + + std::vector<ClipMixer::NoteStart> starts; + std::vector<ClipMixer::NoteEnd> ends; + + for (NoteList::const_iterator ni = notes.begin(); + ni != notes.end(); ++ni) { + + sv_frame_t noteFrame = ni->start; + + if (noteFrame < reqStart || + noteFrame >= reqStart + m_processingBlockSize) continue; + + while (noteOffs.begin() != noteOffs.end() && + noteOffs.begin()->frame <= noteFrame) { + + sv_frame_t eventFrame = noteOffs.begin()->frame; + if (eventFrame < reqStart) eventFrame = reqStart; + + off.frameOffset = eventFrame - reqStart; + off.frequency = noteOffs.begin()->frequency; + +#ifdef DEBUG_AUDIO_GENERATOR + cerr << "mixModel [clip]: adding note-off at frame " << eventFrame << " frame offset " << off.frameOffset << " frequency " << off.frequency << endl; +#endif + + ends.push_back(off); + noteOffs.erase(noteOffs.begin()); + } + + on.frameOffset = noteFrame - reqStart; + on.frequency = ni->getFrequency(); + on.level = float(ni->velocity) / 127.0f; + on.pan = pan; + +#ifdef DEBUG_AUDIO_GENERATOR + cout << "mixModel [clip]: adding note at frame " << noteFrame << ", frame offset " << on.frameOffset << " frequency " << on.frequency << ", level " << on.level << endl; +#endif + + starts.push_back(on); + noteOffs.insert + (NoteOff(on.frequency, noteFrame + ni->duration)); + } + + while (noteOffs.begin() != noteOffs.end() && + noteOffs.begin()->frame <= reqStart + m_processingBlockSize) { + + sv_frame_t eventFrame = noteOffs.begin()->frame; + if (eventFrame < reqStart) eventFrame = reqStart; + + off.frameOffset = eventFrame - reqStart; + off.frequency = noteOffs.begin()->frequency; + +#ifdef DEBUG_AUDIO_GENERATOR + cerr << "mixModel [clip]: adding leftover note-off at frame " << eventFrame << " frame offset " << off.frameOffset << " frequency " << off.frequency << endl; +#endif + + ends.push_back(off); + noteOffs.erase(noteOffs.begin()); + } + + for (int c = 0; c < m_targetChannelCount; ++c) { + bufferIndexes[c] = buffer[c] + i * m_processingBlockSize; + } + + clipMixer->mix(bufferIndexes, gain, starts, ends); + } + + delete[] bufferIndexes; + + return got; +} + +sv_frame_t +AudioGenerator::mixContinuousSynthModel(Model *model, + sv_frame_t startFrame, + sv_frame_t frames, + float **buffer, + float gain, + float pan) +{ + ContinuousSynth *synth = m_continuousSynthMap[model]; + if (!synth) return 0; + + // only type we support here at the moment + SparseTimeValueModel *stvm = qobject_cast<SparseTimeValueModel *>(model); + if (stvm->getScaleUnits() != "Hz") return 0; + + int blocks = int(frames / m_processingBlockSize); + + //!!! todo: see comment in mixClipModel + + sv_frame_t got = blocks * m_processingBlockSize; + +#ifdef DEBUG_AUDIO_GENERATOR + cout << "mixModel [synth]: frames " << frames + << ", blocks " << blocks << endl; +#endif + + float **bufferIndexes = new float *[m_targetChannelCount]; + + for (int i = 0; i < blocks; ++i) { + + sv_frame_t reqStart = startFrame + i * m_processingBlockSize; + + for (int c = 0; c < m_targetChannelCount; ++c) { + bufferIndexes[c] = buffer[c] + i * m_processingBlockSize; + } + + SparseTimeValueModel::PointList points = + stvm->getPoints(reqStart, reqStart + m_processingBlockSize); + + // by default, repeat last frequency + float f0 = 0.f; + + // go straight to the last freq that is genuinely in this range + for (SparseTimeValueModel::PointList::const_iterator itr = points.end(); + itr != points.begin(); ) { + --itr; + if (itr->frame >= reqStart && + itr->frame < reqStart + m_processingBlockSize) { + f0 = itr->value; + break; + } + } + + // if we found no such frequency and the next point is further + // away than twice the model resolution, go silent (same + // criterion TimeValueLayer uses for ending a discrete curve + // segment) + if (f0 == 0.f) { + SparseTimeValueModel::PointList nextPoints = + stvm->getNextPoints(reqStart + m_processingBlockSize); + if (nextPoints.empty() || + nextPoints.begin()->frame > reqStart + 2 * stvm->getResolution()) { + f0 = -1.f; + } + } + +// cerr << "f0 = " << f0 << endl; + + synth->mix(bufferIndexes, + gain, + pan, + f0); + } + + delete[] bufferIndexes; + + return got; +} +
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/AudioGenerator.h Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,168 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Sonic Visualiser + An audio file viewer and annotation editor. + Centre for Digital Music, Queen Mary, University of London. + This file copyright 2006 Chris Cannam. + + This program is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public License as + published by the Free Software Foundation; either version 2 of the + License, or (at your option) any later version. See the file + COPYING included with this distribution for more information. +*/ + +#ifndef _AUDIO_GENERATOR_H_ +#define _AUDIO_GENERATOR_H_ + +class Model; +class NoteModel; +class FlexiNoteModel; +class DenseTimeValueModel; +class SparseOneDimensionalModel; +class Playable; +class ClipMixer; +class ContinuousSynth; + +#include <QObject> +#include <QMutex> + +#include <set> +#include <map> +#include <vector> + +#include "base/BaseTypes.h" + +class AudioGenerator : public QObject +{ + Q_OBJECT + +public: + AudioGenerator(); + virtual ~AudioGenerator(); + + /** + * Add a data model to be played from and initialise any necessary + * audio generation code. Returns true if the model will be + * played. The model will be added regardless of the return + * value. + */ + virtual bool addModel(Model *model); + + /** + * Remove a model. + */ + virtual void removeModel(Model *model); + + /** + * Remove all models. + */ + virtual void clearModels(); + + /** + * Reset playback, clearing buffers and the like. + */ + virtual void reset(); + + /** + * Set the target channel count. The buffer parameter to mixModel + * must always point to at least this number of arrays. + */ + virtual void setTargetChannelCount(int channelCount); + + /** + * Return the internal processing block size. The frameCount + * argument to all mixModel calls must be a multiple of this + * value. + */ + virtual sv_frame_t getBlockSize() const; + + /** + * Mix a single model into an output buffer. + */ + 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. + */ + virtual void setSoloModelSet(std::set<Model *>s); + + /** + * Specify that all models should be played as normal (if not + * muted). + */ + virtual void clearSoloModelSet(); + +protected slots: + void playClipIdChanged(const Playable *, QString); + +protected: + sv_samplerate_t m_sourceSampleRate; + int m_targetChannelCount; + int m_waveType; + + bool m_soloing; + std::set<Model *> m_soloModelSet; + + struct NoteOff { + + NoteOff(float _freq, sv_frame_t _frame) : frequency(_freq), frame(_frame) { } + + float frequency; + sv_frame_t frame; + + struct Comparator { + bool operator()(const NoteOff &n1, const NoteOff &n2) const { + return n1.frame < n2.frame; + } + }; + }; + + + typedef std::map<const Model *, ClipMixer *> ClipMixerMap; + + typedef std::multiset<NoteOff, NoteOff::Comparator> NoteOffSet; + typedef std::map<const Model *, NoteOffSet> NoteOffMap; + + typedef std::map<const Model *, ContinuousSynth *> ContinuousSynthMap; + + QMutex m_mutex; + + ClipMixerMap m_clipMixerMap; + NoteOffMap m_noteOffs; + static QString m_sampleDir; + + ContinuousSynthMap m_continuousSynthMap; + + bool usesClipMixer(const Model *); + bool wantsQuieterClips(const Model *); + bool usesContinuousSynth(const Model *); + + ClipMixer *makeClipMixerFor(const Model *model); + ContinuousSynth *makeSynthFor(const Model *model); + + static void initialiseSampleDir(); + + 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 sv_frame_t mixClipModel + (Model *model, sv_frame_t startFrame, sv_frame_t frameCount, + float **buffer, float gain, float pan); + + virtual sv_frame_t mixContinuousSynthModel + (Model *model, sv_frame_t startFrame, sv_frame_t frameCount, + float **buffer, float gain, float pan); + + static const sv_frame_t m_processingBlockSize; + + float **m_channelBuffer; + sv_frame_t m_channelBufSiz; + int m_channelBufCount; +}; + +#endif +
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/ClipMixer.cpp Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,248 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Sonic Visualiser + An audio file viewer and annotation editor. + Centre for Digital Music, Queen Mary, University of London. + This file copyright 2006 Chris Cannam, 2006-2014 QMUL. + + This program is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public License as + published by the Free Software Foundation; either version 2 of the + License, or (at your option) any later version. See the file + COPYING included with this distribution for more information. +*/ + +#include "ClipMixer.h" + +#include <sndfile.h> +#include <cmath> + +#include "base/Debug.h" + +//#define DEBUG_CLIP_MIXER 1 + +ClipMixer::ClipMixer(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize) : + m_channels(channels), + m_sampleRate(sampleRate), + m_blockSize(blockSize), + m_clipData(0), + m_clipLength(0), + m_clipF0(0), + m_clipRate(0) +{ +} + +ClipMixer::~ClipMixer() +{ + if (m_clipData) free(m_clipData); +} + +void +ClipMixer::setChannelCount(int channels) +{ + m_channels = channels; +} + +bool +ClipMixer::loadClipData(QString path, double f0, double level) +{ + if (m_clipData) { + cerr << "ClipMixer::loadClipData: Already have clip loaded" << endl; + return false; + } + + SF_INFO info; + SNDFILE *file; + float *tmpFrames; + sv_frame_t i; + + info.format = 0; + file = sf_open(path.toLocal8Bit().data(), SFM_READ, &info); + if (!file) { + cerr << "ClipMixer::loadClipData: Failed to open file path \"" + << path << "\": " << sf_strerror(file) << endl; + return false; + } + + tmpFrames = (float *)malloc(info.frames * info.channels * sizeof(float)); + if (!tmpFrames) { + cerr << "ClipMixer::loadClipData: malloc(" << info.frames * info.channels * sizeof(float) << ") failed" << endl; + return false; + } + + sf_readf_float(file, tmpFrames, info.frames); + sf_close(file); + + m_clipData = (float *)malloc(info.frames * sizeof(float)); + if (!m_clipData) { + cerr << "ClipMixer::loadClipData: malloc(" << info.frames * sizeof(float) << ") failed" << endl; + free(tmpFrames); + return false; + } + + for (i = 0; i < info.frames; ++i) { + int j; + m_clipData[i] = 0.0f; + for (j = 0; j < info.channels; ++j) { + m_clipData[i] += tmpFrames[i * info.channels + j] * float(level); + } + } + + free(tmpFrames); + + m_clipLength = info.frames; + m_clipF0 = f0; + m_clipRate = info.samplerate; + + return true; +} + +void +ClipMixer::reset() +{ + m_playing.clear(); +} + +double +ClipMixer::getResampleRatioFor(double frequency) +{ + if (!m_clipData || !m_clipRate) return 1.0; + double pitchRatio = m_clipF0 / frequency; + double resampleRatio = m_sampleRate / m_clipRate; + return pitchRatio * resampleRatio; +} + +sv_frame_t +ClipMixer::getResampledClipDuration(double frequency) +{ + return sv_frame_t(ceil(double(m_clipLength) * getResampleRatioFor(frequency))); +} + +void +ClipMixer::mix(float **toBuffers, + float gain, + std::vector<NoteStart> newNotes, + std::vector<NoteEnd> endingNotes) +{ + foreach (NoteStart note, newNotes) { + if (note.frequency > 20 && + note.frequency < 5000) { + m_playing.push_back(note); + } + } + + std::vector<NoteStart> remaining; + + float *levels = new float[m_channels]; + +#ifdef DEBUG_CLIP_MIXER + cerr << "ClipMixer::mix: have " << m_playing.size() << " playing note(s)" + << " and " << endingNotes.size() << " note(s) ending here" + << endl; +#endif + + foreach (NoteStart note, m_playing) { + + for (int c = 0; c < m_channels; ++c) { + levels[c] = note.level * gain; + } + if (note.pan != 0.0 && m_channels == 2) { + levels[0] *= 1.0f - note.pan; + levels[1] *= note.pan + 1.0f; + } + + sv_frame_t start = note.frameOffset; + sv_frame_t durationHere = m_blockSize; + if (start > 0) durationHere = m_blockSize - start; + + bool ending = false; + + foreach (NoteEnd end, endingNotes) { + if (end.frequency == note.frequency && + end.frameOffset >= start && + end.frameOffset <= m_blockSize) { + ending = true; + durationHere = end.frameOffset; + if (start > 0) durationHere = end.frameOffset - start; + break; + } + } + + sv_frame_t clipDuration = getResampledClipDuration(note.frequency); + if (start + clipDuration > 0) { + if (start < 0 && start + clipDuration < durationHere) { + durationHere = start + clipDuration; + } + if (durationHere > 0) { + mixNote(toBuffers, + levels, + note.frequency, + start < 0 ? -start : 0, + start > 0 ? start : 0, + durationHere, + ending); + } + } + + if (!ending) { + NoteStart adjusted = note; + adjusted.frameOffset -= m_blockSize; + remaining.push_back(adjusted); + } + } + + delete[] levels; + + m_playing = remaining; +} + +void +ClipMixer::mixNote(float **toBuffers, + float *levels, + float frequency, + sv_frame_t sourceOffset, + sv_frame_t targetOffset, + sv_frame_t sampleCount, + bool isEnd) +{ + if (!m_clipData) return; + + double ratio = getResampleRatioFor(frequency); + + double releaseTime = 0.01; + sv_frame_t releaseSampleCount = sv_frame_t(round(releaseTime * m_sampleRate)); + if (releaseSampleCount > sampleCount) { + releaseSampleCount = sampleCount; + } + double releaseFraction = 1.0/double(releaseSampleCount); + + for (sv_frame_t i = 0; i < sampleCount; ++i) { + + sv_frame_t s = sourceOffset + i; + + 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" + 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 - double(osi)); + } + + if (isEnd && i + releaseSampleCount > sampleCount) { + value *= releaseFraction * double(sampleCount - i); // linear ramp for release + } + + for (int c = 0; c < m_channels; ++c) { + toBuffers[c][targetOffset + i] += float(levels[c] * value); + } + } +} + +
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/ClipMixer.h Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,94 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Sonic Visualiser + An audio file viewer and annotation editor. + Centre for Digital Music, Queen Mary, University of London. + This file copyright 2006 Chris Cannam, 2006-2014 QMUL. + + This program is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public License as + published by the Free Software Foundation; either version 2 of the + License, or (at your option) any later version. See the file + COPYING included with this distribution for more information. +*/ + +#ifndef CLIP_MIXER_H +#define CLIP_MIXER_H + +#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 + * musician's sense.) This can mix any number of notes of arbitrary + * frequency, so long as they all use the same sample clip. + */ + +class ClipMixer +{ +public: + ClipMixer(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize); + ~ClipMixer(); + + void setChannelCount(int channels); + + /** + * Load a sample clip from a wav file. This can only happen once: + * construct a new ClipMixer if you want a different clip. The + * clip was recorded at a pitch with fundamental frequency clipF0, + * and should be scaled by level (in the range 0-1) when playing + * back. + */ + bool loadClipData(QString clipFilePath, double clipF0, double level); + + void reset(); // discarding any playing notes + + struct NoteStart { + 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 { + sv_frame_t frameOffset; // in current processing block + float frequency; // matching note start + }; + + void mix(float **toBuffers, + float gain, + std::vector<NoteStart> newNotes, + std::vector<NoteEnd> endingNotes); + +private: + int m_channels; + sv_samplerate_t m_sampleRate; + sv_frame_t m_blockSize; + + QString m_clipPath; + + float *m_clipData; + sv_frame_t m_clipLength; + double m_clipF0; + sv_samplerate_t m_clipRate; + + std::vector<NoteStart> m_playing; + + double getResampleRatioFor(double frequency); + sv_frame_t getResampledClipDuration(double frequency); + + void mixNote(float **toBuffers, + float *levels, + float frequency, + sv_frame_t sourceOffset, // within resampled note + sv_frame_t targetOffset, // within target buffer + sv_frame_t sampleCount, + bool isEnd); +}; + + +#endif
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/ContinuousSynth.cpp Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,149 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Sonic Visualiser + An audio file viewer and annotation editor. + Centre for Digital Music, Queen Mary, University of London. + + This program is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public License as + published by the Free Software Foundation; either version 2 of the + License, or (at your option) any later version. See the file + COPYING included with this distribution for more information. +*/ + +#include "ContinuousSynth.h" + +#include "base/Debug.h" +#include "system/System.h" + +#include <cmath> + +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(-1.0), + m_phase(0.0), + m_wavetype(waveType) // 0: 3 sinusoids, 1: 1 sinusoid, 2: sawtooth, 3: square +{ +} + +ContinuousSynth::~ContinuousSynth() +{ +} + +void +ContinuousSynth::reset() +{ + m_phase = 0; +} + +void +ContinuousSynth::mix(float **toBuffers, float gain, float pan, float f0f) +{ + double f0(f0f); + if (f0 == 0.0) f0 = m_prevF0; + + bool wasOn = (m_prevF0 > 0.0); + bool nowOn = (f0 > 0.0); + + if (!nowOn && !wasOn) { + m_phase = 0; + return; + } + + sv_frame_t fadeLength = 100; + + float *levels = new float[m_channels]; + + for (int c = 0; c < m_channels; ++c) { + levels[c] = gain * 0.5f; // scale gain otherwise too loud compared to source + } + if (pan != 0.0 && m_channels == 2) { + levels[0] *= 1.0f - pan; + levels[1] *= pan + 1.0f; + } + +// cerr << "ContinuousSynth::mix: f0 = " << f0 << " (from " << m_prevF0 << "), phase = " << m_phase << endl; + + 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) * double(i)) / double(fadeLength); + } + + double phasor = (fHere * 2 * M_PI) / m_sampleRate; + + m_phase = m_phase + phasor; + + int harmonics = int((m_sampleRate / 4) / fHere - 1); + if (harmonics < 1) harmonics = 1; + + switch (m_wavetype) { + case 1: + harmonics = 1; + break; + case 2: + break; + case 3: + break; + default: + harmonics = 3; + break; + } + + for (int h = 0; h < harmonics; ++h) { + + double v = 0; + double hn = 0; + double hp = 0; + + switch (m_wavetype) { + case 1: // single sinusoid + v = sin(m_phase); + break; + case 2: // sawtooth + if (h != 0) { + hn = h + 1; + hp = m_phase * hn; + v = -(1.0 / M_PI) * sin(hp) / hn; + } else { + v = 0.5; + } + break; + case 3: // square + hn = h*2 + 1; + hp = m_phase * hn; + v = sin(hp) / hn; + break; + default: // 3 sinusoids + hn = h + 1; + hp = m_phase * hn; + v = sin(hp) / hn; + break; + } + + if (!wasOn && i < fadeLength) { + // fade in + v = v * (double(i) / double(fadeLength)); + } else if (!nowOn) { + // fade out + if (i > fadeLength) v = 0; + else v = v * (1.0 - (double(i) / double(fadeLength))); + } + + for (int c = 0; c < m_channels; ++c) { + toBuffers[c][i] += float(levels[c] * v); + } + } + } + + m_prevF0 = f0; + + delete[] levels; +} +
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/ContinuousSynth.h Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,65 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Sonic Visualiser + An audio file viewer and annotation editor. + Centre for Digital Music, Queen Mary, University of London. + + This program is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public License as + published by the Free Software Foundation; either version 2 of the + License, or (at your option) any later version. See the file + COPYING included with this distribution for more information. +*/ + +#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 + * block, or may be switched on or off. + */ + +class ContinuousSynth +{ +public: + ContinuousSynth(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize, int waveType); + ~ContinuousSynth(); + + void setChannelCount(int channels); + + void reset(); + + /** + * Mix in a signal to be heard at the given fundamental + * frequency. Any oscillator state will be maintained between + * process calls so as to provide a continuous sound. The f0 value + * may vary between calls. + * + * Supply f0 equal to 0 if you want to maintain the f0 from the + * previous block (without having to remember what it was). + * + * Supply f0 less than 0 for silence. You should continue to call + * this even when the signal is silent if you want to ensure the + * sound switches on and off cleanly. + */ + void mix(float **toBuffers, + float gain, + float pan, + float f0); + +private: + int m_channels; + sv_samplerate_t m_sampleRate; + sv_frame_t m_blockSize; + + double m_prevF0; + double m_phase; + + int m_wavetype; +}; + +#endif
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/PlaySpeedRangeMapper.cpp Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,101 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Sonic Visualiser + An audio file viewer and annotation editor. + Centre for Digital Music, Queen Mary, University of London. + This file copyright 2006 QMUL. + + This program is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public License as + published by the Free Software Foundation; either version 2 of the + License, or (at your option) any later version. See the file + COPYING included with this distribution for more information. +*/ + +#include "PlaySpeedRangeMapper.h" + +#include <iostream> +#include <cmath> + +// PlaySpeedRangeMapper maps a position in the range [0,120] on to a +// play speed factor on a logarithmic scale in the range 0.125 -> +// 8. This ensures that the desirable speed factors 0.25, 0.5, 1, 2, +// and 4 are all mapped to exact positions (respectively 20, 40, 60, +// 80, 100). + +// Note that the "factor" referred to below is a play speed factor +// (higher = faster, 1.0 = normal speed), the "value" is a percentage +// (higher = faster, 100 = normal speed), and the "position" is an +// integer step on the dial's scale (0-120, 60 = centre). + +PlaySpeedRangeMapper::PlaySpeedRangeMapper() : + m_minpos(0), + m_maxpos(120) +{ +} + +int +PlaySpeedRangeMapper::getPositionForValue(double value) const +{ + // value is percent + double factor = getFactorForValue(value); + int position = getPositionForFactor(factor); + return position; +} + +int +PlaySpeedRangeMapper::getPositionForValueUnclamped(double value) const +{ + // We don't really provide this + return getPositionForValue(value); +} + +double +PlaySpeedRangeMapper::getValueForPosition(int position) const +{ + double factor = getFactorForPosition(position); + double pc = getValueForFactor(factor); + return pc; +} + +double +PlaySpeedRangeMapper::getValueForPositionUnclamped(int position) const +{ + // We don't really provide this + return getValueForPosition(position); +} + +double +PlaySpeedRangeMapper::getValueForFactor(double factor) const +{ + return factor * 100.0; +} + +double +PlaySpeedRangeMapper::getFactorForValue(double value) const +{ + return value / 100.0; +} + +int +PlaySpeedRangeMapper::getPositionForFactor(double factor) const +{ + if (factor == 0) return m_minpos; + int pos = int(lrint((log2(factor) + 3.0) * 20.0)); + if (pos < m_minpos) pos = m_minpos; + if (pos > m_maxpos) pos = m_maxpos; + return pos; +} + +double +PlaySpeedRangeMapper::getFactorForPosition(int position) const +{ + return pow(2.0, double(position) * 0.05 - 3.0); +} + +QString +PlaySpeedRangeMapper::getUnit() const +{ + return "%"; +}
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/PlaySpeedRangeMapper.h Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,49 @@ +/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ + +/* + Sonic Visualiser + An audio file viewer and annotation editor. + Centre for Digital Music, Queen Mary, University of London. + This file copyright 2006 QMUL. + + This program is free software; you can redistribute it and/or + modify it under the terms of the GNU General Public License as + published by the Free Software Foundation; either version 2 of the + License, or (at your option) any later version. See the file + COPYING included with this distribution for more information. +*/ + +#ifndef _PLAY_SPEED_RANGE_MAPPER_H_ +#define _PLAY_SPEED_RANGE_MAPPER_H_ + +#include "base/RangeMapper.h" + +class PlaySpeedRangeMapper : public RangeMapper +{ +public: + PlaySpeedRangeMapper(); + + int getMinPosition() const { return m_minpos; } + int getMaxPosition() const { return m_maxpos; } + + virtual int getPositionForValue(double value) const; + virtual int getPositionForValueUnclamped(double value) const; + + virtual double getValueForPosition(int position) const; + virtual double getValueForPositionUnclamped(int position) const; + + int getPositionForFactor(double factor) const; + double getValueForFactor(double factor) const; + + double getFactorForPosition(int position) const; + double getFactorForValue(double value) const; + + virtual QString getUnit() const; + +protected: + int m_minpos; + int m_maxpos; +}; + + +#endif
--- a/audioio/AudioCallbackPlaySource.cpp Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,1897 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam and QMUL. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#include "AudioCallbackPlaySource.h" - -#include "AudioGenerator.h" - -#include "data/model/Model.h" -#include "base/ViewManagerBase.h" -#include "base/PlayParameterRepository.h" -#include "base/Preferences.h" -#include "data/model/DenseTimeValueModel.h" -#include "data/model/WaveFileModel.h" -#include "data/model/SparseOneDimensionalModel.h" -#include "plugin/RealTimePluginInstance.h" - -#include "AudioCallbackPlayTarget.h" - -#include <rubberband/RubberBandStretcher.h> -using namespace RubberBand; - -#include <iostream> -#include <cassert> - -//#define DEBUG_AUDIO_PLAY_SOURCE 1 -//#define DEBUG_AUDIO_PLAY_SOURCE_PLAYING 1 - -static const int DEFAULT_RING_BUFFER_SIZE = 131071; - -AudioCallbackPlaySource::AudioCallbackPlaySource(ViewManagerBase *manager, - QString clientName) : - m_viewManager(manager), - m_audioGenerator(new AudioGenerator()), - m_clientName(clientName), - m_readBuffers(0), - m_writeBuffers(0), - m_readBufferFill(0), - m_writeBufferFill(0), - m_bufferScavenger(1), - m_sourceChannelCount(0), - m_blockSize(1024), - m_sourceSampleRate(0), - m_targetSampleRate(0), - m_playLatency(0), - m_target(0), - m_lastRetrievalTimestamp(0.0), - m_lastRetrievedBlockSize(0), - m_trustworthyTimestamps(true), - m_lastCurrentFrame(0), - m_playing(false), - m_exiting(false), - m_lastModelEndFrame(0), - m_ringBufferSize(DEFAULT_RING_BUFFER_SIZE), - m_outputLeft(0.0), - m_outputRight(0.0), - m_auditioningPlugin(0), - m_auditioningPluginBypassed(false), - m_playStartFrame(0), - m_playStartFramePassed(false), - m_timeStretcher(0), - m_monoStretcher(0), - m_stretchRatio(1.0), - m_stretchMono(false), - m_stretcherInputCount(0), - m_stretcherInputs(0), - m_stretcherInputSizes(0), - m_fillThread(0), - m_converter(0), - m_crapConverter(0), - m_resampleQuality(Preferences::getInstance()->getResampleQuality()) -{ - m_viewManager->setAudioPlaySource(this); - - connect(m_viewManager, SIGNAL(selectionChanged()), - this, SLOT(selectionChanged())); - connect(m_viewManager, SIGNAL(playLoopModeChanged()), - this, SLOT(playLoopModeChanged())); - connect(m_viewManager, SIGNAL(playSelectionModeChanged()), - this, SLOT(playSelectionModeChanged())); - - connect(this, SIGNAL(playStatusChanged(bool)), - m_viewManager, SLOT(playStatusChanged(bool))); - - connect(PlayParameterRepository::getInstance(), - SIGNAL(playParametersChanged(PlayParameters *)), - this, SLOT(playParametersChanged(PlayParameters *))); - - connect(Preferences::getInstance(), - SIGNAL(propertyChanged(PropertyContainer::PropertyName)), - this, SLOT(preferenceChanged(PropertyContainer::PropertyName))); -} - -AudioCallbackPlaySource::~AudioCallbackPlaySource() -{ -#ifdef DEBUG_AUDIO_PLAY_SOURCE - SVDEBUG << "AudioCallbackPlaySource::~AudioCallbackPlaySource entering" << endl; -#endif - m_exiting = true; - - if (m_fillThread) { -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySource dtor: awakening thread" << endl; -#endif - m_condition.wakeAll(); - m_fillThread->wait(); - delete m_fillThread; - } - - clearModels(); - - if (m_readBuffers != m_writeBuffers) { - delete m_readBuffers; - } - - delete m_writeBuffers; - - delete m_audioGenerator; - - for (int i = 0; i < m_stretcherInputCount; ++i) { - delete[] m_stretcherInputs[i]; - } - delete[] m_stretcherInputSizes; - delete[] m_stretcherInputs; - - delete m_timeStretcher; - delete m_monoStretcher; - - m_bufferScavenger.scavenge(true); - m_pluginScavenger.scavenge(true); -#ifdef DEBUG_AUDIO_PLAY_SOURCE - SVDEBUG << "AudioCallbackPlaySource::~AudioCallbackPlaySource finishing" << endl; -#endif -} - -void -AudioCallbackPlaySource::addModel(Model *model) -{ - if (m_models.find(model) != m_models.end()) return; - - bool willPlay = m_audioGenerator->addModel(model); - - m_mutex.lock(); - - m_models.insert(model); - if (model->getEndFrame() > m_lastModelEndFrame) { - m_lastModelEndFrame = model->getEndFrame(); - } - - bool buffersChanged = false, srChanged = false; - - int modelChannels = 1; - DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model); - if (dtvm) modelChannels = dtvm->getChannelCount(); - if (modelChannels > m_sourceChannelCount) { - m_sourceChannelCount = modelChannels; - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySource: Adding model with " << modelChannels << " channels at rate " << model->getSampleRate() << endl; -#endif - - if (m_sourceSampleRate == 0) { - - m_sourceSampleRate = model->getSampleRate(); - srChanged = true; - - } else if (model->getSampleRate() != m_sourceSampleRate) { - - // If this is a dense time-value model and we have no other, we - // can just switch to this model's sample rate - - if (dtvm) { - - bool conflicting = false; - - for (std::set<Model *>::const_iterator i = m_models.begin(); - i != m_models.end(); ++i) { - // Only wave file models can be considered conflicting -- - // writable wave file models are derived and we shouldn't - // take their rates into account. Also, don't give any - // particular weight to a file that's already playing at - // the wrong rate anyway - WaveFileModel *wfm = dynamic_cast<WaveFileModel *>(*i); - if (wfm && wfm != dtvm && - wfm->getSampleRate() != model->getSampleRate() && - wfm->getSampleRate() == m_sourceSampleRate) { - SVDEBUG << "AudioCallbackPlaySource::addModel: Conflicting wave file model " << *i << " found" << endl; - conflicting = true; - break; - } - } - - if (conflicting) { - - SVDEBUG << "AudioCallbackPlaySource::addModel: ERROR: " - << "New model sample rate does not match" << endl - << "existing model(s) (new " << model->getSampleRate() - << " vs " << m_sourceSampleRate - << "), playback will be wrong" - << endl; - - emit sampleRateMismatch(model->getSampleRate(), - m_sourceSampleRate, - false); - } else { - m_sourceSampleRate = model->getSampleRate(); - srChanged = true; - } - } - } - - if (!m_writeBuffers || (int)m_writeBuffers->size() < getTargetChannelCount()) { - clearRingBuffers(true, getTargetChannelCount()); - buffersChanged = true; - } else { - if (willPlay) clearRingBuffers(true); - } - - if (buffersChanged || srChanged) { - if (m_converter) { - src_delete(m_converter); - src_delete(m_crapConverter); - m_converter = 0; - m_crapConverter = 0; - } - } - - rebuildRangeLists(); - - m_mutex.unlock(); - - m_audioGenerator->setTargetChannelCount(getTargetChannelCount()); - - if (!m_fillThread) { - m_fillThread = new FillThread(*this); - m_fillThread->start(); - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySource::addModel: now have " << m_models.size() << " model(s) -- emitting modelReplaced" << endl; -#endif - - if (buffersChanged || srChanged) { - emit modelReplaced(); - } - - 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; -#endif - - m_condition.wakeAll(); -} - -void -AudioCallbackPlaySource::modelChangedWithin(sv_frame_t -#ifdef DEBUG_AUDIO_PLAY_SOURCE - startFrame -#endif - , sv_frame_t endFrame) -{ -#ifdef DEBUG_AUDIO_PLAY_SOURCE - SVDEBUG << "AudioCallbackPlaySource::modelChangedWithin(" << startFrame << "," << endFrame << ")" << endl; -#endif - if (endFrame > m_lastModelEndFrame) { - m_lastModelEndFrame = endFrame; - rebuildRangeLists(); - } -} - -void -AudioCallbackPlaySource::removeModel(Model *model) -{ - m_mutex.lock(); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySource::removeModel(" << model << ")" << endl; -#endif - - disconnect(model, SIGNAL(modelChangedWithin(sv_frame_t, sv_frame_t)), - this, SLOT(modelChangedWithin(sv_frame_t, sv_frame_t))); - - m_models.erase(model); - - if (m_models.empty()) { - if (m_converter) { - src_delete(m_converter); - src_delete(m_crapConverter); - m_converter = 0; - m_crapConverter = 0; - } - m_sourceSampleRate = 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 - cout << "AudioCallbackPlaySource::removeModel(" << model << "): checking end frame on model " << *i << endl; -#endif - if ((*i)->getEndFrame() > lastEnd) { - lastEnd = (*i)->getEndFrame(); - } -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "(done, lastEnd now " << lastEnd << ")" << endl; -#endif - } - m_lastModelEndFrame = lastEnd; - - m_audioGenerator->removeModel(model); - - m_mutex.unlock(); - - clearRingBuffers(); -} - -void -AudioCallbackPlaySource::clearModels() -{ - m_mutex.lock(); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySource::clearModels()" << endl; -#endif - - m_models.clear(); - - if (m_converter) { - src_delete(m_converter); - src_delete(m_crapConverter); - m_converter = 0; - m_crapConverter = 0; - } - - m_lastModelEndFrame = 0; - - m_sourceSampleRate = 0; - - m_mutex.unlock(); - - m_audioGenerator->clearModels(); - - clearRingBuffers(); -} - -void -AudioCallbackPlaySource::clearRingBuffers(bool haveLock, int count) -{ - if (!haveLock) m_mutex.lock(); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cerr << "clearRingBuffers" << endl; -#endif - - rebuildRangeLists(); - - if (count == 0) { - if (m_writeBuffers) count = int(m_writeBuffers->size()); - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cerr << "current playing frame = " << getCurrentPlayingFrame() << endl; - - cerr << "write buffer fill (before) = " << m_writeBufferFill << endl; -#endif - - m_writeBufferFill = getCurrentBufferedFrame(); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cerr << "current buffered frame = " << m_writeBufferFill << endl; -#endif - - if (m_readBuffers != m_writeBuffers) { - delete m_writeBuffers; - } - - m_writeBuffers = new RingBufferVector; - - for (int i = 0; i < count; ++i) { - m_writeBuffers->push_back(new RingBuffer<float>(m_ringBufferSize)); - } - - m_audioGenerator->reset(); - -// cout << "AudioCallbackPlaySource::clearRingBuffers: Created " -// << count << " write buffers" << endl; - - if (!haveLock) { - m_mutex.unlock(); - } -} - -void -AudioCallbackPlaySource::play(sv_frame_t startFrame) -{ - if (!m_sourceSampleRate) { - cerr << "AudioCallbackPlaySource::play: No source sample rate available, not playing" << endl; - return; - } - - if (m_viewManager->getPlaySelectionMode() && - !m_viewManager->getSelections().empty()) { - - SVDEBUG << "AudioCallbackPlaySource::play: constraining frame " << startFrame << " to selection = "; - - startFrame = m_viewManager->constrainFrameToSelection(startFrame); - - SVDEBUG << startFrame << endl; - - } else { - if (startFrame < 0) { - startFrame = 0; - } - if (startFrame >= m_lastModelEndFrame) { - startFrame = 0; - } - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cerr << "play(" << startFrame << ") -> playback model "; -#endif - - startFrame = m_viewManager->alignReferenceToPlaybackFrame(startFrame); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cerr << startFrame << endl; -#endif - - // The fill thread will automatically empty its buffers before - // starting again if we have not so far been playing, but not if - // we're just re-seeking. - // NO -- we can end up playing some first -- always reset here - - m_mutex.lock(); - - if (m_timeStretcher) { - m_timeStretcher->reset(); - } - if (m_monoStretcher) { - m_monoStretcher->reset(); - } - - m_readBufferFill = m_writeBufferFill = startFrame; - if (m_readBuffers) { - for (int c = 0; c < getTargetChannelCount(); ++c) { - RingBuffer<float> *rb = getReadRingBuffer(c); -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cerr << "reset ring buffer for channel " << c << endl; -#endif - if (rb) rb->reset(); - } - } - if (m_converter) src_reset(m_converter); - if (m_crapConverter) src_reset(m_crapConverter); - - m_mutex.unlock(); - - m_audioGenerator->reset(); - - m_playStartFrame = startFrame; - m_playStartFramePassed = false; - m_playStartedAt = RealTime::zeroTime; - if (m_target) { - m_playStartedAt = RealTime::fromSeconds(m_target->getCurrentTime()); - } - - bool changed = !m_playing; - m_lastRetrievalTimestamp = 0; - m_lastCurrentFrame = 0; - m_playing = true; - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySource::play: awakening thread" << endl; -#endif - - m_condition.wakeAll(); - if (changed) { - emit playStatusChanged(m_playing); - emit activity(tr("Play from %1").arg - (RealTime::frame2RealTime - (m_playStartFrame, m_sourceSampleRate).toText().c_str())); - } -} - -void -AudioCallbackPlaySource::stop() -{ -#ifdef DEBUG_AUDIO_PLAY_SOURCE - SVDEBUG << "AudioCallbackPlaySource::stop()" << endl; -#endif - bool changed = m_playing; - m_playing = false; - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySource::stop: awakening thread" << endl; -#endif - - m_condition.wakeAll(); - m_lastRetrievalTimestamp = 0; - if (changed) { - emit playStatusChanged(m_playing); - emit activity(tr("Stop at %1").arg - (RealTime::frame2RealTime - (m_lastCurrentFrame, m_sourceSampleRate).toText().c_str())); - } - m_lastCurrentFrame = 0; -} - -void -AudioCallbackPlaySource::selectionChanged() -{ - if (m_viewManager->getPlaySelectionMode()) { - clearRingBuffers(); - } -} - -void -AudioCallbackPlaySource::playLoopModeChanged() -{ - clearRingBuffers(); -} - -void -AudioCallbackPlaySource::playSelectionModeChanged() -{ - if (!m_viewManager->getSelections().empty()) { - clearRingBuffers(); - } -} - -void -AudioCallbackPlaySource::playParametersChanged(PlayParameters *) -{ - clearRingBuffers(); -} - -void -AudioCallbackPlaySource::preferenceChanged(PropertyContainer::PropertyName n) -{ - if (n == "Resample Quality") { - setResampleQuality(Preferences::getInstance()->getResampleQuality()); - } -} - -void -AudioCallbackPlaySource::audioProcessingOverload() -{ - cerr << "Audio processing overload!" << endl; - - if (!m_playing) return; - - RealTimePluginInstance *ap = m_auditioningPlugin; - if (ap && !m_auditioningPluginBypassed) { - m_auditioningPluginBypassed = true; - emit audioOverloadPluginDisabled(); - return; - } - - if (m_timeStretcher && - m_timeStretcher->getTimeRatio() < 1.0 && - m_stretcherInputCount > 1 && - m_monoStretcher && !m_stretchMono) { - m_stretchMono = true; - emit audioTimeStretchMultiChannelDisabled(); - return; - } -} - -void -AudioCallbackPlaySource::setTarget(AudioCallbackPlayTarget *target, int size) -{ - m_target = target; - cout << "AudioCallbackPlaySource::setTarget: Block size -> " << size << endl; - if (size != 0) { - m_blockSize = size; - } - if (size * 4 > m_ringBufferSize) { - SVDEBUG << "AudioCallbackPlaySource::setTarget: Buffer size " - << size << " > a quarter of ring buffer size " - << m_ringBufferSize << ", calling for more ring buffer" - << endl; - m_ringBufferSize = size * 4; - if (m_writeBuffers && !m_writeBuffers->empty()) { - clearRingBuffers(); - } - } -} - -int -AudioCallbackPlaySource::getTargetBlockSize() const -{ -// cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << endl; - return int(m_blockSize); -} - -void -AudioCallbackPlaySource::setTargetPlayLatency(sv_frame_t latency) -{ - m_playLatency = latency; -} - -sv_frame_t -AudioCallbackPlaySource::getTargetPlayLatency() const -{ - return m_playLatency; -} - -sv_frame_t -AudioCallbackPlaySource::getCurrentPlayingFrame() -{ - // This method attempts to estimate which audio sample frame is - // "currently coming through the speakers". - - sv_samplerate_t targetRate = getTargetSampleRate(); - sv_frame_t latency = m_playLatency; // at target rate - RealTime latency_t = RealTime::zeroTime; - - if (targetRate != 0) { - latency_t = RealTime::frame2RealTime(latency, targetRate); - } - - return getCurrentFrame(latency_t); -} - -sv_frame_t -AudioCallbackPlaySource::getCurrentBufferedFrame() -{ - return getCurrentFrame(RealTime::zeroTime); -} - -sv_frame_t -AudioCallbackPlaySource::getCurrentFrame(RealTime latency_t) -{ - // We resample when filling the ring buffer, and time-stretch when - // draining it. The buffer contains data at the "target rate" and - // the latency provided by the target is also at the target rate. - // Because of the multiple rates involved, we do the actual - // calculation using RealTime instead. - - sv_samplerate_t sourceRate = getSourceSampleRate(); - sv_samplerate_t targetRate = getTargetSampleRate(); - - if (sourceRate == 0 || targetRate == 0) return 0; - - int inbuffer = 0; // at target rate - - for (int c = 0; c < getTargetChannelCount(); ++c) { - RingBuffer<float> *rb = getReadRingBuffer(c); - if (rb) { - int here = rb->getReadSpace(); - if (c == 0 || here < inbuffer) inbuffer = here; - } - } - - 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(); - - bool looping = m_viewManager->getPlayLoopMode(); - - RealTime inbuffer_t = RealTime::frame2RealTime(inbuffer, targetRate); - - sv_frame_t stretchlat = 0; - double timeRatio = 1.0; - - if (m_timeStretcher) { - stretchlat = m_timeStretcher->getLatency(); - timeRatio = m_timeStretcher->getTimeRatio(); - } - - RealTime stretchlat_t = RealTime::frame2RealTime(stretchlat, targetRate); - - // When the target has just requested a block from us, the last - // sample it obtained was our buffer fill frame count minus the - // amount of read space (converted back to source sample rate) - // remaining now. That sample is not expected to be played until - // the target's play latency has elapsed. By the time the - // following block is requested, that sample will be at the - // target's play latency minus the last requested block size away - // from being played. - - RealTime sincerequest_t = RealTime::zeroTime; - RealTime lastretrieved_t = RealTime::zeroTime; - - if (m_target && - m_trustworthyTimestamps && - lastRetrievalTimestamp != 0.0) { - - lastretrieved_t = RealTime::frame2RealTime - (lastRetrievedBlockSize, targetRate); - - // calculate number of frames at target rate that have elapsed - // since the end of the last call to getSourceSamples - - if (m_trustworthyTimestamps && !looping) { - - // this adjustment seems to cause more problems when looping - double elapsed = currentTime - lastRetrievalTimestamp; - - if (elapsed > 0.0) { - sincerequest_t = RealTime::fromSeconds(elapsed); - } - } - - } else { - - lastretrieved_t = RealTime::frame2RealTime - (getTargetBlockSize(), targetRate); - } - - RealTime bufferedto_t = RealTime::frame2RealTime(readBufferFill, sourceRate); - - if (timeRatio != 1.0) { - lastretrieved_t = lastretrieved_t / timeRatio; - sincerequest_t = sincerequest_t / timeRatio; - latency_t = latency_t / timeRatio; - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - cerr << "\nbuffered to: " << bufferedto_t << ", in buffer: " << inbuffer_t << ", time ratio " << timeRatio << "\n stretcher latency: " << stretchlat_t << ", device latency: " << latency_t << "\n since request: " << sincerequest_t << ", last retrieved quantity: " << lastretrieved_t << endl; -#endif - - // Normally the range lists should contain at least one item each - // -- if playback is unconstrained, that item should report the - // entire source audio duration. - - if (m_rangeStarts.empty()) { - rebuildRangeLists(); - } - - if (m_rangeStarts.empty()) { - // this code is only used in case of error in rebuildRangeLists - RealTime playing_t = bufferedto_t - - latency_t - stretchlat_t - lastretrieved_t - inbuffer_t - + sincerequest_t; - if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime; - sv_frame_t frame = RealTime::realTime2Frame(playing_t, sourceRate); - return m_viewManager->alignPlaybackFrameToReference(frame); - } - - int inRange = 0; - int index = 0; - - for (int i = 0; i < (int)m_rangeStarts.size(); ++i) { - if (bufferedto_t >= m_rangeStarts[i]) { - inRange = index; - } else { - break; - } - ++index; - } - - if (inRange >= int(m_rangeStarts.size())) { - inRange = int(m_rangeStarts.size())-1; - } - - RealTime playing_t = bufferedto_t; - - playing_t = playing_t - - latency_t - stretchlat_t - lastretrieved_t - inbuffer_t - + sincerequest_t; - - // This rather gross little hack is used to ensure that latency - // compensation doesn't result in the playback pointer appearing - // to start earlier than the actual playback does. It doesn't - // work properly (hence the bail-out in the middle) because if we - // are playing a relatively short looped region, the playing time - // estimated from the buffer fill frame may have wrapped around - // the region boundary and end up being much smaller than the - // theoretical play start frame, perhaps even for the entire - // duration of playback! - - if (!m_playStartFramePassed) { - RealTime playstart_t = RealTime::frame2RealTime(m_playStartFrame, - sourceRate); - if (playing_t < playstart_t) { -// cerr << "playing_t " << playing_t << " < playstart_t " -// << playstart_t << endl; - if (/*!!! sincerequest_t > RealTime::zeroTime && */ - m_playStartedAt + latency_t + stretchlat_t < - RealTime::fromSeconds(currentTime)) { -// cerr << "but we've been playing for long enough that I think we should disregard it (it probably results from loop wrapping)" << endl; - m_playStartFramePassed = true; - } else { - playing_t = playstart_t; - } - } else { - m_playStartFramePassed = true; - } - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - cerr << "playing_t " << playing_t; -#endif - - playing_t = playing_t - m_rangeStarts[inRange]; - -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - cerr << " as offset into range " << inRange << " (start =" << m_rangeStarts[inRange] << " duration =" << m_rangeDurations[inRange] << ") = " << playing_t << endl; -#endif - - while (playing_t < RealTime::zeroTime) { - - if (inRange == 0) { - if (looping) { - inRange = int(m_rangeStarts.size()) - 1; - } else { - break; - } - } else { - --inRange; - } - - playing_t = playing_t + m_rangeDurations[inRange]; - } - - playing_t = playing_t + m_rangeStarts[inRange]; - -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - cerr << " playing time: " << playing_t << endl; -#endif - - if (!looping) { - if (inRange == (int)m_rangeStarts.size()-1 && - playing_t >= m_rangeStarts[inRange] + m_rangeDurations[inRange]) { -cerr << "Not looping, inRange " << inRange << " == rangeStarts.size()-1, playing_t " << playing_t << " >= m_rangeStarts[inRange] " << m_rangeStarts[inRange] << " + m_rangeDurations[inRange] " << m_rangeDurations[inRange] << " -- stopping" << endl; - stop(); - } - } - - if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime; - - sv_frame_t frame = RealTime::realTime2Frame(playing_t, sourceRate); - - if (m_lastCurrentFrame > 0 && !looping) { - if (frame < m_lastCurrentFrame) { - frame = m_lastCurrentFrame; - } - } - - m_lastCurrentFrame = frame; - - return m_viewManager->alignPlaybackFrameToReference(frame); -} - -void -AudioCallbackPlaySource::rebuildRangeLists() -{ - bool constrained = (m_viewManager->getPlaySelectionMode()); - - m_rangeStarts.clear(); - m_rangeDurations.clear(); - - sv_samplerate_t sourceRate = getSourceSampleRate(); - if (sourceRate == 0) return; - - RealTime end = RealTime::frame2RealTime(m_lastModelEndFrame, sourceRate); - if (end == RealTime::zeroTime) return; - - if (!constrained) { - m_rangeStarts.push_back(RealTime::zeroTime); - m_rangeDurations.push_back(end); - return; - } - - MultiSelection::SelectionList selections = m_viewManager->getSelections(); - MultiSelection::SelectionList::const_iterator i; - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - SVDEBUG << "AudioCallbackPlaySource::rebuildRangeLists" << endl; -#endif - - if (!selections.empty()) { - - for (i = selections.begin(); i != selections.end(); ++i) { - - RealTime start = - (RealTime::frame2RealTime - (m_viewManager->alignReferenceToPlaybackFrame(i->getStartFrame()), - sourceRate)); - RealTime duration = - (RealTime::frame2RealTime - (m_viewManager->alignReferenceToPlaybackFrame(i->getEndFrame()) - - m_viewManager->alignReferenceToPlaybackFrame(i->getStartFrame()), - sourceRate)); - - m_rangeStarts.push_back(start); - m_rangeDurations.push_back(duration); - } - } else { - m_rangeStarts.push_back(RealTime::zeroTime); - m_rangeDurations.push_back(end); - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cerr << "Now have " << m_rangeStarts.size() << " play ranges" << endl; -#endif -} - -void -AudioCallbackPlaySource::setOutputLevels(float left, float right) -{ - m_outputLeft = left; - m_outputRight = right; -} - -bool -AudioCallbackPlaySource::getOutputLevels(float &left, float &right) -{ - left = m_outputLeft; - right = m_outputRight; - return true; -} - -void -AudioCallbackPlaySource::setTargetSampleRate(sv_samplerate_t sr) -{ - bool first = (m_targetSampleRate == 0); - - m_targetSampleRate = sr; - initialiseConverter(); - - if (first && (m_stretchRatio != 1.f)) { - // couldn't create a stretcher before because we had no sample - // rate: make one now - setTimeStretch(m_stretchRatio); - } -} - -void -AudioCallbackPlaySource::initialiseConverter() -{ - m_mutex.lock(); - - if (m_converter) { - src_delete(m_converter); - src_delete(m_crapConverter); - m_converter = 0; - m_crapConverter = 0; - } - - if (getSourceSampleRate() != getTargetSampleRate()) { - - int err = 0; - - m_converter = src_new(m_resampleQuality == 2 ? SRC_SINC_BEST_QUALITY : - m_resampleQuality == 1 ? SRC_SINC_MEDIUM_QUALITY : - m_resampleQuality == 0 ? SRC_SINC_FASTEST : - SRC_SINC_MEDIUM_QUALITY, - getTargetChannelCount(), &err); - - if (m_converter) { - m_crapConverter = src_new(SRC_LINEAR, - getTargetChannelCount(), - &err); - } - - if (!m_converter || !m_crapConverter) { - cerr - << "AudioCallbackPlaySource::setModel: ERROR in creating samplerate converter: " - << src_strerror(err) << endl; - - if (m_converter) { - src_delete(m_converter); - m_converter = 0; - } - - if (m_crapConverter) { - src_delete(m_crapConverter); - m_crapConverter = 0; - } - - m_mutex.unlock(); - - emit sampleRateMismatch(getSourceSampleRate(), - getTargetSampleRate(), - false); - } else { - - m_mutex.unlock(); - - emit sampleRateMismatch(getSourceSampleRate(), - getTargetSampleRate(), - true); - } - } else { - m_mutex.unlock(); - } -} - -void -AudioCallbackPlaySource::setResampleQuality(int q) -{ - if (q == m_resampleQuality) return; - m_resampleQuality = q; - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - SVDEBUG << "AudioCallbackPlaySource::setResampleQuality: setting to " - << m_resampleQuality << endl; -#endif - - initialiseConverter(); -} - -void -AudioCallbackPlaySource::setAuditioningEffect(Auditionable *a) -{ - RealTimePluginInstance *plugin = dynamic_cast<RealTimePluginInstance *>(a); - if (a && !plugin) { - cerr << "WARNING: AudioCallbackPlaySource::setAuditioningEffect: auditionable object " << a << " is not a real-time plugin instance" << endl; - } - - m_mutex.lock(); - m_auditioningPlugin = plugin; - m_auditioningPluginBypassed = false; - m_mutex.unlock(); -} - -void -AudioCallbackPlaySource::setSoloModelSet(std::set<Model *> s) -{ - m_audioGenerator->setSoloModelSet(s); - clearRingBuffers(); -} - -void -AudioCallbackPlaySource::clearSoloModelSet() -{ - m_audioGenerator->clearSoloModelSet(); - clearRingBuffers(); -} - -sv_samplerate_t -AudioCallbackPlaySource::getTargetSampleRate() const -{ - if (m_targetSampleRate) return m_targetSampleRate; - else return getSourceSampleRate(); -} - -int -AudioCallbackPlaySource::getSourceChannelCount() const -{ - return m_sourceChannelCount; -} - -int -AudioCallbackPlaySource::getTargetChannelCount() const -{ - if (m_sourceChannelCount < 2) return 2; - return m_sourceChannelCount; -} - -sv_samplerate_t -AudioCallbackPlaySource::getSourceSampleRate() const -{ - return m_sourceSampleRate; -} - -void -AudioCallbackPlaySource::setTimeStretch(double factor) -{ - m_stretchRatio = factor; - - if (!getTargetSampleRate()) return; // have to make our stretcher later - - 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 - (int(getTargetSampleRate()), - m_stretcherInputCount, - RubberBandStretcher::OptionProcessRealTime, - factor); - RubberBandStretcher *monoStretcher = new RubberBandStretcher - (int(getTargetSampleRate()), - 1, - RubberBandStretcher::OptionProcessRealTime, - factor); - m_stretcherInputs = new float *[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]]; - } - m_monoStretcher = monoStretcher; - m_timeStretcher = stretcher; - } - - emit activity(tr("Change time-stretch factor to %1").arg(factor)); -} - -sv_frame_t -AudioCallbackPlaySource::getSourceSamples(sv_frame_t count, float **buffer) -{ - if (!m_playing) { -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - SVDEBUG << "AudioCallbackPlaySource::getSourceSamples: Not playing" << endl; -#endif - for (int ch = 0; ch < getTargetChannelCount(); ++ch) { - for (int i = 0; i < count; ++i) { - buffer[ch][i] = 0.0; - } - } - return 0; - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - SVDEBUG << "AudioCallbackPlaySource::getSourceSamples: Playing" << endl; -#endif - - // Ensure that all buffers have at least the amount of data we - // need -- else reduce the size of our requests correspondingly - - for (int ch = 0; ch < getTargetChannelCount(); ++ch) { - - RingBuffer<float> *rb = getReadRingBuffer(ch); - - if (!rb) { - cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: " - << "No ring buffer available for channel " << ch - << ", returning no data here" << endl; - count = 0; - break; - } - - int rs = rb->getReadSpace(); - if (rs < count) { -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: " - << "Ring buffer for channel " << ch << " has only " - << rs << " (of " << count << ") samples available (" - << "ring buffer size is " << rb->getSize() << ", write " - << "space " << rb->getWriteSpace() << "), " - << "reducing request size" << endl; -#endif - count = rs; - } - } - - if (count == 0) return 0; - - RubberBandStretcher *ts = m_timeStretcher; - RubberBandStretcher *ms = m_monoStretcher; - - 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 = 1.0; - } else { - ts->setTimeRatio(m_stretchRatio); - if (ms) ms->setTimeRatio(m_stretchRatio); - if (m_stretchRatio >= 1.0) m_stretchMono = false; - } - } - - int stretchChannels = m_stretcherInputCount; - if (m_stretchMono) { - if (ms) { - ts = ms; - stretchChannels = 1; - } else { - m_stretchMono = false; - } - } - - if (m_target) { - m_lastRetrievedBlockSize = count; - m_lastRetrievalTimestamp = m_target->getCurrentTime(); - } - - if (!ts || ratio == 1.f) { - - int got = 0; - - for (int ch = 0; ch < getTargetChannelCount(); ++ch) { - - RingBuffer<float> *rb = getReadRingBuffer(ch); - - if (rb) { - - // this is marginally more likely to leave our channels in - // sync after a processing failure than just passing "count": - sv_frame_t request = count; - if (ch > 0) request = got; - - 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; -#endif - } - - for (int ch = 0; ch < getTargetChannelCount(); ++ch) { - for (int i = got; i < count; ++i) { - buffer[ch][i] = 0.0; - } - } - } - - applyAuditioningEffect(count, buffer); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySource::getSamples: awakening thread" << endl; -#endif - - m_condition.wakeAll(); - - return got; - } - - int channels = getTargetChannelCount(); - sv_frame_t available; - sv_frame_t fedToStretcher = 0; - int warned = 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) { - - sv_frame_t reqd = lrint(double(count - available) / ratio); - reqd = std::max(reqd, sv_frame_t(ts->getSamplesRequired())); - if (reqd == 0) reqd = 1; - - sv_frame_t got = reqd; - -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - cerr << "reqd = " <<reqd << ", channels = " << channels << ", ic = " << m_stretcherInputCount << endl; -#endif - - for (int c = 0; c < channels; ++c) { - if (c >= m_stretcherInputCount) continue; - if (reqd > m_stretcherInputSizes[c]) { - if (c == 0) { - cerr << "WARNING: resizing stretcher input buffer from " << m_stretcherInputSizes[c] << " to " << (reqd * 2) << endl; - } - delete[] m_stretcherInputs[c]; - m_stretcherInputSizes[c] = reqd * 2; - m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]]; - } - } - - for (int c = 0; c < channels; ++c) { - if (c >= m_stretcherInputCount) continue; - RingBuffer<float> *rb = getReadRingBuffer(c); - if (rb) { - sv_frame_t gotHere; - if (stretchChannels == 1 && c > 0) { - gotHere = rb->readAdding(m_stretcherInputs[0], int(got)); - } else { - gotHere = rb->read(m_stretcherInputs[c], int(got)); - } - if (gotHere < got) got = gotHere; - -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - if (c == 0) { - SVDEBUG << "feeding stretcher: got " << gotHere - << ", " << rb->getReadSpace() << " remain" << endl; - } -#endif - - } else { - cerr << "WARNING: No ring buffer available for channel " << c << " in stretcher input block" << endl; - } - } - - if (got < reqd) { - cerr << "WARNING: Read underrun in playback (" - << got << " < " << reqd << ")" << endl; - } - - ts->process(m_stretcherInputs, size_t(got), false); - - fedToStretcher += got; - - if (got == 0) break; - - if (ts->available() == available) { - cerr << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples (warned = " << warned << ")" << endl; - if (++warned == 5) break; - } - } - - ts->retrieve(buffer, size_t(count)); - - for (int c = stretchChannels; c < getTargetChannelCount(); ++c) { - for (int i = 0; i < count; ++i) { - buffer[c][i] = buffer[0][i]; - } - } - - applyAuditioningEffect(count, buffer); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySource::getSamples [stretched]: awakening thread" << endl; -#endif - - m_condition.wakeAll(); - - return count; -} - -void -AudioCallbackPlaySource::applyAuditioningEffect(sv_frame_t count, float **buffers) -{ - if (m_auditioningPluginBypassed) return; - RealTimePluginInstance *plugin = m_auditioningPlugin; - if (!plugin) return; - - if ((int)plugin->getAudioInputCount() != getTargetChannelCount()) { -// cerr << "plugin input count " << plugin->getAudioInputCount() -// << " != our channel count " << getTargetChannelCount() -// << endl; - return; - } - if ((int)plugin->getAudioOutputCount() != getTargetChannelCount()) { -// cerr << "plugin output count " << plugin->getAudioOutputCount() -// << " != our channel count " << getTargetChannelCount() -// << endl; - return; - } - if ((int)plugin->getBufferSize() < count) { -// cerr << "plugin buffer size " << plugin->getBufferSize() -// << " < our block size " << count -// << endl; - return; - } - - float **ib = plugin->getAudioInputBuffers(); - float **ob = plugin->getAudioOutputBuffers(); - - for (int c = 0; c < getTargetChannelCount(); ++c) { - for (int i = 0; i < count; ++i) { - ib[c][i] = buffers[c][i]; - } - } - - plugin->run(Vamp::RealTime::zeroTime, int(count)); - - for (int c = 0; c < getTargetChannelCount(); ++c) { - for (int i = 0; i < count; ++i) { - buffers[c][i] = ob[c][i]; - } - } -} - -// Called from fill thread, m_playing true, mutex held -bool -AudioCallbackPlaySource::fillBuffers() -{ - static float *tmp = 0; - static sv_frame_t tmpSize = 0; - - sv_frame_t space = 0; - for (int c = 0; c < getTargetChannelCount(); ++c) { - RingBuffer<float> *wb = getWriteRingBuffer(c); - if (wb) { - sv_frame_t spaceHere = wb->getWriteSpace(); - if (c == 0 || spaceHere < space) space = spaceHere; - } - } - - if (space == 0) { -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySourceFillThread: no space to fill" << endl; -#endif - return false; - } - - sv_frame_t f = m_writeBufferFill; - - bool readWriteEqual = (m_readBuffers == m_writeBuffers); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - if (!readWriteEqual) { - cout << "AudioCallbackPlaySourceFillThread: note read buffers != write buffers" << endl; - } - cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << endl; -#endif - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "buffered to " << f << " already" << endl; -#endif - - bool resample = (getSourceSampleRate() != getTargetSampleRate()); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << (resample ? "" : "not ") << "resampling (source " << getSourceSampleRate() << ", target " << getTargetSampleRate() << ")" << endl; -#endif - - int channels = getTargetChannelCount(); - - sv_frame_t orig = space; - sv_frame_t got = 0; - - static float **bufferPtrs = 0; - static int bufferPtrCount = 0; - - if (bufferPtrCount < channels) { - if (bufferPtrs) delete[] bufferPtrs; - bufferPtrs = new float *[channels]; - bufferPtrCount = channels; - } - - sv_frame_t generatorBlockSize = m_audioGenerator->getBlockSize(); - - if (resample && !m_converter) { - static bool warned = false; - if (!warned) { - cerr << "WARNING: sample rates differ, but no converter available!" << endl; - warned = true; - } - } - - if (resample && m_converter) { - - double ratio = - double(getTargetSampleRate()) / double(getSourceSampleRate()); - 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; - - 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 - // the buffer (orig samples for channel 0, orig samples for - // channel 1 etc), and then interleave them into the first - // half of the buffer. Then we resample back into the second - // half (interleaved) and de-interleave the results back to - // the start of the buffer for insertion into the ringbuffers. - // What a faff -- especially as we've already de-interleaved - // the audio data from the source file elsewhere before we - // even reach this point. - - if (tmpSize < channels * work * 2) { - delete[] tmp; - tmp = new float[channels * work * 2]; - tmpSize = channels * work * 2; - } - - float *nonintlv = tmp + channels * work; - float *intlv = tmp; - float *srcout = tmp + channels * work; - - for (int c = 0; c < channels; ++c) { - for (int i = 0; i < orig; ++i) { - nonintlv[channels * i + c] = 0.0f; - } - } - - for (int c = 0; c < channels; ++c) { - bufferPtrs[c] = nonintlv + c * orig; - } - - got = mixModels(f, orig, bufferPtrs); // also modifies f - - // and interleave into first half - for (int c = 0; c < channels; ++c) { - for (int i = 0; i < got; ++i) { - float sample = nonintlv[c * got + i]; - intlv[channels * i + c] = sample; - } - } - - SRC_DATA data; - data.data_in = intlv; - data.data_out = srcout; - data.input_frames = long(got); - data.output_frames = long(work); - data.src_ratio = ratio; - data.end_of_input = 0; - - int err = 0; - - if (m_timeStretcher && m_timeStretcher->getTimeRatio() < 0.4) { -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "Using crappy converter" << endl; -#endif - err = src_process(m_crapConverter, &data); - } else { - err = src_process(m_converter, &data); - } - - sv_frame_t toCopy = sv_frame_t(double(got) * ratio + 0.1); - - if (err) { - cerr - << "AudioCallbackPlaySourceFillThread: ERROR in samplerate conversion: " - << src_strerror(err) << endl; - //!!! Then what? - } else { - got = data.input_frames_used; - toCopy = data.output_frames_gen; -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "Resampled " << got << " frames to " << toCopy << " frames" << endl; -#endif - } - - for (int c = 0; c < channels; ++c) { - for (int i = 0; i < toCopy; ++i) { - tmp[i] = srcout[channels * i + c]; - } - RingBuffer<float> *wb = getWriteRingBuffer(c); - if (wb) wb->write(tmp, int(toCopy)); - } - - m_writeBufferFill = f; - if (readWriteEqual) m_readBufferFill = f; - - } else { - - // space must be a multiple of generatorBlockSize - sv_frame_t reqSpace = space; - space = (reqSpace / generatorBlockSize) * generatorBlockSize; - if (space == 0) { -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "requested fill of " << reqSpace - << " is less than generator block size of " - << generatorBlockSize << ", leaving it" << endl; -#endif - return false; - } - - if (tmpSize < channels * space) { - delete[] tmp; - tmp = new float[channels * space]; - tmpSize = channels * space; - } - - for (int c = 0; c < channels; ++c) { - - bufferPtrs[c] = tmp + c * space; - - for (int i = 0; i < space; ++i) { - tmp[c * space + i] = 0.0f; - } - } - - 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], int(got)); -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "Wrote " << actual << " samples for ch " << c << ", now " - << wb->getReadSpace() << " to read" - << endl; -#endif - if (actual < got) { - cerr << "WARNING: Buffer overrun in channel " << c - << ": wrote " << actual << " of " << got - << " samples" << endl; - } - } - } - - m_writeBufferFill = f; - if (readWriteEqual) m_readBufferFill = f; - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "Read buffer fill is now " << m_readBufferFill << endl; -#endif - - //!!! how do we know when ended? need to mark up a fully-buffered flag and check this if we find the buffers empty in getSourceSamples - } - - return true; -} - -sv_frame_t -AudioCallbackPlaySource::mixModels(sv_frame_t &frame, sv_frame_t count, float **buffers) -{ - 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() && - !m_viewManager->getSelections().empty()); - - static float **chunkBufferPtrs = 0; - static int chunkBufferPtrCount = 0; - int channels = getTargetChannelCount(); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "Selection playback: start " << frame << ", size " << count <<", channels " << channels << endl; -#endif - - if (chunkBufferPtrCount < channels) { - if (chunkBufferPtrs) delete[] chunkBufferPtrs; - chunkBufferPtrs = new float *[channels]; - chunkBufferPtrCount = channels; - } - - for (int c = 0; c < channels; ++c) { - chunkBufferPtrs[c] = buffers[c]; - } - - while (processed < count) { - - chunkSize = count - processed; - nextChunkStart = chunkStart + chunkSize; - selectionSize = 0; - - sv_frame_t fadeIn = 0, fadeOut = 0; - - if (constrained) { - - sv_frame_t rChunkStart = - m_viewManager->alignPlaybackFrameToReference(chunkStart); - - Selection selection = - m_viewManager->getContainingSelection(rChunkStart, true); - - if (selection.isEmpty()) { - if (looping) { - selection = *m_viewManager->getSelections().begin(); - chunkStart = m_viewManager->alignReferenceToPlaybackFrame - (selection.getStartFrame()); - fadeIn = 50; - } - } - - if (selection.isEmpty()) { - - chunkSize = 0; - nextChunkStart = chunkStart; - - } else { - - sv_frame_t sf = m_viewManager->alignReferenceToPlaybackFrame - (selection.getStartFrame()); - sv_frame_t ef = m_viewManager->alignReferenceToPlaybackFrame - (selection.getEndFrame()); - - selectionSize = ef - sf; - - if (chunkStart < sf) { - chunkStart = sf; - fadeIn = 50; - } - - nextChunkStart = chunkStart + chunkSize; - - if (nextChunkStart >= ef) { - nextChunkStart = ef; - fadeOut = 50; - } - - chunkSize = nextChunkStart - chunkStart; - } - - } else if (looping && m_lastModelEndFrame > 0) { - - if (chunkStart >= m_lastModelEndFrame) { - chunkStart = 0; - } - if (chunkSize > m_lastModelEndFrame - chunkStart) { - chunkSize = m_lastModelEndFrame - chunkStart; - } - nextChunkStart = chunkStart + chunkSize; - } - -// cout << "chunkStart " << chunkStart << ", chunkSize " << chunkSize << ", nextChunkStart " << nextChunkStart << ", frame " << frame << ", count " << count << ", processed " << processed << endl; - - if (!chunkSize) { -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "Ending selection playback at " << nextChunkStart << endl; -#endif - // We need to maintain full buffers so that the other - // thread can tell where it's got to in the playback -- so - // return the full amount here - frame = frame + count; - return count; - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "Selection playback: chunk at " << chunkStart << " -> " << nextChunkStart << " (size " << chunkSize << ")" << endl; -#endif - - if (selectionSize < 100) { - fadeIn = 0; - fadeOut = 0; - } else if (selectionSize < 300) { - if (fadeIn > 0) fadeIn = 10; - if (fadeOut > 0) fadeOut = 10; - } - - if (fadeIn > 0) { - if (processed * 2 < fadeIn) { - fadeIn = processed * 2; - } - } - - if (fadeOut > 0) { - if ((count - processed - chunkSize) * 2 < fadeOut) { - fadeOut = (count - processed - chunkSize) * 2; - } - } - - for (std::set<Model *>::iterator mi = m_models.begin(); - mi != m_models.end(); ++mi) { - - (void) m_audioGenerator->mixModel(*mi, chunkStart, - chunkSize, chunkBufferPtrs, - fadeIn, fadeOut); - } - - for (int c = 0; c < channels; ++c) { - chunkBufferPtrs[c] += chunkSize; - } - - processed += chunkSize; - chunkStart = nextChunkStart; - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "Returning selection playback " << processed << " frames to " << nextChunkStart << endl; -#endif - - frame = nextChunkStart; - return processed; -} - -void -AudioCallbackPlaySource::unifyRingBuffers() -{ - if (m_readBuffers == m_writeBuffers) return; - - // only unify if there will be something to read - for (int c = 0; c < getTargetChannelCount(); ++c) { - RingBuffer<float> *wb = getWriteRingBuffer(c); - if (wb) { - if (wb->getReadSpace() < m_blockSize * 2) { - if ((m_writeBufferFill + m_blockSize * 2) < - m_lastModelEndFrame) { - // OK, we don't have enough and there's more to - // read -- don't unify until we can do better -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - SVDEBUG << "AudioCallbackPlaySource::unifyRingBuffers: Not unifying: write buffer has less (" << wb->getReadSpace() << ") than " << m_blockSize*2 << " to read and write buffer fill (" << m_writeBufferFill << ") is not close to end frame (" << m_lastModelEndFrame << ")" << endl; -#endif - return; - } - } - break; - } - } - - sv_frame_t rf = m_readBufferFill; - RingBuffer<float> *rb = getReadRingBuffer(0); - if (rb) { - int rs = rb->getReadSpace(); - //!!! incorrect when in non-contiguous selection, see comments elsewhere -// cout << "rs = " << rs << endl; - if (rs < rf) rf -= rs; - else rf = 0; - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - SVDEBUG << "AudioCallbackPlaySource::unifyRingBuffers: m_readBufferFill = " << m_readBufferFill << ", rf = " << rf << ", m_writeBufferFill = " << m_writeBufferFill << endl; -#endif - - 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) { - if (c == 0) { - - int wrs = wb->getReadSpace(); -// cout << "wrs = " << wrs << endl; - - if (wrs < wf) wf -= wrs; - else wf = 0; -// cout << "wf = " << wf << endl; - - if (wf < rf) skip = rf - wf; - if (skip == 0) break; - } - -// cout << "skipping " << skip << endl; - wb->skip(int(skip)); - } - } - - m_bufferScavenger.claim(m_readBuffers); - m_readBuffers = m_writeBuffers; - m_readBufferFill = m_writeBufferFill; -#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING - cerr << "unified" << endl; -#endif -} - -void -AudioCallbackPlaySource::FillThread::run() -{ - AudioCallbackPlaySource &s(m_source); - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySourceFillThread starting" << endl; -#endif - - s.m_mutex.lock(); - - bool previouslyPlaying = s.m_playing; - bool work = false; - - while (!s.m_exiting) { - - s.unifyRingBuffers(); - s.m_bufferScavenger.scavenge(); - s.m_pluginScavenger.scavenge(); - - if (work && s.m_playing && s.getSourceSampleRate()) { - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySourceFillThread: not waiting" << endl; -#endif - - s.m_mutex.unlock(); - s.m_mutex.lock(); - - } else { - - double ms = 100; - if (s.getSourceSampleRate() > 0) { - ms = double(s.m_ringBufferSize) / s.getSourceSampleRate() * 1000.0; - } - - if (s.m_playing) ms /= 10; - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - if (!s.m_playing) cout << endl; - cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms << "ms..." << endl; -#endif - - s.m_condition.wait(&s.m_mutex, int(ms)); - } - -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySourceFillThread: awoken" << endl; -#endif - - work = false; - - if (!s.getSourceSampleRate()) { -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySourceFillThread: source sample rate is zero" << endl; -#endif - continue; - } - - bool playing = s.m_playing; - - if (playing && !previouslyPlaying) { -#ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << endl; -#endif - for (int c = 0; c < s.getTargetChannelCount(); ++c) { - RingBuffer<float> *rb = s.getReadRingBuffer(c); - if (rb) rb->reset(); - } - } - previouslyPlaying = playing; - - work = s.fillBuffers(); - } - - s.m_mutex.unlock(); -} -
--- a/audioio/AudioCallbackPlaySource.h Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,384 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam and QMUL. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifndef _AUDIO_CALLBACK_PLAY_SOURCE_H_ -#define _AUDIO_CALLBACK_PLAY_SOURCE_H_ - -#include "base/RingBuffer.h" -#include "base/AudioPlaySource.h" -#include "base/PropertyContainer.h" -#include "base/Scavenger.h" - -#include <QObject> -#include <QMutex> -#include <QWaitCondition> - -#include "base/Thread.h" -#include "base/RealTime.h" - -#include <samplerate.h> - -#include <set> -#include <map> - -namespace RubberBand { - class RubberBandStretcher; -} - -class Model; -class ViewManagerBase; -class AudioGenerator; -class PlayParameters; -class RealTimePluginInstance; -class AudioCallbackPlayTarget; - -/** - * AudioCallbackPlaySource manages audio data supply to callback-based - * audio APIs such as JACK or CoreAudio. It maintains one ring buffer - * per channel, filled during playback by a non-realtime thread, and - * provides a method for a realtime thread to pick up the latest - * available sample data from these buffers. - */ -class AudioCallbackPlaySource : public QObject, - public AudioPlaySource -{ - Q_OBJECT - -public: - AudioCallbackPlaySource(ViewManagerBase *, QString clientName); - virtual ~AudioCallbackPlaySource(); - - /** - * Add a data model to be played from. The source can mix - * playback from a number of sources including dense and sparse - * models. The models must match in sample rate, but they don't - * have to have identical numbers of channels. - */ - virtual void addModel(Model *model); - - /** - * Remove a model. - */ - virtual void removeModel(Model *model); - - /** - * Remove all models. (Silence will ensue.) - */ - virtual void clearModels(); - - /** - * Start making data available in the ring buffers for playback, - * from the given frame. If playback is already under way, reseek - * to the given frame and continue. - */ - virtual void play(sv_frame_t startFrame); - - /** - * Stop playback and ensure that no more data is returned. - */ - virtual void stop(); - - /** - * Return whether playback is currently supposed to be happening. - */ - virtual bool isPlaying() const { return m_playing; } - - /** - * Return the frame number that is currently expected to be coming - * out of the speakers. (i.e. compensating for playback latency.) - */ - virtual sv_frame_t getCurrentPlayingFrame(); - - /** - * Return the last frame that would come out of the speakers if we - * stopped playback right now. - */ - virtual sv_frame_t getCurrentBufferedFrame(); - - /** - * Return the frame at which playback is expected to end (if not looping). - */ - virtual sv_frame_t getPlayEndFrame() { return m_lastModelEndFrame; } - - /** - * Set the target and the block size of the target audio device. - * This should be called by the target class. - */ - void setTarget(AudioCallbackPlayTarget *, int blockSize); - - /** - * Get the block size of the target audio device. This may be an - * estimate or upper bound, if the target has a variable block - * size; the source should behave itself even if this value turns - * out to be inaccurate. - */ - int getTargetBlockSize() const; - - /** - * Set the playback latency of the target audio device, in frames - * at the target sample rate. This is the difference between the - * frame currently "leaving the speakers" and the last frame (or - * highest last frame across all channels) requested via - * getSamples(). The default is zero. - */ - void setTargetPlayLatency(sv_frame_t); - - /** - * Get the playback latency of the target audio device. - */ - sv_frame_t getTargetPlayLatency() const; - - /** - * Specify that the target audio device has a fixed sample rate - * (i.e. cannot accommodate arbitrary sample rates based on the - * source). If the target sets this to something other than the - * source sample rate, this class will resample automatically to - * fit. - */ - 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 sv_samplerate_t getTargetSampleRate() const; - - /** - * Set the current output levels for metering (for call from the - * target) - */ - void setOutputLevels(float left, float right); - - /** - * Return the current (or thereabouts) output levels in the range - * 0.0 -> 1.0, for metering purposes. - */ - virtual bool getOutputLevels(float &left, float &right); - - /** - * Get the number of channels of audio that in the source models. - * This may safely be called from a realtime thread. Returns 0 if - * there is no source yet available. - */ - int getSourceChannelCount() const; - - /** - * Get the number of channels of audio that will be provided - * to the play target. This may be more than the source channel - * count: for example, a mono source will provide 2 channels - * after pan. - * This may safely be called from a realtime thread. Returns 0 if - * there is no source yet available. - */ - int getTargetChannelCount() const; - - /** - * Get the actual sample rate of the source material. This may - * safely be called from a realtime thread. Returns 0 if there is - * no source yet available. - */ - 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. - */ - sv_frame_t getSourceSamples(sv_frame_t count, float **buffer); - - /** - * Set the time stretcher factor (i.e. playback speed). - */ - void setTimeStretch(double factor); - - /** - * Set the resampler quality, 0 - 2 where 0 is fastest and 2 is - * highest quality. - */ - void setResampleQuality(int q); - - /** - * Set a single real-time plugin as a processing effect for - * auditioning during playback. - * - * The plugin must have been initialised with - * getTargetChannelCount() channels and a getTargetBlockSize() - * sample frame processing block size. - * - * This playback source takes ownership of the plugin, which will - * be deleted at some point after the following call to - * setAuditioningEffect (depending on real-time constraints). - * - * Pass a null pointer to remove the current auditioning plugin, - * if any. - */ - void setAuditioningEffect(Auditionable *plugin); - - /** - * Specify that only the given set of models should be played. - */ - void setSoloModelSet(std::set<Model *>s); - - /** - * Specify that all models should be played as normal (if not - * muted). - */ - void clearSoloModelSet(); - - QString getClientName() const { return m_clientName; } - -signals: - void modelReplaced(); - - void playStatusChanged(bool isPlaying); - - void sampleRateMismatch(sv_samplerate_t requested, - sv_samplerate_t available, - bool willResample); - - void audioOverloadPluginDisabled(); - void audioTimeStretchMultiChannelDisabled(); - - void activity(QString); - -public slots: - void audioProcessingOverload(); - -protected slots: - void selectionChanged(); - void playLoopModeChanged(); - void playSelectionModeChanged(); - void playParametersChanged(PlayParameters *); - void preferenceChanged(PropertyContainer::PropertyName); - void modelChangedWithin(sv_frame_t startFrame, sv_frame_t endFrame); - -protected: - ViewManagerBase *m_viewManager; - AudioGenerator *m_audioGenerator; - QString m_clientName; - - class RingBufferVector : public std::vector<RingBuffer<float> *> { - public: - virtual ~RingBufferVector() { - while (!empty()) { - delete *begin(); - erase(begin()); - } - } - }; - - std::set<Model *> m_models; - RingBufferVector *m_readBuffers; - RingBufferVector *m_writeBuffers; - sv_frame_t m_readBufferFill; - sv_frame_t m_writeBufferFill; - Scavenger<RingBufferVector> m_bufferScavenger; - int m_sourceChannelCount; - 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; - sv_frame_t m_lastRetrievedBlockSize; - bool m_trustworthyTimestamps; - sv_frame_t m_lastCurrentFrame; - bool m_playing; - bool m_exiting; - sv_frame_t m_lastModelEndFrame; - int m_ringBufferSize; - float m_outputLeft; - float m_outputRight; - RealTimePluginInstance *m_auditioningPlugin; - bool m_auditioningPluginBypassed; - Scavenger<RealTimePluginInstance> m_pluginScavenger; - sv_frame_t m_playStartFrame; - bool m_playStartFramePassed; - RealTime m_playStartedAt; - - RingBuffer<float> *getWriteRingBuffer(int c) { - if (m_writeBuffers && c < (int)m_writeBuffers->size()) { - return (*m_writeBuffers)[c]; - } else { - return 0; - } - } - - RingBuffer<float> *getReadRingBuffer(int c) { - RingBufferVector *rb = m_readBuffers; - if (rb && c < (int)rb->size()) { - return (*rb)[c]; - } else { - return 0; - } - } - - void clearRingBuffers(bool haveLock = false, int count = 0); - void unifyRingBuffers(); - - RubberBand::RubberBandStretcher *m_timeStretcher; - RubberBand::RubberBandStretcher *m_monoStretcher; - double m_stretchRatio; - bool m_stretchMono; - - int m_stretcherInputCount; - float **m_stretcherInputs; - sv_frame_t *m_stretcherInputSizes; - - // Called from fill thread, m_playing true, mutex held - // Return true if work done - bool fillBuffers(); - - // Called from fillBuffers. Return the number of frames written, - // 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). - sv_frame_t mixModels(sv_frame_t &frame, sv_frame_t count, float **buffers); - - // Called from getSourceSamples. - 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(); - - sv_frame_t getCurrentFrame(RealTime outputLatency); - - class FillThread : public Thread - { - public: - FillThread(AudioCallbackPlaySource &source) : - Thread(Thread::NonRTThread), - m_source(source) { } - - virtual void run(); - - protected: - AudioCallbackPlaySource &m_source; - }; - - QMutex m_mutex; - QWaitCondition m_condition; - FillThread *m_fillThread; - SRC_STATE *m_converter; - SRC_STATE *m_crapConverter; // for use when playing very fast - int m_resampleQuality; - void initialiseConverter(); -}; - -#endif - -
--- a/audioio/AudioCallbackPlayTarget.cpp Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,40 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#include "AudioCallbackPlayTarget.h" -#include "AudioCallbackPlaySource.h" - -#include <iostream> - -AudioCallbackPlayTarget::AudioCallbackPlayTarget(AudioCallbackPlaySource *source) : - m_source(source), - m_outputGain(1.0) -{ - if (m_source) { - connect(m_source, SIGNAL(modelReplaced()), - this, SLOT(sourceModelReplaced())); - } -} - -AudioCallbackPlayTarget::~AudioCallbackPlayTarget() -{ -} - -void -AudioCallbackPlayTarget::setOutputGain(float gain) -{ - m_outputGain = gain; -} -
--- a/audioio/AudioCallbackPlayTarget.h Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,63 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifndef _AUDIO_CALLBACK_PLAY_TARGET_H_ -#define _AUDIO_CALLBACK_PLAY_TARGET_H_ - -#include <QObject> - -class AudioCallbackPlaySource; - -class AudioCallbackPlayTarget : public QObject -{ - Q_OBJECT - -public: - AudioCallbackPlayTarget(AudioCallbackPlaySource *source); - virtual ~AudioCallbackPlayTarget(); - - virtual bool isOK() const = 0; - - virtual void shutdown() = 0; - - virtual double getCurrentTime() const = 0; - - float getOutputGain() const { - return m_outputGain; - } - -public slots: - /** - * Set the playback gain (0.0 = silence, 1.0 = levels unmodified) - */ - virtual void setOutputGain(float gain); - - /** - * The main source model (providing the playback sample rate) has - * been changed. The target should query the source's sample - * rate, set its output sample rate accordingly, and call back on - * the source's setTargetSampleRate to indicate what sample rate - * it succeeded in setting at the output. If this differs from - * the model rate, the source will resample. - */ - virtual void sourceModelReplaced() = 0; - -protected: - AudioCallbackPlaySource *m_source; - float m_outputGain; -}; - -#endif -
--- a/audioio/AudioGenerator.cpp Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,709 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#include "AudioGenerator.h" - -#include "base/TempDirectory.h" -#include "base/PlayParameters.h" -#include "base/PlayParameterRepository.h" -#include "base/Pitch.h" -#include "base/Exceptions.h" - -#include "data/model/NoteModel.h" -#include "data/model/FlexiNoteModel.h" -#include "data/model/DenseTimeValueModel.h" -#include "data/model/SparseTimeValueModel.h" -#include "data/model/SparseOneDimensionalModel.h" -#include "data/model/NoteData.h" - -#include "ClipMixer.h" -#include "ContinuousSynth.h" - -#include <iostream> -#include <cmath> - -#include <QDir> -#include <QFile> - -const sv_frame_t -AudioGenerator::m_processingBlockSize = 1024; - -QString -AudioGenerator::m_sampleDir = ""; - -//#define DEBUG_AUDIO_GENERATOR 1 - -AudioGenerator::AudioGenerator() : - m_sourceSampleRate(0), - m_targetChannelCount(1), - m_waveType(0), - m_soloing(false), - m_channelBuffer(0), - m_channelBufSiz(0), - m_channelBufCount(0) -{ - initialiseSampleDir(); - - connect(PlayParameterRepository::getInstance(), - SIGNAL(playClipIdChanged(const Playable *, QString)), - this, - SLOT(playClipIdChanged(const Playable *, QString))); -} - -AudioGenerator::~AudioGenerator() -{ -#ifdef DEBUG_AUDIO_GENERATOR - SVDEBUG << "AudioGenerator::~AudioGenerator" << endl; -#endif -} - -void -AudioGenerator::initialiseSampleDir() -{ - if (m_sampleDir != "") return; - - try { - m_sampleDir = TempDirectory::getInstance()->getSubDirectoryPath("samples"); - } catch (DirectoryCreationFailed f) { - cerr << "WARNING: AudioGenerator::initialiseSampleDir:" - << " Failed to create temporary sample directory" - << endl; - m_sampleDir = ""; - return; - } - - QDir sampleResourceDir(":/samples", "*.wav"); - - for (unsigned int i = 0; i < sampleResourceDir.count(); ++i) { - - QString fileName(sampleResourceDir[i]); - QFile file(sampleResourceDir.filePath(fileName)); - QString target = QDir(m_sampleDir).filePath(fileName); - - if (!file.copy(target)) { - cerr << "WARNING: AudioGenerator::getSampleDir: " - << "Unable to copy " << fileName - << " into temporary directory \"" - << m_sampleDir << "\"" << endl; - } else { - QFile tf(target); - tf.setPermissions(tf.permissions() | - QFile::WriteOwner | - QFile::WriteUser); - } - } -} - -bool -AudioGenerator::addModel(Model *model) -{ - if (m_sourceSampleRate == 0) { - - m_sourceSampleRate = model->getSampleRate(); - - } else { - - DenseTimeValueModel *dtvm = - dynamic_cast<DenseTimeValueModel *>(model); - - if (dtvm) { - m_sourceSampleRate = model->getSampleRate(); - return true; - } - } - - const Playable *playable = model; - if (!playable || !playable->canPlay()) return 0; - - PlayParameters *parameters = - PlayParameterRepository::getInstance()->getPlayParameters(playable); - - bool willPlay = !parameters->isPlayMuted(); - - if (usesClipMixer(model)) { - ClipMixer *mixer = makeClipMixerFor(model); - if (mixer) { - QMutexLocker locker(&m_mutex); - m_clipMixerMap[model] = mixer; - return willPlay; - } - } - - if (usesContinuousSynth(model)) { - ContinuousSynth *synth = makeSynthFor(model); - if (synth) { - QMutexLocker locker(&m_mutex); - m_continuousSynthMap[model] = synth; - return willPlay; - } - } - - return false; -} - -void -AudioGenerator::playClipIdChanged(const Playable *playable, QString) -{ - const Model *model = dynamic_cast<const Model *>(playable); - if (!model) { - cerr << "WARNING: AudioGenerator::playClipIdChanged: playable " - << playable << " is not a supported model type" - << endl; - return; - } - - if (m_clipMixerMap.find(model) == m_clipMixerMap.end()) return; - - ClipMixer *mixer = makeClipMixerFor(model); - if (mixer) { - QMutexLocker locker(&m_mutex); - m_clipMixerMap[model] = mixer; - } -} - -bool -AudioGenerator::usesClipMixer(const Model *model) -{ - bool clip = - (qobject_cast<const SparseOneDimensionalModel *>(model) || - qobject_cast<const NoteModel *>(model) || - qobject_cast<const FlexiNoteModel *>(model)); - return clip; -} - -bool -AudioGenerator::wantsQuieterClips(const Model *model) -{ - // basically, anything that usually has sustain (like notes) or - // often has multiple sounds at once (like notes) wants to use a - // quieter level than simple click tracks - bool does = - (qobject_cast<const NoteModel *>(model) || - qobject_cast<const FlexiNoteModel *>(model)); - return does; -} - -bool -AudioGenerator::usesContinuousSynth(const Model *model) -{ - bool cont = - (qobject_cast<const SparseTimeValueModel *>(model)); - return cont; -} - -ClipMixer * -AudioGenerator::makeClipMixerFor(const Model *model) -{ - QString clipId; - - const Playable *playable = model; - if (!playable || !playable->canPlay()) return 0; - - PlayParameters *parameters = - PlayParameterRepository::getInstance()->getPlayParameters(playable); - if (parameters) { - clipId = parameters->getPlayClipId(); - } - -#ifdef DEBUG_AUDIO_GENERATOR - std::cerr << "AudioGenerator::makeClipMixerFor(" << model << "): sample id = " << clipId << std::endl; -#endif - - if (clipId == "") { - SVDEBUG << "AudioGenerator::makeClipMixerFor(" << model << "): no sample, skipping" << endl; - return 0; - } - - ClipMixer *mixer = new ClipMixer(m_targetChannelCount, - m_sourceSampleRate, - m_processingBlockSize); - - double clipF0 = Pitch::getFrequencyForPitch(60, 0, 440.0); // required - - QString clipPath = QString("%1/%2.wav").arg(m_sampleDir).arg(clipId); - - double level = wantsQuieterClips(model) ? 0.5 : 1.0; - if (!mixer->loadClipData(clipPath, clipF0, level)) { - delete mixer; - return 0; - } - -#ifdef DEBUG_AUDIO_GENERATOR - std::cerr << "AudioGenerator::makeClipMixerFor(" << model << "): loaded clip " << clipId << std::endl; -#endif - - return mixer; -} - -ContinuousSynth * -AudioGenerator::makeSynthFor(const Model *model) -{ - const Playable *playable = model; - if (!playable || !playable->canPlay()) return 0; - - ContinuousSynth *synth = new ContinuousSynth(m_targetChannelCount, - m_sourceSampleRate, - m_processingBlockSize, - m_waveType); - -#ifdef DEBUG_AUDIO_GENERATOR - std::cerr << "AudioGenerator::makeSynthFor(" << model << "): created synth" << std::endl; -#endif - - return synth; -} - -void -AudioGenerator::removeModel(Model *model) -{ - SparseOneDimensionalModel *sodm = - dynamic_cast<SparseOneDimensionalModel *>(model); - if (!sodm) return; // nothing to do - - QMutexLocker locker(&m_mutex); - - if (m_clipMixerMap.find(sodm) == m_clipMixerMap.end()) return; - - ClipMixer *mixer = m_clipMixerMap[sodm]; - m_clipMixerMap.erase(sodm); - delete mixer; -} - -void -AudioGenerator::clearModels() -{ - QMutexLocker locker(&m_mutex); - - while (!m_clipMixerMap.empty()) { - ClipMixer *mixer = m_clipMixerMap.begin()->second; - m_clipMixerMap.erase(m_clipMixerMap.begin()); - delete mixer; - } -} - -void -AudioGenerator::reset() -{ - QMutexLocker locker(&m_mutex); - -#ifdef DEBUG_AUDIO_GENERATOR - cerr << "AudioGenerator::reset()" << endl; -#endif - - for (ClipMixerMap::iterator i = m_clipMixerMap.begin(); i != m_clipMixerMap.end(); ++i) { - if (i->second) { - i->second->reset(); - } - } - - m_noteOffs.clear(); -} - -void -AudioGenerator::setTargetChannelCount(int targetChannelCount) -{ - if (m_targetChannelCount == targetChannelCount) return; - -// SVDEBUG << "AudioGenerator::setTargetChannelCount(" << targetChannelCount << ")" << endl; - - QMutexLocker locker(&m_mutex); - m_targetChannelCount = targetChannelCount; - - for (ClipMixerMap::iterator i = m_clipMixerMap.begin(); i != m_clipMixerMap.end(); ++i) { - if (i->second) i->second->setChannelCount(targetChannelCount); - } -} - -sv_frame_t -AudioGenerator::getBlockSize() const -{ - return m_processingBlockSize; -} - -void -AudioGenerator::setSoloModelSet(std::set<Model *> s) -{ - QMutexLocker locker(&m_mutex); - - m_soloModelSet = s; - m_soloing = true; -} - -void -AudioGenerator::clearSoloModelSet() -{ - QMutexLocker locker(&m_mutex); - - m_soloModelSet.clear(); - m_soloing = false; -} - -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; - return frameCount; - } - - QMutexLocker locker(&m_mutex); - - Playable *playable = model; - if (!playable || !playable->canPlay()) return frameCount; - - PlayParameters *parameters = - PlayParameterRepository::getInstance()->getPlayParameters(playable); - if (!parameters) return frameCount; - - bool playing = !parameters->isPlayMuted(); - if (!playing) { -#ifdef DEBUG_AUDIO_GENERATOR - cout << "AudioGenerator::mixModel(" << model << "): muted" << endl; -#endif - return frameCount; - } - - if (m_soloing) { - if (m_soloModelSet.find(model) == m_soloModelSet.end()) { -#ifdef DEBUG_AUDIO_GENERATOR - cout << "AudioGenerator::mixModel(" << model << "): not one of the solo'd models" << endl; -#endif - return frameCount; - } - } - - float gain = parameters->getPlayGain(); - float pan = parameters->getPlayPan(); - - DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model); - if (dtvm) { - return mixDenseTimeValueModel(dtvm, startFrame, frameCount, - buffer, gain, pan, fadeIn, fadeOut); - } - - if (usesClipMixer(model)) { - return mixClipModel(model, startFrame, frameCount, - buffer, gain, pan); - } - - if (usesContinuousSynth(model)) { - return mixContinuousSynthModel(model, startFrame, frameCount, - buffer, gain, pan); - } - - std::cerr << "AudioGenerator::mixModel: WARNING: Model " << model << " of type " << model->getTypeName() << " is marked as playable, but I have no mechanism to play it" << std::endl; - - return frameCount; -} - -sv_frame_t -AudioGenerator::mixDenseTimeValueModel(DenseTimeValueModel *dtvm, - sv_frame_t startFrame, sv_frame_t frames, - float **buffer, float gain, float pan, - sv_frame_t fadeIn, sv_frame_t fadeOut) -{ - sv_frame_t maxFrames = frames + std::max(fadeIn, fadeOut); - - int modelChannels = dtvm->getChannelCount(); - - if (m_channelBufSiz < maxFrames || m_channelBufCount < modelChannels) { - - for (int c = 0; c < m_channelBufCount; ++c) { - delete[] m_channelBuffer[c]; - } - - delete[] m_channelBuffer; - m_channelBuffer = new float *[modelChannels]; - - for (int c = 0; c < modelChannels; ++c) { - m_channelBuffer[c] = new float[maxFrames]; - } - - m_channelBufCount = modelChannels; - m_channelBufSiz = maxFrames; - } - - sv_frame_t got = 0; - - if (startFrame >= fadeIn/2) { - got = dtvm->getMultiChannelData(0, modelChannels - 1, - startFrame - fadeIn/2, - frames + fadeOut/2 + fadeIn/2, - m_channelBuffer); - } else { - sv_frame_t missing = fadeIn/2 - startFrame; - - for (int c = 0; c < modelChannels; ++c) { - m_channelBuffer[c] += missing; - } - - if (missing > 0) { - cerr << "note: channelBufSiz = " << m_channelBufSiz - << ", frames + fadeOut/2 = " << frames + fadeOut/2 - << ", startFrame = " << startFrame - << ", missing = " << missing << endl; - } - - got = dtvm->getMultiChannelData(0, modelChannels - 1, - startFrame, - frames + fadeOut/2, - m_channelBuffer); - - for (int c = 0; c < modelChannels; ++c) { - m_channelBuffer[c] -= missing; - } - - got += missing; - } - - for (int c = 0; c < m_targetChannelCount; ++c) { - - int sourceChannel = (c % modelChannels); - -// SVDEBUG << "mixing channel " << c << " from source channel " << sourceChannel << endl; - - float channelGain = gain; - if (pan != 0.0) { - if (c == 0) { - if (pan > 0.0) channelGain *= 1.0f - pan; - } else { - if (pan < 0.0) channelGain *= pan + 1.0f; - } - } - - for (sv_frame_t i = 0; i < fadeIn/2; ++i) { - float *back = buffer[c]; - back -= fadeIn/2; - back[i] += - (channelGain * m_channelBuffer[sourceChannel][i] * float(i)) - / float(fadeIn); - } - - for (sv_frame_t i = 0; i < frames + fadeOut/2; ++i) { - float mult = channelGain; - if (i < fadeIn/2) { - mult = (mult * float(i)) / float(fadeIn); - } - if (i > frames - fadeOut/2) { - mult = (mult * float((frames + fadeOut/2) - i)) / float(fadeOut); - } - float val = m_channelBuffer[sourceChannel][i]; - if (i >= got) val = 0.f; - buffer[c][i] += mult * val; - } - } - - return got; -} - -sv_frame_t -AudioGenerator::mixClipModel(Model *model, - 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 = int(frames / m_processingBlockSize); - - //!!! todo: the below -- it matters - - //!!! hang on -- the fact that the audio callback play source's - //buffer is a multiple of the plugin's buffer size doesn't mean - //that we always get called for a multiple of it here (because it - //also depends on the JACK block size). how should we ensure that - //all models write the same amount in to the mix, and that we - //always have a multiple of the plugin buffer size? I guess this - //class has to be queryable for the plugin buffer size & the - //callback play source has to use that as a multiple for all the - //calls to mixModel - - sv_frame_t got = blocks * m_processingBlockSize; - -#ifdef DEBUG_AUDIO_GENERATOR - cout << "mixModel [clip]: start " << startFrame << ", frames " << frames - << ", blocks " << blocks << ", have " << m_noteOffs.size() - << " note-offs" << endl; -#endif - - ClipMixer::NoteStart on; - ClipMixer::NoteEnd off; - - NoteOffSet ¬eOffs = m_noteOffs[model]; - - float **bufferIndexes = new float *[m_targetChannelCount]; - - for (int i = 0; i < blocks; ++i) { - - sv_frame_t reqStart = startFrame + i * m_processingBlockSize; - - NoteList notes; - NoteExportable *exportable = dynamic_cast<NoteExportable *>(model); - if (exportable) { - notes = exportable->getNotesWithin(reqStart, - reqStart + m_processingBlockSize); - } - - std::vector<ClipMixer::NoteStart> starts; - std::vector<ClipMixer::NoteEnd> ends; - - for (NoteList::const_iterator ni = notes.begin(); - ni != notes.end(); ++ni) { - - sv_frame_t noteFrame = ni->start; - - if (noteFrame < reqStart || - noteFrame >= reqStart + m_processingBlockSize) continue; - - while (noteOffs.begin() != noteOffs.end() && - noteOffs.begin()->frame <= noteFrame) { - - sv_frame_t eventFrame = noteOffs.begin()->frame; - if (eventFrame < reqStart) eventFrame = reqStart; - - off.frameOffset = eventFrame - reqStart; - off.frequency = noteOffs.begin()->frequency; - -#ifdef DEBUG_AUDIO_GENERATOR - cerr << "mixModel [clip]: adding note-off at frame " << eventFrame << " frame offset " << off.frameOffset << " frequency " << off.frequency << endl; -#endif - - ends.push_back(off); - noteOffs.erase(noteOffs.begin()); - } - - on.frameOffset = noteFrame - reqStart; - on.frequency = ni->getFrequency(); - on.level = float(ni->velocity) / 127.0f; - on.pan = pan; - -#ifdef DEBUG_AUDIO_GENERATOR - cout << "mixModel [clip]: adding note at frame " << noteFrame << ", frame offset " << on.frameOffset << " frequency " << on.frequency << ", level " << on.level << endl; -#endif - - starts.push_back(on); - noteOffs.insert - (NoteOff(on.frequency, noteFrame + ni->duration)); - } - - while (noteOffs.begin() != noteOffs.end() && - noteOffs.begin()->frame <= reqStart + m_processingBlockSize) { - - sv_frame_t eventFrame = noteOffs.begin()->frame; - if (eventFrame < reqStart) eventFrame = reqStart; - - off.frameOffset = eventFrame - reqStart; - off.frequency = noteOffs.begin()->frequency; - -#ifdef DEBUG_AUDIO_GENERATOR - cerr << "mixModel [clip]: adding leftover note-off at frame " << eventFrame << " frame offset " << off.frameOffset << " frequency " << off.frequency << endl; -#endif - - ends.push_back(off); - noteOffs.erase(noteOffs.begin()); - } - - for (int c = 0; c < m_targetChannelCount; ++c) { - bufferIndexes[c] = buffer[c] + i * m_processingBlockSize; - } - - clipMixer->mix(bufferIndexes, gain, starts, ends); - } - - delete[] bufferIndexes; - - return got; -} - -sv_frame_t -AudioGenerator::mixContinuousSynthModel(Model *model, - sv_frame_t startFrame, - sv_frame_t frames, - float **buffer, - float gain, - float pan) -{ - ContinuousSynth *synth = m_continuousSynthMap[model]; - if (!synth) return 0; - - // only type we support here at the moment - SparseTimeValueModel *stvm = qobject_cast<SparseTimeValueModel *>(model); - if (stvm->getScaleUnits() != "Hz") return 0; - - int blocks = int(frames / m_processingBlockSize); - - //!!! todo: see comment in mixClipModel - - sv_frame_t got = blocks * m_processingBlockSize; - -#ifdef DEBUG_AUDIO_GENERATOR - cout << "mixModel [synth]: frames " << frames - << ", blocks " << blocks << endl; -#endif - - float **bufferIndexes = new float *[m_targetChannelCount]; - - for (int i = 0; i < blocks; ++i) { - - sv_frame_t reqStart = startFrame + i * m_processingBlockSize; - - for (int c = 0; c < m_targetChannelCount; ++c) { - bufferIndexes[c] = buffer[c] + i * m_processingBlockSize; - } - - SparseTimeValueModel::PointList points = - stvm->getPoints(reqStart, reqStart + m_processingBlockSize); - - // by default, repeat last frequency - float f0 = 0.f; - - // go straight to the last freq that is genuinely in this range - for (SparseTimeValueModel::PointList::const_iterator itr = points.end(); - itr != points.begin(); ) { - --itr; - if (itr->frame >= reqStart && - itr->frame < reqStart + m_processingBlockSize) { - f0 = itr->value; - break; - } - } - - // if we found no such frequency and the next point is further - // away than twice the model resolution, go silent (same - // criterion TimeValueLayer uses for ending a discrete curve - // segment) - if (f0 == 0.f) { - SparseTimeValueModel::PointList nextPoints = - stvm->getNextPoints(reqStart + m_processingBlockSize); - if (nextPoints.empty() || - nextPoints.begin()->frame > reqStart + 2 * stvm->getResolution()) { - f0 = -1.f; - } - } - -// cerr << "f0 = " << f0 << endl; - - synth->mix(bufferIndexes, - gain, - pan, - f0); - } - - delete[] bufferIndexes; - - return got; -} -
--- a/audioio/AudioGenerator.h Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,168 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifndef _AUDIO_GENERATOR_H_ -#define _AUDIO_GENERATOR_H_ - -class Model; -class NoteModel; -class FlexiNoteModel; -class DenseTimeValueModel; -class SparseOneDimensionalModel; -class Playable; -class ClipMixer; -class ContinuousSynth; - -#include <QObject> -#include <QMutex> - -#include <set> -#include <map> -#include <vector> - -#include "base/BaseTypes.h" - -class AudioGenerator : public QObject -{ - Q_OBJECT - -public: - AudioGenerator(); - virtual ~AudioGenerator(); - - /** - * Add a data model to be played from and initialise any necessary - * audio generation code. Returns true if the model will be - * played. The model will be added regardless of the return - * value. - */ - virtual bool addModel(Model *model); - - /** - * Remove a model. - */ - virtual void removeModel(Model *model); - - /** - * Remove all models. - */ - virtual void clearModels(); - - /** - * Reset playback, clearing buffers and the like. - */ - virtual void reset(); - - /** - * Set the target channel count. The buffer parameter to mixModel - * must always point to at least this number of arrays. - */ - virtual void setTargetChannelCount(int channelCount); - - /** - * Return the internal processing block size. The frameCount - * argument to all mixModel calls must be a multiple of this - * value. - */ - virtual sv_frame_t getBlockSize() const; - - /** - * Mix a single model into an output buffer. - */ - 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. - */ - virtual void setSoloModelSet(std::set<Model *>s); - - /** - * Specify that all models should be played as normal (if not - * muted). - */ - virtual void clearSoloModelSet(); - -protected slots: - void playClipIdChanged(const Playable *, QString); - -protected: - sv_samplerate_t m_sourceSampleRate; - int m_targetChannelCount; - int m_waveType; - - bool m_soloing; - std::set<Model *> m_soloModelSet; - - struct NoteOff { - - NoteOff(float _freq, sv_frame_t _frame) : frequency(_freq), frame(_frame) { } - - float frequency; - sv_frame_t frame; - - struct Comparator { - bool operator()(const NoteOff &n1, const NoteOff &n2) const { - return n1.frame < n2.frame; - } - }; - }; - - - typedef std::map<const Model *, ClipMixer *> ClipMixerMap; - - typedef std::multiset<NoteOff, NoteOff::Comparator> NoteOffSet; - typedef std::map<const Model *, NoteOffSet> NoteOffMap; - - typedef std::map<const Model *, ContinuousSynth *> ContinuousSynthMap; - - QMutex m_mutex; - - ClipMixerMap m_clipMixerMap; - NoteOffMap m_noteOffs; - static QString m_sampleDir; - - ContinuousSynthMap m_continuousSynthMap; - - bool usesClipMixer(const Model *); - bool wantsQuieterClips(const Model *); - bool usesContinuousSynth(const Model *); - - ClipMixer *makeClipMixerFor(const Model *model); - ContinuousSynth *makeSynthFor(const Model *model); - - static void initialiseSampleDir(); - - 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 sv_frame_t mixClipModel - (Model *model, sv_frame_t startFrame, sv_frame_t frameCount, - float **buffer, float gain, float pan); - - virtual sv_frame_t mixContinuousSynthModel - (Model *model, sv_frame_t startFrame, sv_frame_t frameCount, - float **buffer, float gain, float pan); - - static const sv_frame_t m_processingBlockSize; - - float **m_channelBuffer; - sv_frame_t m_channelBufSiz; - int m_channelBufCount; -}; - -#endif -
--- a/audioio/AudioJACKTarget.cpp Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,487 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifdef HAVE_JACK - -#include "AudioJACKTarget.h" -#include "AudioCallbackPlaySource.h" - -#include <iostream> -#include <cmath> - -#include <alloca.h> - -//#define DEBUG_AUDIO_JACK_TARGET 1 - -#ifdef BUILD_STATIC -#ifdef Q_OS_LINUX - -// Some lunacy to enable JACK support in static builds. JACK isn't -// supposed to be linked statically, because it depends on a -// consistent shared memory layout between client library and daemon, -// so it's very fragile in the face of version mismatches. -// -// Therefore for static builds on Linux we avoid linking against JACK -// at all during the build, instead using dlopen and runtime symbol -// lookup to switch on JACK support at runtime. The following big -// mess (down to the #endifs) is the code that implements this. - -static void *symbol(const char *name) -{ - static bool attempted = false; - static void *library = 0; - static std::map<const char *, void *> symbols; - if (symbols.find(name) != symbols.end()) return symbols[name]; - if (!library) { - if (!attempted) { - library = ::dlopen("libjack.so.1", RTLD_NOW); - if (!library) library = ::dlopen("libjack.so.0", RTLD_NOW); - if (!library) library = ::dlopen("libjack.so", RTLD_NOW); - if (!library) { - cerr << "WARNING: AudioJACKTarget: Failed to load JACK library: " - << ::dlerror() << " (tried .so, .so.0, .so.1)" - << endl; - } - attempted = true; - } - if (!library) return 0; - } - void *symbol = ::dlsym(library, name); - if (!symbol) { - cerr << "WARNING: AudioJACKTarget: Failed to locate symbol " - << name << ": " << ::dlerror() << endl; - } - symbols[name] = symbol; - return symbol; -} - -static jack_client_t *dynamic_jack_client_open(const char *client_name, - jack_options_t options, - jack_status_t *status, ...) -{ - typedef jack_client_t *(*func)(const char *client_name, - jack_options_t options, - jack_status_t *status, ...); - void *s = symbol("jack_client_open"); - if (!s) return 0; - func f = (func)s; - return f(client_name, options, status); // varargs not supported here -} - -static int dynamic_jack_set_process_callback(jack_client_t *client, - JackProcessCallback process_callback, - void *arg) -{ - typedef int (*func)(jack_client_t *client, - JackProcessCallback process_callback, - void *arg); - void *s = symbol("jack_set_process_callback"); - if (!s) return 1; - func f = (func)s; - return f(client, process_callback, arg); -} - -static int dynamic_jack_set_xrun_callback(jack_client_t *client, - JackXRunCallback xrun_callback, - void *arg) -{ - typedef int (*func)(jack_client_t *client, - JackXRunCallback xrun_callback, - void *arg); - void *s = symbol("jack_set_xrun_callback"); - if (!s) return 1; - func f = (func)s; - return f(client, xrun_callback, arg); -} - -static const char **dynamic_jack_get_ports(jack_client_t *client, - const char *port_name_pattern, - const char *type_name_pattern, - unsigned long flags) -{ - typedef const char **(*func)(jack_client_t *client, - const char *port_name_pattern, - const char *type_name_pattern, - unsigned long flags); - void *s = symbol("jack_get_ports"); - if (!s) return 0; - func f = (func)s; - return f(client, port_name_pattern, type_name_pattern, flags); -} - -static jack_port_t *dynamic_jack_port_register(jack_client_t *client, - const char *port_name, - const char *port_type, - unsigned long flags, - unsigned long buffer_size) -{ - typedef jack_port_t *(*func)(jack_client_t *client, - const char *port_name, - const char *port_type, - unsigned long flags, - unsigned long buffer_size); - void *s = symbol("jack_port_register"); - if (!s) return 0; - func f = (func)s; - return f(client, port_name, port_type, flags, buffer_size); -} - -static int dynamic_jack_connect(jack_client_t *client, - const char *source, - const char *dest) -{ - typedef int (*func)(jack_client_t *client, - const char *source, - const char *dest); - void *s = symbol("jack_connect"); - if (!s) return 1; - func f = (func)s; - return f(client, source, dest); -} - -static void *dynamic_jack_port_get_buffer(jack_port_t *port, - jack_nframes_t sz) -{ - typedef void *(*func)(jack_port_t *, jack_nframes_t); - void *s = symbol("jack_port_get_buffer"); - if (!s) return 0; - func f = (func)s; - return f(port, sz); -} - -static int dynamic_jack_port_unregister(jack_client_t *client, - jack_port_t *port) -{ - typedef int(*func)(jack_client_t *, jack_port_t *); - void *s = symbol("jack_port_unregister"); - if (!s) return 0; - func f = (func)s; - return f(client, port); -} - -static void dynamic_jack_port_get_latency_range(jack_port_t *port, - jack_latency_callback_mode_t mode, - jack_latency_range_t *range) -{ - typedef void (*func)(jack_port_t *, jack_latency_callback_mode_t, jack_latency_range_t *); - void *s = symbol("jack_port_get_latency_range"); - if (!s) { - range->min = range->max = 0; - return; - } - func f = (func)s; - f(port, mode, range); -} - -#define dynamic1(rv, name, argtype, failval) \ - static rv dynamic_##name(argtype arg) { \ - typedef rv (*func) (argtype); \ - void *s = symbol(#name); \ - if (!s) return failval; \ - func f = (func) s; \ - return f(arg); \ - } - -dynamic1(jack_client_t *, jack_client_new, const char *, 0); -dynamic1(jack_nframes_t, jack_get_buffer_size, jack_client_t *, 0); -dynamic1(jack_nframes_t, jack_get_sample_rate, jack_client_t *, 0); -dynamic1(int, jack_activate, jack_client_t *, 1); -dynamic1(int, jack_deactivate, jack_client_t *, 1); -dynamic1(int, jack_client_close, jack_client_t *, 1); -dynamic1(jack_nframes_t, jack_frame_time, jack_client_t *, 0); -dynamic1(const char *, jack_port_name, const jack_port_t *, 0); - -#define jack_client_new dynamic_jack_client_new -#define jack_client_open dynamic_jack_client_open -#define jack_get_buffer_size dynamic_jack_get_buffer_size -#define jack_get_sample_rate dynamic_jack_get_sample_rate -#define jack_set_process_callback dynamic_jack_set_process_callback -#define jack_set_xrun_callback dynamic_jack_set_xrun_callback -#define jack_activate dynamic_jack_activate -#define jack_deactivate dynamic_jack_deactivate -#define jack_client_close dynamic_jack_client_close -#define jack_frame_time dynamic_jack_frame_time -#define jack_get_ports dynamic_jack_get_ports -#define jack_port_register dynamic_jack_port_register -#define jack_port_unregister dynamic_jack_port_unregister -#define jack_port_name dynamic_jack_port_name -#define jack_connect dynamic_jack_connect -#define jack_port_get_buffer dynamic_jack_port_get_buffer - -#endif -#endif - -AudioJACKTarget::AudioJACKTarget(AudioCallbackPlaySource *source) : - AudioCallbackPlayTarget(source), - m_client(0), - m_bufferSize(0), - m_sampleRate(0), - m_done(false) -{ - JackOptions options = JackNullOption; -#ifdef HAVE_PORTAUDIO_2_0 - options = JackNoStartServer; -#endif -#ifdef HAVE_LIBPULSE - options = JackNoStartServer; -#endif - - JackStatus status = JackStatus(0); - m_client = jack_client_open(source->getClientName().toLocal8Bit().data(), - options, &status); - - if (!m_client) { - cerr << "AudioJACKTarget: Failed to connect to JACK server: status code " - << status << endl; - return; - } - - m_bufferSize = jack_get_buffer_size(m_client); - m_sampleRate = jack_get_sample_rate(m_client); - - jack_set_xrun_callback(m_client, xrunStatic, this); - jack_set_process_callback(m_client, processStatic, this); - - if (jack_activate(m_client)) { - cerr << "ERROR: AudioJACKTarget: Failed to activate JACK client" - << endl; - } - - if (m_source) { - sourceModelReplaced(); - } - - // Mainstream JACK (though not jackdmp) calls mlockall() to lock - // down all memory for real-time operation. That isn't a terribly - // good idea in an application like this that may have very high - // dynamic memory usage in other threads, as mlockall() applies - // across all threads. We're far better off undoing it here and - // accepting the possible loss of true RT capability. - MUNLOCKALL(); -} - -AudioJACKTarget::~AudioJACKTarget() -{ - SVDEBUG << "AudioJACKTarget::~AudioJACKTarget()" << endl; - - if (m_source) { - m_source->setTarget(0, m_bufferSize); - } - - shutdown(); - - if (m_client) { - - while (m_outputs.size() > 0) { - std::vector<jack_port_t *>::iterator itr = m_outputs.end(); - --itr; - jack_port_t *port = *itr; - cerr << "unregister " << m_outputs.size() << endl; - if (port) jack_port_unregister(m_client, port); - m_outputs.erase(itr); - } - cerr << "Deactivating... "; - jack_deactivate(m_client); - cerr << "done\nClosing... "; - jack_client_close(m_client); - cerr << "done" << endl; - } - - m_client = 0; - - SVDEBUG << "AudioJACKTarget::~AudioJACKTarget() done" << endl; -} - -void -AudioJACKTarget::shutdown() -{ - m_done = true; -} - -bool -AudioJACKTarget::isOK() const -{ - return (m_client != 0); -} - -double -AudioJACKTarget::getCurrentTime() const -{ - if (m_client && m_sampleRate) { - return double(jack_frame_time(m_client)) / double(m_sampleRate); - } else { - return 0.0; - } -} - -int -AudioJACKTarget::processStatic(jack_nframes_t nframes, void *arg) -{ - return ((AudioJACKTarget *)arg)->process(nframes); -} - -int -AudioJACKTarget::xrunStatic(void *arg) -{ - return ((AudioJACKTarget *)arg)->xrun(); -} - -void -AudioJACKTarget::sourceModelReplaced() -{ - m_mutex.lock(); - - m_source->setTarget(this, m_bufferSize); - m_source->setTargetSampleRate(m_sampleRate); - - int channels = m_source->getSourceChannelCount(); - - // Because we offer pan, we always want at least 2 channels - if (channels < 2) channels = 2; - - if (channels == (int)m_outputs.size() || !m_client) { - m_mutex.unlock(); - return; - } - - const char **ports = - jack_get_ports(m_client, NULL, NULL, - JackPortIsPhysical | JackPortIsInput); - int physicalPortCount = 0; - while (ports[physicalPortCount]) ++physicalPortCount; - -#ifdef DEBUG_AUDIO_JACK_TARGET - SVDEBUG << "AudioJACKTarget::sourceModelReplaced: have " << channels << " channels and " << physicalPortCount << " physical ports" << endl; -#endif - - while ((int)m_outputs.size() < channels) { - - const int namelen = 30; - char name[namelen]; - jack_port_t *port; - - snprintf(name, namelen, "out %d", int(m_outputs.size() + 1)); - - port = jack_port_register(m_client, - name, - JACK_DEFAULT_AUDIO_TYPE, - JackPortIsOutput, - 0); - - if (!port) { - cerr - << "ERROR: AudioJACKTarget: Failed to create JACK output port " - << m_outputs.size() << endl; - } else { - jack_latency_range_t range; - jack_port_get_latency_range(port, JackPlaybackLatency, &range); - m_source->setTargetPlayLatency(range.max); - cerr << "AudioJACKTarget: output latency is " << range.max << endl; - } - - if ((int)m_outputs.size() < physicalPortCount) { - jack_connect(m_client, jack_port_name(port), ports[m_outputs.size()]); - } - - m_outputs.push_back(port); - } - - while ((int)m_outputs.size() > channels) { - std::vector<jack_port_t *>::iterator itr = m_outputs.end(); - --itr; - jack_port_t *port = *itr; - if (port) jack_port_unregister(m_client, port); - m_outputs.erase(itr); - } - - m_mutex.unlock(); -} - -int -AudioJACKTarget::process(jack_nframes_t nframes) -{ - if (m_done) return 0; - - if (!m_mutex.tryLock()) { - return 0; - } - - if (m_outputs.empty()) { - m_mutex.unlock(); - return 0; - } - -#ifdef DEBUG_AUDIO_JACK_TARGET - cout << "AudioJACKTarget::process(" << nframes << "): have a source" << endl; -#endif - -#ifdef DEBUG_AUDIO_JACK_TARGET - if (m_bufferSize != nframes) { - cerr << "WARNING: m_bufferSize != nframes (" << m_bufferSize << " != " << nframes << ")" << endl; - } -#endif - - float **buffers = (float **)alloca(m_outputs.size() * sizeof(float *)); - - for (int ch = 0; ch < (int)m_outputs.size(); ++ch) { - buffers[ch] = (float *)jack_port_get_buffer(m_outputs[ch], nframes); - } - - 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 (sv_frame_t i = received; i < nframes; ++i) { - buffers[ch][i] = 0.0; - } - } - - float peakLeft = 0.0, peakRight = 0.0; - - for (int ch = 0; ch < (int)m_outputs.size(); ++ch) { - - float peak = 0.0; - - for (int i = 0; i < (int)nframes; ++i) { - buffers[ch][i] *= m_outputGain; - float sample = fabsf(buffers[ch][i]); - if (sample > peak) peak = sample; - } - - if (ch == 0) peakLeft = peak; - if (ch > 0 || m_outputs.size() == 1) peakRight = peak; - } - - if (m_source) { - m_source->setOutputLevels(peakLeft, peakRight); - } - - m_mutex.unlock(); - return 0; -} - -int -AudioJACKTarget::xrun() -{ - cerr << "AudioJACKTarget: xrun!" << endl; - if (m_source) m_source->audioProcessingOverload(); - return 0; -} - -#endif /* HAVE_JACK */ -
--- a/audioio/AudioJACKTarget.h Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,65 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifndef _AUDIO_JACK_TARGET_H_ -#define _AUDIO_JACK_TARGET_H_ - -#ifdef HAVE_JACK - -#include <jack/jack.h> -#include <vector> - -#include "AudioCallbackPlayTarget.h" - -#include <QMutex> - -class AudioCallbackPlaySource; - -class AudioJACKTarget : public AudioCallbackPlayTarget -{ - Q_OBJECT - -public: - AudioJACKTarget(AudioCallbackPlaySource *source); - virtual ~AudioJACKTarget(); - - virtual void shutdown(); - - virtual bool isOK() const; - - virtual double getCurrentTime() const; - -public slots: - virtual void sourceModelReplaced(); - -protected: - int process(jack_nframes_t nframes); - int xrun(); - - static int processStatic(jack_nframes_t, void *); - static int xrunStatic(void *); - - jack_client_t *m_client; - std::vector<jack_port_t *> m_outputs; - jack_nframes_t m_bufferSize; - jack_nframes_t m_sampleRate; - QMutex m_mutex; - bool m_done; -}; - -#endif /* HAVE_JACK */ - -#endif -
--- a/audioio/AudioPortAudioTarget.cpp Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,300 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifdef HAVE_PORTAUDIO_2_0 - -#include "AudioPortAudioTarget.h" -#include "AudioCallbackPlaySource.h" - -#include <iostream> -#include <cassert> -#include <cmath> - -#ifndef _WIN32 -#include <pthread.h> -#endif - -//#define DEBUG_AUDIO_PORT_AUDIO_TARGET 1 - -AudioPortAudioTarget::AudioPortAudioTarget(AudioCallbackPlaySource *source) : - AudioCallbackPlayTarget(source), - m_stream(0), - m_bufferSize(0), - m_sampleRate(0), - m_latency(0), - m_prioritySet(false), - m_done(false) -{ - PaError err; - -#ifdef DEBUG_AUDIO_PORT_AUDIO_TARGET - cerr << "AudioPortAudioTarget: Initialising for PortAudio v19" << endl; -#endif - - err = Pa_Initialize(); - if (err != paNoError) { - cerr << "ERROR: AudioPortAudioTarget: Failed to initialize PortAudio: " << Pa_GetErrorText(err) << endl; - return; - } - - m_bufferSize = 2048; - m_sampleRate = 44100; - if (m_source && (m_source->getSourceSampleRate() != 0)) { - m_sampleRate = int(m_source->getSourceSampleRate()); - } - - PaStreamParameters op; - op.device = Pa_GetDefaultOutputDevice(); - op.channelCount = 2; - op.sampleFormat = paFloat32; - op.suggestedLatency = 0.2; - op.hostApiSpecificStreamInfo = 0; - err = Pa_OpenStream(&m_stream, 0, &op, m_sampleRate, - paFramesPerBufferUnspecified, - paNoFlag, processStatic, this); - - if (err != paNoError) { - - cerr << "WARNING: AudioPortAudioTarget: Failed to open PortAudio stream with default frames per buffer, trying again with fixed frames per buffer..." << endl; - - err = Pa_OpenStream(&m_stream, 0, &op, m_sampleRate, - 1024, - paNoFlag, processStatic, this); - m_bufferSize = 1024; - } - - if (err != paNoError) { - cerr << "ERROR: AudioPortAudioTarget: Failed to open PortAudio stream: " << Pa_GetErrorText(err) << endl; - cerr << "Note: device ID was " << op.device << endl; - m_stream = 0; - Pa_Terminate(); - return; - } - - const PaStreamInfo *info = Pa_GetStreamInfo(m_stream); - m_latency = int(info->outputLatency * m_sampleRate + 0.001); - if (m_bufferSize < m_latency) m_bufferSize = m_latency; - - cerr << "PortAudio latency = " << m_latency << " frames" << endl; - - err = Pa_StartStream(m_stream); - - if (err != paNoError) { - cerr << "ERROR: AudioPortAudioTarget: Failed to start PortAudio stream: " << Pa_GetErrorText(err) << endl; - Pa_CloseStream(m_stream); - m_stream = 0; - Pa_Terminate(); - return; - } - - if (m_source) { - cerr << "AudioPortAudioTarget: block size " << m_bufferSize << endl; - m_source->setTarget(this, m_bufferSize); - m_source->setTargetSampleRate(m_sampleRate); - m_source->setTargetPlayLatency(m_latency); - } - -#ifdef DEBUG_PORT_AUDIO_TARGET - cerr << "AudioPortAudioTarget: initialised OK" << endl; -#endif -} - -AudioPortAudioTarget::~AudioPortAudioTarget() -{ - SVDEBUG << "AudioPortAudioTarget::~AudioPortAudioTarget()" << endl; - - if (m_source) { - m_source->setTarget(0, m_bufferSize); - } - - shutdown(); - - if (m_stream) { - - SVDEBUG << "closing stream" << endl; - - PaError err; - err = Pa_CloseStream(m_stream); - if (err != paNoError) { - cerr << "ERROR: AudioPortAudioTarget: Failed to close PortAudio stream: " << Pa_GetErrorText(err) << endl; - } - - cerr << "terminating" << endl; - - err = Pa_Terminate(); - if (err != paNoError) { - cerr << "ERROR: AudioPortAudioTarget: Failed to terminate PortAudio: " << Pa_GetErrorText(err) << endl; - } - } - - m_stream = 0; - - SVDEBUG << "AudioPortAudioTarget::~AudioPortAudioTarget() done" << endl; -} - -void -AudioPortAudioTarget::shutdown() -{ -#ifdef DEBUG_PORT_AUDIO_TARGET - SVDEBUG << "AudioPortAudioTarget::shutdown" << endl; -#endif - m_done = true; -} - -bool -AudioPortAudioTarget::isOK() const -{ - return (m_stream != 0); -} - -double -AudioPortAudioTarget::getCurrentTime() const -{ - if (!m_stream) return 0.0; - else return Pa_GetStreamTime(m_stream); -} - -int -AudioPortAudioTarget::processStatic(const void *input, void *output, - unsigned long nframes, - const PaStreamCallbackTimeInfo *timeInfo, - PaStreamCallbackFlags flags, void *data) -{ - return ((AudioPortAudioTarget *)data)->process(input, output, - nframes, timeInfo, - flags); -} - -void -AudioPortAudioTarget::sourceModelReplaced() -{ - m_source->setTargetSampleRate(m_sampleRate); -} - -int -AudioPortAudioTarget::process(const void *, void *outputBuffer, - sv_frame_t nframes, - const PaStreamCallbackTimeInfo *, - PaStreamCallbackFlags) -{ -#ifdef DEBUG_AUDIO_PORT_AUDIO_TARGET - SVDEBUG << "AudioPortAudioTarget::process(" << nframes << ")" << endl; -#endif - - if (!m_source || m_done) { -#ifdef DEBUG_AUDIO_PORT_AUDIO_TARGET - SVDEBUG << "AudioPortAudioTarget::process: Doing nothing, no source or application done" << endl; -#endif - return 0; - } - - if (!m_prioritySet) { -#ifndef _WIN32 - sched_param param; - param.sched_priority = 20; - if (pthread_setschedparam(pthread_self(), SCHED_RR, ¶m)) { - SVDEBUG << "AudioPortAudioTarget: NOTE: couldn't set RT scheduling class" << endl; - } else { - SVDEBUG << "AudioPortAudioTarget: NOTE: successfully set RT scheduling class" << endl; - } -#endif - m_prioritySet = true; - } - - float *output = (float *)outputBuffer; - - assert(nframes <= m_bufferSize); - - static float **tmpbuf = 0; - static int tmpbufch = 0; - static int tmpbufsz = 0; - - int sourceChannels = m_source->getSourceChannelCount(); - - // Because we offer pan, we always want at least 2 channels - if (sourceChannels < 2) sourceChannels = 2; - - if (!tmpbuf || tmpbufch != sourceChannels || int(tmpbufsz) < m_bufferSize) { - - if (tmpbuf) { - for (int i = 0; i < tmpbufch; ++i) { - delete[] tmpbuf[i]; - } - delete[] tmpbuf; - } - - tmpbufch = sourceChannels; - tmpbufsz = m_bufferSize; - tmpbuf = new float *[tmpbufch]; - - for (int i = 0; i < tmpbufch; ++i) { - tmpbuf[i] = new float[tmpbufsz]; - } - } - - sv_frame_t received = m_source->getSourceSamples(nframes, tmpbuf); - - float peakLeft = 0.0, peakRight = 0.0; - - for (int ch = 0; ch < 2; ++ch) { - - float peak = 0.0; - - if (ch < sourceChannels) { - - // PortAudio samples are interleaved - for (int i = 0; i < nframes; ++i) { - if (i < received) { - output[i * 2 + ch] = tmpbuf[ch][i] * m_outputGain; - float sample = fabsf(output[i * 2 + ch]); - if (sample > peak) peak = sample; - } else { - output[i * 2 + ch] = 0; - } - } - - } else if (ch == 1 && sourceChannels == 1) { - - for (int i = 0; i < nframes; ++i) { - if (i < received) { - output[i * 2 + ch] = tmpbuf[0][i] * m_outputGain; - float sample = fabsf(output[i * 2 + ch]); - if (sample > peak) peak = sample; - } else { - output[i * 2 + ch] = 0; - } - } - - } else { - for (int i = 0; i < nframes; ++i) { - output[i * 2 + ch] = 0; - } - } - - if (ch == 0) peakLeft = peak; - if (ch > 0 || sourceChannels == 1) peakRight = peak; - } - - m_source->setOutputLevels(peakLeft, peakRight); - - if (Pa_GetStreamCpuLoad(m_stream) > 0.7) { - if (m_source) m_source->audioProcessingOverload(); - } - - return 0; -} - -#endif /* HAVE_PORTAUDIO */ -
--- a/audioio/AudioPortAudioTarget.h Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,71 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifndef _AUDIO_PORT_AUDIO_TARGET_H_ -#define _AUDIO_PORT_AUDIO_TARGET_H_ - -#ifdef HAVE_PORTAUDIO_2_0 - -// This code requires PortAudio v19 -- it won't work with v18. - -#include <portaudio.h> - -#include <QObject> - -#include "AudioCallbackPlayTarget.h" - -#include "base/BaseTypes.h" - -class AudioCallbackPlaySource; - -class AudioPortAudioTarget : public AudioCallbackPlayTarget -{ - Q_OBJECT - -public: - AudioPortAudioTarget(AudioCallbackPlaySource *source); - virtual ~AudioPortAudioTarget(); - - virtual void shutdown(); - - virtual bool isOK() const; - - virtual double getCurrentTime() const; - -public slots: - virtual void sourceModelReplaced(); - -protected: - int process(const void *input, void *output, sv_frame_t frames, - const PaStreamCallbackTimeInfo *timeInfo, - PaStreamCallbackFlags statusFlags); - - static int processStatic(const void *, void *, unsigned long, - const PaStreamCallbackTimeInfo *, - PaStreamCallbackFlags, void *); - - PaStream *m_stream; - - int m_bufferSize; - int m_sampleRate; - int m_latency; - bool m_prioritySet; - bool m_done; -}; - -#endif /* HAVE_PORTAUDIO */ - -#endif -
--- a/audioio/AudioPulseAudioTarget.cpp Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,416 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2008 QMUL. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifdef HAVE_LIBPULSE - -#include "AudioPulseAudioTarget.h" -#include "AudioCallbackPlaySource.h" - -#include <QMutexLocker> - -#include <iostream> -#include <cassert> -#include <cmath> - -#define DEBUG_AUDIO_PULSE_AUDIO_TARGET 1 -//#define DEBUG_AUDIO_PULSE_AUDIO_TARGET_PLAY 1 - -AudioPulseAudioTarget::AudioPulseAudioTarget(AudioCallbackPlaySource *source) : - AudioCallbackPlayTarget(source), - m_mutex(QMutex::Recursive), - m_loop(0), - m_api(0), - m_context(0), - m_stream(0), - m_loopThread(0), - m_bufferSize(0), - m_sampleRate(0), - m_latency(0), - m_done(false) -{ -#ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET - cerr << "AudioPulseAudioTarget: Initialising for PulseAudio" << endl; -#endif - - m_loop = pa_mainloop_new(); - if (!m_loop) { - cerr << "ERROR: AudioPulseAudioTarget: Failed to create main loop" << endl; - return; - } - - m_api = pa_mainloop_get_api(m_loop); - - //!!! handle signals how? - - m_bufferSize = 20480; - m_sampleRate = 44100; - if (m_source && (m_source->getSourceSampleRate() != 0)) { - m_sampleRate = int(m_source->getSourceSampleRate()); - } - m_spec.rate = m_sampleRate; - m_spec.channels = 2; - m_spec.format = PA_SAMPLE_FLOAT32NE; - -#ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET - cerr << "AudioPulseAudioTarget: Creating context" << endl; -#endif - - m_context = pa_context_new(m_api, source->getClientName().toLocal8Bit().data()); - if (!m_context) { - cerr << "ERROR: AudioPulseAudioTarget: Failed to create context object" << endl; - return; - } - - pa_context_set_state_callback(m_context, contextStateChangedStatic, this); - -#ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET - cerr << "AudioPulseAudioTarget: Connecting to default server..." << endl; -#endif - - pa_context_connect(m_context, 0, // default server - (pa_context_flags_t)PA_CONTEXT_NOAUTOSPAWN, 0); - -#ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET - cerr << "AudioPulseAudioTarget: Starting main loop" << endl; -#endif - - m_loopThread = new MainLoopThread(m_loop); - m_loopThread->start(); - -#ifdef DEBUG_PULSE_AUDIO_TARGET - cerr << "AudioPulseAudioTarget: initialised OK" << endl; -#endif -} - -AudioPulseAudioTarget::~AudioPulseAudioTarget() -{ - SVDEBUG << "AudioPulseAudioTarget::~AudioPulseAudioTarget()" << endl; - - if (m_source) { - m_source->setTarget(0, m_bufferSize); - } - - shutdown(); - - QMutexLocker locker(&m_mutex); - - if (m_stream) pa_stream_unref(m_stream); - - if (m_context) pa_context_unref(m_context); - - if (m_loop) { - pa_signal_done(); - pa_mainloop_free(m_loop); - } - - m_stream = 0; - m_context = 0; - m_loop = 0; - - SVDEBUG << "AudioPulseAudioTarget::~AudioPulseAudioTarget() done" << endl; -} - -void -AudioPulseAudioTarget::shutdown() -{ - m_done = true; -} - -bool -AudioPulseAudioTarget::isOK() const -{ - return (m_context != 0); -} - -double -AudioPulseAudioTarget::getCurrentTime() const -{ - if (!m_stream) return 0.0; - - pa_usec_t usec = 0; - pa_stream_get_time(m_stream, &usec); - return double(usec) / 1000000.0; -} - -void -AudioPulseAudioTarget::sourceModelReplaced() -{ - m_source->setTargetSampleRate(m_sampleRate); -} - -void -AudioPulseAudioTarget::streamWriteStatic(pa_stream *, - size_t length, - void *data) -{ - AudioPulseAudioTarget *target = (AudioPulseAudioTarget *)data; - -// assert(stream == target->m_stream); - - target->streamWrite(length); -} - -void -AudioPulseAudioTarget::streamWrite(sv_frame_t requested) -{ -#ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET_PLAY - cout << "AudioPulseAudioTarget::streamWrite(" << requested << ")" << endl; -#endif - if (m_done) return; - - QMutexLocker locker(&m_mutex); - - pa_usec_t latency = 0; - int negative = 0; - if (!pa_stream_get_latency(m_stream, &latency, &negative)) { - 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 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; - - sv_frame_t nframes = requested / (sourceChannels * sizeof(float)); - - if (nframes > m_bufferSize) { - cerr << "WARNING: AudioPulseAudioTarget::streamWrite: nframes " << nframes << " > m_bufferSize " << m_bufferSize << endl; - } - -#ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET_PLAY - cout << "AudioPulseAudioTarget::streamWrite: nframes = " << nframes << endl; -#endif - - if (!tmpbuf || tmpbufch != sourceChannels || int(tmpbufsz) < nframes) { - - if (tmpbuf) { - for (int i = 0; i < tmpbufch; ++i) { - delete[] tmpbuf[i]; - } - delete[] tmpbuf; - } - - if (output) { - delete[] output; - } - - tmpbufch = sourceChannels; - tmpbufsz = nframes; - tmpbuf = new float *[tmpbufch]; - - for (int i = 0; i < tmpbufch; ++i) { - tmpbuf[i] = new float[tmpbufsz]; - } - - output = new float[tmpbufsz * tmpbufch]; - } - - sv_frame_t received = m_source->getSourceSamples(nframes, tmpbuf); - -#ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET_PLAY - cerr << "requested " << nframes << ", received " << received << endl; - - if (received < nframes) { - cerr << "*** WARNING: Wrong number of frames received" << endl; - } -#endif - - float peakLeft = 0.0, peakRight = 0.0; - - for (int ch = 0; ch < 2; ++ch) { - - float peak = 0.0; - - // PulseAudio samples are interleaved - for (int i = 0; i < nframes; ++i) { - if (i < received) { - output[i * 2 + ch] = tmpbuf[ch][i] * m_outputGain; - float sample = fabsf(output[i * 2 + ch]); - if (sample > peak) peak = sample; - } else { - output[i * 2 + ch] = 0; - } - } - - if (ch == 0) peakLeft = peak; - if (ch == 1) peakRight = peak; - } - -#ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET_PLAY - SVDEBUG << "calling pa_stream_write with " - << nframes * tmpbufch * sizeof(float) << " bytes" << endl; -#endif - - pa_stream_write(m_stream, output, - size_t(nframes * tmpbufch * sizeof(float)), - 0, 0, PA_SEEK_RELATIVE); - - m_source->setOutputLevels(peakLeft, peakRight); - - return; -} - -void -AudioPulseAudioTarget::streamStateChangedStatic(pa_stream *, - void *data) -{ - AudioPulseAudioTarget *target = (AudioPulseAudioTarget *)data; - -// assert(stream == target->m_stream); - - target->streamStateChanged(); -} - -void -AudioPulseAudioTarget::streamStateChanged() -{ -#ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET - SVDEBUG << "AudioPulseAudioTarget::streamStateChanged" << endl; -#endif - QMutexLocker locker(&m_mutex); - - switch (pa_stream_get_state(m_stream)) { - - case PA_STREAM_UNCONNECTED: - case PA_STREAM_CREATING: - case PA_STREAM_TERMINATED: - break; - - case PA_STREAM_READY: - { - SVDEBUG << "AudioPulseAudioTarget::streamStateChanged: Ready" << endl; - - pa_usec_t latency = 0; - int negative = 0; - if (pa_stream_get_latency(m_stream, &latency, &negative)) { - cerr << "AudioPulseAudioTarget::streamStateChanged: Failed to query latency" << endl; - } - cerr << "Latency = " << latency << " usec" << endl; - int latframes = int(double(latency) / 1000000.0 * m_sampleRate); - cerr << "that's " << latframes << " frames" << endl; - - const pa_buffer_attr *attr; - if (!(attr = pa_stream_get_buffer_attr(m_stream))) { - SVDEBUG << "AudioPulseAudioTarget::streamStateChanged: Cannot query stream buffer attributes" << endl; - m_source->setTarget(this, m_bufferSize); - m_source->setTargetSampleRate(m_sampleRate); - if (latframes != 0) m_source->setTargetPlayLatency(latframes); - } else { - int targetLength = attr->tlength; - SVDEBUG << "AudioPulseAudioTarget::streamStateChanged: stream target length = " << targetLength << endl; - m_source->setTarget(this, targetLength); - m_source->setTargetSampleRate(m_sampleRate); - if (latframes == 0) latframes = targetLength; - cerr << "latency = " << latframes << endl; - m_source->setTargetPlayLatency(latframes); - } - } - break; - - case PA_STREAM_FAILED: - default: - cerr << "AudioPulseAudioTarget::streamStateChanged: Error: " - << pa_strerror(pa_context_errno(m_context)) << endl; - //!!! do something... - break; - } -} - -void -AudioPulseAudioTarget::contextStateChangedStatic(pa_context *, - void *data) -{ - AudioPulseAudioTarget *target = (AudioPulseAudioTarget *)data; - -// assert(context == target->m_context); - - target->contextStateChanged(); -} - -void -AudioPulseAudioTarget::contextStateChanged() -{ -#ifdef DEBUG_AUDIO_PULSE_AUDIO_TARGET - SVDEBUG << "AudioPulseAudioTarget::contextStateChanged" << endl; -#endif - QMutexLocker locker(&m_mutex); - - switch (pa_context_get_state(m_context)) { - - case PA_CONTEXT_UNCONNECTED: - case PA_CONTEXT_CONNECTING: - case PA_CONTEXT_AUTHORIZING: - case PA_CONTEXT_SETTING_NAME: - break; - - case PA_CONTEXT_READY: - SVDEBUG << "AudioPulseAudioTarget::contextStateChanged: Ready" - << endl; - - m_stream = pa_stream_new(m_context, "stream", &m_spec, 0); - assert(m_stream); //!!! - - pa_stream_set_state_callback(m_stream, streamStateChangedStatic, this); - pa_stream_set_write_callback(m_stream, streamWriteStatic, this); - pa_stream_set_overflow_callback(m_stream, streamOverflowStatic, this); - pa_stream_set_underflow_callback(m_stream, streamUnderflowStatic, this); - if (pa_stream_connect_playback - (m_stream, 0, 0, - pa_stream_flags_t(PA_STREAM_INTERPOLATE_TIMING | - PA_STREAM_AUTO_TIMING_UPDATE), - 0, 0)) { //??? return value - cerr << "AudioPulseAudioTarget: Failed to connect playback stream" << endl; - } - - break; - - case PA_CONTEXT_TERMINATED: - SVDEBUG << "AudioPulseAudioTarget::contextStateChanged: Terminated" << endl; - //!!! do something... - break; - - case PA_CONTEXT_FAILED: - default: - cerr << "AudioPulseAudioTarget::contextStateChanged: Error: " - << pa_strerror(pa_context_errno(m_context)) << endl; - //!!! do something... - break; - } -} - -void -AudioPulseAudioTarget::streamOverflowStatic(pa_stream *, void *) -{ - SVDEBUG << "AudioPulseAudioTarget::streamOverflowStatic: Overflow!" << endl; -} - -void -AudioPulseAudioTarget::streamUnderflowStatic(pa_stream *, void *data) -{ - SVDEBUG << "AudioPulseAudioTarget::streamUnderflowStatic: Underflow!" << endl; - AudioPulseAudioTarget *target = (AudioPulseAudioTarget *)data; - if (target && target->m_source) { - target->m_source->audioProcessingOverload(); - } -} - -#endif /* HAVE_PULSEAUDIO */ -
--- a/audioio/AudioPulseAudioTarget.h Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,91 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2008 QMUL. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifndef _AUDIO_PULSE_AUDIO_TARGET_H_ -#define _AUDIO_PULSE_AUDIO_TARGET_H_ - -#ifdef HAVE_LIBPULSE - -#include <pulse/pulseaudio.h> - -#include <QObject> -#include <QMutex> -#include "base/Thread.h" - -#include "AudioCallbackPlayTarget.h" - -class AudioCallbackPlaySource; - -class AudioPulseAudioTarget : public AudioCallbackPlayTarget -{ - Q_OBJECT - -public: - AudioPulseAudioTarget(AudioCallbackPlaySource *source); - virtual ~AudioPulseAudioTarget(); - - virtual void shutdown(); - - virtual bool isOK() const; - - virtual double getCurrentTime() const; - -public slots: - virtual void sourceModelReplaced(); - -protected: - void streamWrite(sv_frame_t); - void streamStateChanged(); - void contextStateChanged(); - - static void streamWriteStatic(pa_stream *, size_t, void *); - static void streamStateChangedStatic(pa_stream *, void *); - static void streamOverflowStatic(pa_stream *, void *); - static void streamUnderflowStatic(pa_stream *, void *); - static void contextStateChangedStatic(pa_context *, void *); - - QMutex m_mutex; - - class MainLoopThread : public Thread - { - public: - MainLoopThread(pa_mainloop *loop) : Thread(NonRTThread), m_loop(loop) { } //!!! or RTThread - virtual void run() { - int rv = 0; - pa_mainloop_run(m_loop, &rv); //!!! check return value from this, and rv - } - - private: - pa_mainloop *m_loop; - }; - - pa_mainloop *m_loop; - pa_mainloop_api *m_api; - pa_context *m_context; - pa_stream *m_stream; - pa_sample_spec m_spec; - - MainLoopThread *m_loopThread; - - int m_bufferSize; - int m_sampleRate; - int m_latency; - bool m_done; -}; - -#endif /* HAVE_PULSEAUDIO */ - -#endif -
--- a/audioio/AudioTargetFactory.cpp Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,164 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#include "AudioTargetFactory.h" - -#include "AudioJACKTarget.h" -#include "AudioPortAudioTarget.h" -#include "AudioPulseAudioTarget.h" - -#include "AudioCallbackPlayTarget.h" - -#include <QCoreApplication> - -#include <iostream> - -AudioTargetFactory * -AudioTargetFactory::m_instance = 0; - -AudioTargetFactory * -AudioTargetFactory::getInstance() -{ - if (!m_instance) m_instance = new AudioTargetFactory(); - return m_instance; -} - -AudioTargetFactory::AudioTargetFactory() -{ -} - -std::vector<QString> -AudioTargetFactory::getCallbackTargetNames(bool includeAuto) const -{ - std::vector<QString> names; - if (includeAuto) names.push_back("auto"); - -#ifdef HAVE_JACK - names.push_back("jack"); -#endif - -#ifdef HAVE_LIBPULSE - names.push_back("pulse"); -#endif - -#ifdef HAVE_PORTAUDIO_2_0 - names.push_back("port"); -#endif - - return names; -} - -QString -AudioTargetFactory::getCallbackTargetDescription(QString name) const -{ - if (name == "auto") { - return QCoreApplication::translate("AudioTargetFactory", - "(auto)"); - } - if (name == "jack") { - return QCoreApplication::translate("AudioTargetFactory", - "JACK Audio Connection Kit"); - } - if (name == "pulse") { - return QCoreApplication::translate("AudioTargetFactory", - "PulseAudio Server"); - } - if (name == "port") { - return QCoreApplication::translate("AudioTargetFactory", - "Default Soundcard Device"); - } - - return "(unknown)"; -} - -QString -AudioTargetFactory::getDefaultCallbackTarget() const -{ - if (m_default == "") return "auto"; - return m_default; -} - -bool -AudioTargetFactory::isAutoCallbackTarget(QString name) const -{ - return (name == "auto" || name == ""); -} - -void -AudioTargetFactory::setDefaultCallbackTarget(QString target) -{ - m_default = target; -} - -AudioCallbackPlayTarget * -AudioTargetFactory::createCallbackTarget(AudioCallbackPlaySource *source) -{ - AudioCallbackPlayTarget *target = 0; - - if (m_default != "" && m_default != "auto") { - -#ifdef HAVE_JACK - if (m_default == "jack") target = new AudioJACKTarget(source); -#endif - -#ifdef HAVE_LIBPULSE - if (m_default == "pulse") target = new AudioPulseAudioTarget(source); -#endif - -#ifdef HAVE_PORTAUDIO_2_0 - if (m_default == "port") target = new AudioPortAudioTarget(source); -#endif - - if (!target || !target->isOK()) { - cerr << "WARNING: AudioTargetFactory::createCallbackTarget: Failed to open the requested target (\"" << m_default << "\")" << endl; - delete target; - return 0; - } else { - return target; - } - } - -#ifdef HAVE_JACK - target = new AudioJACKTarget(source); - if (target->isOK()) return target; - else { - cerr << "WARNING: AudioTargetFactory::createCallbackTarget: Failed to open JACK target" << endl; - delete target; - } -#endif - -#ifdef HAVE_LIBPULSE - target = new AudioPulseAudioTarget(source); - if (target->isOK()) return target; - else { - cerr << "WARNING: AudioTargetFactory::createCallbackTarget: Failed to open PulseAudio target" << endl; - delete target; - } -#endif - -#ifdef HAVE_PORTAUDIO_2_0 - target = new AudioPortAudioTarget(source); - if (target->isOK()) return target; - else { - cerr << "WARNING: AudioTargetFactory::createCallbackTarget: Failed to open PortAudio target" << endl; - delete target; - } -#endif - - cerr << "WARNING: AudioTargetFactory::createCallbackTarget: No suitable targets available" << endl; - return 0; -} - -
--- a/audioio/AudioTargetFactory.h Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,47 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifndef _AUDIO_TARGET_FACTORY_H_ -#define _AUDIO_TARGET_FACTORY_H_ - -#include <vector> -#include <QString> - -#include "base/Debug.h" - -class AudioCallbackPlaySource; -class AudioCallbackPlayTarget; - -class AudioTargetFactory -{ -public: - static AudioTargetFactory *getInstance(); - - std::vector<QString> getCallbackTargetNames(bool includeAuto = true) const; - QString getCallbackTargetDescription(QString name) const; - QString getDefaultCallbackTarget() const; - bool isAutoCallbackTarget(QString name) const; - void setDefaultCallbackTarget(QString name); - - AudioCallbackPlayTarget *createCallbackTarget(AudioCallbackPlaySource *); - -protected: - AudioTargetFactory(); - static AudioTargetFactory *m_instance; - QString m_default; -}; - -#endif -
--- a/audioio/ClipMixer.cpp Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,248 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam, 2006-2014 QMUL. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#include "ClipMixer.h" - -#include <sndfile.h> -#include <cmath> - -#include "base/Debug.h" - -//#define DEBUG_CLIP_MIXER 1 - -ClipMixer::ClipMixer(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize) : - m_channels(channels), - m_sampleRate(sampleRate), - m_blockSize(blockSize), - m_clipData(0), - m_clipLength(0), - m_clipF0(0), - m_clipRate(0) -{ -} - -ClipMixer::~ClipMixer() -{ - if (m_clipData) free(m_clipData); -} - -void -ClipMixer::setChannelCount(int channels) -{ - m_channels = channels; -} - -bool -ClipMixer::loadClipData(QString path, double f0, double level) -{ - if (m_clipData) { - cerr << "ClipMixer::loadClipData: Already have clip loaded" << endl; - return false; - } - - SF_INFO info; - SNDFILE *file; - float *tmpFrames; - sv_frame_t i; - - info.format = 0; - file = sf_open(path.toLocal8Bit().data(), SFM_READ, &info); - if (!file) { - cerr << "ClipMixer::loadClipData: Failed to open file path \"" - << path << "\": " << sf_strerror(file) << endl; - return false; - } - - tmpFrames = (float *)malloc(info.frames * info.channels * sizeof(float)); - if (!tmpFrames) { - cerr << "ClipMixer::loadClipData: malloc(" << info.frames * info.channels * sizeof(float) << ") failed" << endl; - return false; - } - - sf_readf_float(file, tmpFrames, info.frames); - sf_close(file); - - m_clipData = (float *)malloc(info.frames * sizeof(float)); - if (!m_clipData) { - cerr << "ClipMixer::loadClipData: malloc(" << info.frames * sizeof(float) << ") failed" << endl; - free(tmpFrames); - return false; - } - - for (i = 0; i < info.frames; ++i) { - int j; - m_clipData[i] = 0.0f; - for (j = 0; j < info.channels; ++j) { - m_clipData[i] += tmpFrames[i * info.channels + j] * float(level); - } - } - - free(tmpFrames); - - m_clipLength = info.frames; - m_clipF0 = f0; - m_clipRate = info.samplerate; - - return true; -} - -void -ClipMixer::reset() -{ - m_playing.clear(); -} - -double -ClipMixer::getResampleRatioFor(double frequency) -{ - if (!m_clipData || !m_clipRate) return 1.0; - double pitchRatio = m_clipF0 / frequency; - double resampleRatio = m_sampleRate / m_clipRate; - return pitchRatio * resampleRatio; -} - -sv_frame_t -ClipMixer::getResampledClipDuration(double frequency) -{ - return sv_frame_t(ceil(double(m_clipLength) * getResampleRatioFor(frequency))); -} - -void -ClipMixer::mix(float **toBuffers, - float gain, - std::vector<NoteStart> newNotes, - std::vector<NoteEnd> endingNotes) -{ - foreach (NoteStart note, newNotes) { - if (note.frequency > 20 && - note.frequency < 5000) { - m_playing.push_back(note); - } - } - - std::vector<NoteStart> remaining; - - float *levels = new float[m_channels]; - -#ifdef DEBUG_CLIP_MIXER - cerr << "ClipMixer::mix: have " << m_playing.size() << " playing note(s)" - << " and " << endingNotes.size() << " note(s) ending here" - << endl; -#endif - - foreach (NoteStart note, m_playing) { - - for (int c = 0; c < m_channels; ++c) { - levels[c] = note.level * gain; - } - if (note.pan != 0.0 && m_channels == 2) { - levels[0] *= 1.0f - note.pan; - levels[1] *= note.pan + 1.0f; - } - - sv_frame_t start = note.frameOffset; - sv_frame_t durationHere = m_blockSize; - if (start > 0) durationHere = m_blockSize - start; - - bool ending = false; - - foreach (NoteEnd end, endingNotes) { - if (end.frequency == note.frequency && - end.frameOffset >= start && - end.frameOffset <= m_blockSize) { - ending = true; - durationHere = end.frameOffset; - if (start > 0) durationHere = end.frameOffset - start; - break; - } - } - - sv_frame_t clipDuration = getResampledClipDuration(note.frequency); - if (start + clipDuration > 0) { - if (start < 0 && start + clipDuration < durationHere) { - durationHere = start + clipDuration; - } - if (durationHere > 0) { - mixNote(toBuffers, - levels, - note.frequency, - start < 0 ? -start : 0, - start > 0 ? start : 0, - durationHere, - ending); - } - } - - if (!ending) { - NoteStart adjusted = note; - adjusted.frameOffset -= m_blockSize; - remaining.push_back(adjusted); - } - } - - delete[] levels; - - m_playing = remaining; -} - -void -ClipMixer::mixNote(float **toBuffers, - float *levels, - float frequency, - sv_frame_t sourceOffset, - sv_frame_t targetOffset, - sv_frame_t sampleCount, - bool isEnd) -{ - if (!m_clipData) return; - - double ratio = getResampleRatioFor(frequency); - - double releaseTime = 0.01; - sv_frame_t releaseSampleCount = sv_frame_t(round(releaseTime * m_sampleRate)); - if (releaseSampleCount > sampleCount) { - releaseSampleCount = sampleCount; - } - double releaseFraction = 1.0/double(releaseSampleCount); - - for (sv_frame_t i = 0; i < sampleCount; ++i) { - - sv_frame_t s = sourceOffset + i; - - 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" - 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 - double(osi)); - } - - if (isEnd && i + releaseSampleCount > sampleCount) { - value *= releaseFraction * double(sampleCount - i); // linear ramp for release - } - - for (int c = 0; c < m_channels; ++c) { - toBuffers[c][targetOffset + i] += float(levels[c] * value); - } - } -} - -
--- a/audioio/ClipMixer.h Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,94 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 Chris Cannam, 2006-2014 QMUL. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifndef CLIP_MIXER_H -#define CLIP_MIXER_H - -#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 - * musician's sense.) This can mix any number of notes of arbitrary - * frequency, so long as they all use the same sample clip. - */ - -class ClipMixer -{ -public: - ClipMixer(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize); - ~ClipMixer(); - - void setChannelCount(int channels); - - /** - * Load a sample clip from a wav file. This can only happen once: - * construct a new ClipMixer if you want a different clip. The - * clip was recorded at a pitch with fundamental frequency clipF0, - * and should be scaled by level (in the range 0-1) when playing - * back. - */ - bool loadClipData(QString clipFilePath, double clipF0, double level); - - void reset(); // discarding any playing notes - - struct NoteStart { - 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 { - sv_frame_t frameOffset; // in current processing block - float frequency; // matching note start - }; - - void mix(float **toBuffers, - float gain, - std::vector<NoteStart> newNotes, - std::vector<NoteEnd> endingNotes); - -private: - int m_channels; - sv_samplerate_t m_sampleRate; - sv_frame_t m_blockSize; - - QString m_clipPath; - - float *m_clipData; - sv_frame_t m_clipLength; - double m_clipF0; - sv_samplerate_t m_clipRate; - - std::vector<NoteStart> m_playing; - - double getResampleRatioFor(double frequency); - sv_frame_t getResampledClipDuration(double frequency); - - void mixNote(float **toBuffers, - float *levels, - float frequency, - sv_frame_t sourceOffset, // within resampled note - sv_frame_t targetOffset, // within target buffer - sv_frame_t sampleCount, - bool isEnd); -}; - - -#endif
--- a/audioio/ContinuousSynth.cpp Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,149 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#include "ContinuousSynth.h" - -#include "base/Debug.h" -#include "system/System.h" - -#include <cmath> - -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(-1.0), - m_phase(0.0), - m_wavetype(waveType) // 0: 3 sinusoids, 1: 1 sinusoid, 2: sawtooth, 3: square -{ -} - -ContinuousSynth::~ContinuousSynth() -{ -} - -void -ContinuousSynth::reset() -{ - m_phase = 0; -} - -void -ContinuousSynth::mix(float **toBuffers, float gain, float pan, float f0f) -{ - double f0(f0f); - if (f0 == 0.0) f0 = m_prevF0; - - bool wasOn = (m_prevF0 > 0.0); - bool nowOn = (f0 > 0.0); - - if (!nowOn && !wasOn) { - m_phase = 0; - return; - } - - sv_frame_t fadeLength = 100; - - float *levels = new float[m_channels]; - - for (int c = 0; c < m_channels; ++c) { - levels[c] = gain * 0.5f; // scale gain otherwise too loud compared to source - } - if (pan != 0.0 && m_channels == 2) { - levels[0] *= 1.0f - pan; - levels[1] *= pan + 1.0f; - } - -// cerr << "ContinuousSynth::mix: f0 = " << f0 << " (from " << m_prevF0 << "), phase = " << m_phase << endl; - - 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) * double(i)) / double(fadeLength); - } - - double phasor = (fHere * 2 * M_PI) / m_sampleRate; - - m_phase = m_phase + phasor; - - int harmonics = int((m_sampleRate / 4) / fHere - 1); - if (harmonics < 1) harmonics = 1; - - switch (m_wavetype) { - case 1: - harmonics = 1; - break; - case 2: - break; - case 3: - break; - default: - harmonics = 3; - break; - } - - for (int h = 0; h < harmonics; ++h) { - - double v = 0; - double hn = 0; - double hp = 0; - - switch (m_wavetype) { - case 1: // single sinusoid - v = sin(m_phase); - break; - case 2: // sawtooth - if (h != 0) { - hn = h + 1; - hp = m_phase * hn; - v = -(1.0 / M_PI) * sin(hp) / hn; - } else { - v = 0.5; - } - break; - case 3: // square - hn = h*2 + 1; - hp = m_phase * hn; - v = sin(hp) / hn; - break; - default: // 3 sinusoids - hn = h + 1; - hp = m_phase * hn; - v = sin(hp) / hn; - break; - } - - if (!wasOn && i < fadeLength) { - // fade in - v = v * (double(i) / double(fadeLength)); - } else if (!nowOn) { - // fade out - if (i > fadeLength) v = 0; - else v = v * (1.0 - (double(i) / double(fadeLength))); - } - - for (int c = 0; c < m_channels; ++c) { - toBuffers[c][i] += float(levels[c] * v); - } - } - } - - m_prevF0 = f0; - - delete[] levels; -} -
--- a/audioio/ContinuousSynth.h Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,65 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#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 - * block, or may be switched on or off. - */ - -class ContinuousSynth -{ -public: - ContinuousSynth(int channels, sv_samplerate_t sampleRate, sv_frame_t blockSize, int waveType); - ~ContinuousSynth(); - - void setChannelCount(int channels); - - void reset(); - - /** - * Mix in a signal to be heard at the given fundamental - * frequency. Any oscillator state will be maintained between - * process calls so as to provide a continuous sound. The f0 value - * may vary between calls. - * - * Supply f0 equal to 0 if you want to maintain the f0 from the - * previous block (without having to remember what it was). - * - * Supply f0 less than 0 for silence. You should continue to call - * this even when the signal is silent if you want to ensure the - * sound switches on and off cleanly. - */ - void mix(float **toBuffers, - float gain, - float pan, - float f0); - -private: - int m_channels; - sv_samplerate_t m_sampleRate; - sv_frame_t m_blockSize; - - double m_prevF0; - double m_phase; - - int m_wavetype; -}; - -#endif
--- a/audioio/PlaySpeedRangeMapper.cpp Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,101 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 QMUL. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#include "PlaySpeedRangeMapper.h" - -#include <iostream> -#include <cmath> - -// PlaySpeedRangeMapper maps a position in the range [0,120] on to a -// play speed factor on a logarithmic scale in the range 0.125 -> -// 8. This ensures that the desirable speed factors 0.25, 0.5, 1, 2, -// and 4 are all mapped to exact positions (respectively 20, 40, 60, -// 80, 100). - -// Note that the "factor" referred to below is a play speed factor -// (higher = faster, 1.0 = normal speed), the "value" is a percentage -// (higher = faster, 100 = normal speed), and the "position" is an -// integer step on the dial's scale (0-120, 60 = centre). - -PlaySpeedRangeMapper::PlaySpeedRangeMapper() : - m_minpos(0), - m_maxpos(120) -{ -} - -int -PlaySpeedRangeMapper::getPositionForValue(double value) const -{ - // value is percent - double factor = getFactorForValue(value); - int position = getPositionForFactor(factor); - return position; -} - -int -PlaySpeedRangeMapper::getPositionForValueUnclamped(double value) const -{ - // We don't really provide this - return getPositionForValue(value); -} - -double -PlaySpeedRangeMapper::getValueForPosition(int position) const -{ - double factor = getFactorForPosition(position); - double pc = getValueForFactor(factor); - return pc; -} - -double -PlaySpeedRangeMapper::getValueForPositionUnclamped(int position) const -{ - // We don't really provide this - return getValueForPosition(position); -} - -double -PlaySpeedRangeMapper::getValueForFactor(double factor) const -{ - return factor * 100.0; -} - -double -PlaySpeedRangeMapper::getFactorForValue(double value) const -{ - return value / 100.0; -} - -int -PlaySpeedRangeMapper::getPositionForFactor(double factor) const -{ - if (factor == 0) return m_minpos; - int pos = int(lrint((log2(factor) + 3.0) * 20.0)); - if (pos < m_minpos) pos = m_minpos; - if (pos > m_maxpos) pos = m_maxpos; - return pos; -} - -double -PlaySpeedRangeMapper::getFactorForPosition(int position) const -{ - return pow(2.0, double(position) * 0.05 - 3.0); -} - -QString -PlaySpeedRangeMapper::getUnit() const -{ - return "%"; -}
--- a/audioio/PlaySpeedRangeMapper.h Mon Jul 13 14:39:41 2015 +0100 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,49 +0,0 @@ -/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ - -/* - Sonic Visualiser - An audio file viewer and annotation editor. - Centre for Digital Music, Queen Mary, University of London. - This file copyright 2006 QMUL. - - This program is free software; you can redistribute it and/or - modify it under the terms of the GNU General Public License as - published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. See the file - COPYING included with this distribution for more information. -*/ - -#ifndef _PLAY_SPEED_RANGE_MAPPER_H_ -#define _PLAY_SPEED_RANGE_MAPPER_H_ - -#include "base/RangeMapper.h" - -class PlaySpeedRangeMapper : public RangeMapper -{ -public: - PlaySpeedRangeMapper(); - - int getMinPosition() const { return m_minpos; } - int getMaxPosition() const { return m_maxpos; } - - virtual int getPositionForValue(double value) const; - virtual int getPositionForValueUnclamped(double value) const; - - virtual double getValueForPosition(int position) const; - virtual double getValueForPositionUnclamped(int position) const; - - int getPositionForFactor(double factor) const; - double getValueForFactor(double factor) const; - - double getFactorForPosition(int position) const; - double getFactorForValue(double value) const; - - virtual QString getUnit() const; - -protected: - int m_minpos; - int m_maxpos; -}; - - -#endif
--- a/configure.ac Mon Jul 13 14:39:41 2015 +0100 +++ b/configure.ac Tue Aug 04 13:27:42 2015 +0100 @@ -88,7 +88,7 @@ SV_MODULE_REQUIRED([rubberband],[rubberband],[rubberband/RubberBandStretcher.h],[rubberband],[rubberband_new]) SV_MODULE_OPTIONAL([liblo],[],[lo/lo.h],[lo],[lo_address_new]) -SV_MODULE_OPTIONAL([portaudio_2_0],[portaudio-2.0 >= 19],[portaudio.h],[portaudio],[Pa_IsFormatSupported]) +SV_MODULE_OPTIONAL([portaudio],[portaudio-2.0 >= 19],[portaudio.h],[portaudio],[Pa_IsFormatSupported]) SV_MODULE_OPTIONAL([JACK],[jack >= 0.100],[jack/jack.h],[jack],[jack_client_open]) SV_MODULE_OPTIONAL([libpulse],[libpulse >= 0.9],[pulse/pulseaudio.h],[pulse],[pa_stream_new]) SV_MODULE_OPTIONAL([lrdf],[lrdf >= 0.2],[lrdf.h],[lrdf],[lrdf_init])
--- a/framework/MainWindowBase.cpp Mon Jul 13 14:39:41 2015 +0100 +++ b/framework/MainWindowBase.cpp Tue Aug 04 13:27:42 2015 +0100 @@ -47,10 +47,8 @@ #include "widgets/ModelDataTableDialog.h" #include "widgets/InteractiveFileFinder.h" -#include "audioio/AudioCallbackPlaySource.h" -#include "audioio/AudioCallbackPlayTarget.h" -#include "audioio/AudioTargetFactory.h" -#include "audioio/PlaySpeedRangeMapper.h" +#include "audio/AudioCallbackPlaySource.h" +#include "audio/PlaySpeedRangeMapper.h" #include "data/fileio/DataFileReaderFactory.h" #include "data/fileio/PlaylistFileReader.h" #include "data/fileio/WavFileWriter.h" @@ -75,6 +73,9 @@ #include "data/osc/OSCQueue.h" #include "data/midi/MIDIInput.h" +#include <bqaudioio/SystemPlaybackTarget.h> +#include <bqaudioio/AudioFactory.h> + #include <QApplication> #include <QMessageBox> #include <QGridLayout> @@ -268,8 +269,7 @@ MainWindowBase::~MainWindowBase() { SVDEBUG << "MainWindowBase::~MainWindowBase" << endl; - if (m_playTarget) m_playTarget->shutdown(); -// delete m_playTarget; + delete m_playTarget; delete m_playSource; delete m_viewManager; delete m_oscQueue; @@ -2164,24 +2164,31 @@ { if (m_playTarget) return; + //!!! how to handle preferences +/* QSettings settings; settings.beginGroup("Preferences"); QString targetName = settings.value("audio-target", "").toString(); settings.endGroup(); - AudioTargetFactory *factory = AudioTargetFactory::getInstance(); factory->setDefaultCallbackTarget(targetName); - m_playTarget = factory->createCallbackTarget(m_playSource); +*/ + + m_playTarget = + breakfastquay::AudioFactory::createCallbackPlayTarget(m_playSource); + + m_playSource->setSystemPlaybackTarget(m_playTarget); if (!m_playTarget) { emit hideSplash(); - if (factory->isAutoCallbackTarget(targetName)) { +// if (factory->isAutoCallbackTarget(targetName)) { QMessageBox::warning (this, tr("Couldn't open audio device"), tr("<b>No audio available</b><p>Could not open an audio device for playback.<p>Automatic audio device detection failed. Audio playback will not be available during this session.</p>"), QMessageBox::Ok); +/* } else { QMessageBox::warning (this, tr("Couldn't open audio device"), @@ -2189,6 +2196,7 @@ .arg(factory->getCallbackTargetDescription(targetName)), QMessageBox::Ok); } +*/ } }
--- a/framework/MainWindowBase.h Mon Jul 13 14:39:41 2015 +0100 +++ b/framework/MainWindowBase.h Tue Aug 04 13:27:42 2015 +0100 @@ -46,7 +46,6 @@ class WaveformLayer; class WaveFileModel; class AudioCallbackPlaySource; -class AudioCallbackPlayTarget; class CommandHistory; class QMenu; class AudioDial; @@ -63,6 +62,10 @@ class QSignalMapper; class QShortcut; +namespace breakfastquay { +class SystemPlaybackTarget; +} + /** * The base class for the SV main window. This includes everything to * do with general document and pane stack management, but nothing @@ -306,7 +309,7 @@ bool m_audioOutput; AudioCallbackPlaySource *m_playSource; - AudioCallbackPlayTarget *m_playTarget; + breakfastquay::SystemPlaybackTarget *m_playTarget; class OSCQueueStarter : public QThread {
--- a/svapp.pro Mon Jul 13 14:39:41 2015 +0100 +++ b/svapp.pro Tue Aug 04 13:27:42 2015 +0100 @@ -23,10 +23,10 @@ } win* { - DEFINES += HAVE_PORTAUDIO_2_0 + DEFINES += HAVE_PORTAUDIO } macx* { - DEFINES += HAVE_COREAUDIO HAVE_PORTAUDIO_2_0 + DEFINES += HAVE_COREAUDIO HAVE_PORTAUDIO } } @@ -35,32 +35,22 @@ TARGET = svapp -DEPENDPATH += . ../svcore ../svgui -INCLUDEPATH += . ../svcore ../svgui +DEPENDPATH += . ../bqaudioio ../svcore ../svgui +INCLUDEPATH += . ../bqaudioio ../svcore ../svgui OBJECTS_DIR = o MOC_DIR = o -HEADERS += audioio/AudioCallbackPlaySource.h \ - audioio/AudioCallbackPlayTarget.h \ - audioio/AudioGenerator.h \ - audioio/AudioJACKTarget.h \ - audioio/AudioPortAudioTarget.h \ - audioio/AudioPulseAudioTarget.h \ - audioio/AudioTargetFactory.h \ - audioio/ClipMixer.h \ - audioio/ContinuousSynth.h \ - audioio/PlaySpeedRangeMapper.h +HEADERS += audio/AudioCallbackPlaySource.h \ + audio/AudioGenerator.h \ + audio/ClipMixer.h \ + audio/ContinuousSynth.h \ + audio/PlaySpeedRangeMapper.h -SOURCES += audioio/AudioCallbackPlaySource.cpp \ - audioio/AudioCallbackPlayTarget.cpp \ - audioio/AudioGenerator.cpp \ - audioio/AudioJACKTarget.cpp \ - audioio/AudioPortAudioTarget.cpp \ - audioio/AudioPulseAudioTarget.cpp \ - audioio/AudioTargetFactory.cpp \ - audioio/ClipMixer.cpp \ - audioio/ContinuousSynth.cpp \ - audioio/PlaySpeedRangeMapper.cpp +SOURCES += audio/AudioCallbackPlaySource.cpp \ + audio/AudioGenerator.cpp \ + audio/ClipMixer.cpp \ + audio/ContinuousSynth.cpp \ + audio/PlaySpeedRangeMapper.cpp HEADERS += framework/Document.h \ framework/MainWindowBase.h \