view transform/RealTimeEffectModelTransformer.cpp @ 1196:c7b9c902642f spectrogram-minor-refactor

Fix threshold in spectrogram -- it wasn't working in the last release.
There is a new protocol for this. Formerly the threshold parameter had a
range from -50dB to 0 with the default at -50, and -50 treated internally
as "no threshold". However, there was a hardcoded, hidden internal threshold
for spectrogram colour mapping at -80dB with anything below this being rounded
to zero. Now the threshold parameter has range -81 to -1 with the default
at -80, -81 is treated internally as "no threshold", and there is no hidden
internal threshold. So the default behaviour is the same as before, an
effective -80dB threshold, but it is now possible to change this in both
directions. Sessions reloaded from prior versions may look slightly different
because, if the session says there should be no threshold, there will now
actually be no threshold instead of having the hidden internal one.

Still need to do something in the UI to make it apparent that the -81dB
setting removes the threshold entirely. This is at least no worse than the
previous, also obscured, magic -50dB setting.

/* -*- 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 "RealTimeEffectModelTransformer.h"

#include "plugin/RealTimePluginFactory.h"
#include "plugin/RealTimePluginInstance.h"
#include "plugin/PluginXml.h"

#include "data/model/Model.h"
#include "data/model/SparseTimeValueModel.h"
#include "data/model/DenseTimeValueModel.h"
#include "data/model/WritableWaveFileModel.h"
#include "data/model/WaveFileModel.h"

#include "TransformFactory.h"

#include <iostream>

RealTimeEffectModelTransformer::RealTimeEffectModelTransformer(Input in,
                                                               const Transform &t) :
    ModelTransformer(in, t),
    m_plugin(0)
{
    Transform transform(t);
    if (!transform.getBlockSize()) {
        transform.setBlockSize(1024);
        m_transforms[0] = transform;
    }

    m_units = TransformFactory::getInstance()->getTransformUnits
        (transform.getIdentifier());
    m_outputNo =
        (transform.getOutput() == "A") ? -1 : transform.getOutput().toInt();

    QString pluginId = transform.getPluginIdentifier();

//    SVDEBUG << "RealTimeEffectModelTransformer::RealTimeEffectModelTransformer: plugin " << pluginId << ", output " << output << endl;

    RealTimePluginFactory *factory =
	RealTimePluginFactory::instanceFor(pluginId);

    if (!factory) {
	cerr << "RealTimeEffectModelTransformer: No factory available for plugin id \""
		  << pluginId << "\"" << endl;
	return;
    }

    DenseTimeValueModel *input = getConformingInput();
    if (!input) return;

    m_plugin = factory->instantiatePlugin(pluginId, 0, 0,
                                          input->getSampleRate(),
                                          transform.getBlockSize(),
                                          input->getChannelCount());

    if (!m_plugin) {
	cerr << "RealTimeEffectModelTransformer: Failed to instantiate plugin \""
             << pluginId << "\"" << endl;
	return;
    }

    TransformFactory::getInstance()->setPluginParameters(transform, m_plugin);

    if (m_outputNo >= 0 &&
        m_outputNo >= int(m_plugin->getControlOutputCount())) {
        cerr << "RealTimeEffectModelTransformer: Plugin has fewer than desired " << m_outputNo << " control outputs" << endl;
        return;
    }

    if (m_outputNo == -1) {

        int outputChannels = (int)m_plugin->getAudioOutputCount();
        if (outputChannels > input->getChannelCount()) {
            outputChannels = input->getChannelCount();
        }

        WritableWaveFileModel *model = new WritableWaveFileModel
            (input->getSampleRate(), outputChannels);

        m_outputs.push_back(model);

    } else {
	
        SparseTimeValueModel *model = new SparseTimeValueModel
            (input->getSampleRate(), transform.getBlockSize(), 0.0, 0.0, false);

        if (m_units != "") model->setScaleUnits(m_units);

        m_outputs.push_back(model);
    }
}

RealTimeEffectModelTransformer::~RealTimeEffectModelTransformer()
{
    delete m_plugin;
}

DenseTimeValueModel *
RealTimeEffectModelTransformer::getConformingInput()
{
    DenseTimeValueModel *dtvm =
	dynamic_cast<DenseTimeValueModel *>(getInputModel());
    if (!dtvm) {
	SVDEBUG << "RealTimeEffectModelTransformer::getConformingInput: WARNING: Input model is not conformable to DenseTimeValueModel" << endl;
    }
    return dtvm;
}

void
RealTimeEffectModelTransformer::run()
{
    DenseTimeValueModel *input = getConformingInput();
    if (!input) return;

    while (!input->isReady() && !m_abandoned) {
        SVDEBUG << "RealTimeEffectModelTransformer::run: Waiting for input model to be ready..." << endl;
        usleep(500000);
    }
    if (m_abandoned) return;

    SparseTimeValueModel *stvm = dynamic_cast<SparseTimeValueModel *>(m_outputs[0]);
    WritableWaveFileModel *wwfm = dynamic_cast<WritableWaveFileModel *>(m_outputs[0]);
    if (!stvm && !wwfm) return;

    if (stvm && (m_outputNo >= int(m_plugin->getControlOutputCount()))) return;

    sv_samplerate_t sampleRate = input->getSampleRate();
    int channelCount = input->getChannelCount();
    if (!wwfm && m_input.getChannel() != -1) channelCount = 1;

    sv_frame_t blockSize = m_plugin->getBufferSize();

    float **inbufs = m_plugin->getAudioInputBuffers();

    sv_frame_t startFrame = m_input.getModel()->getStartFrame();
    sv_frame_t endFrame = m_input.getModel()->getEndFrame();

    Transform transform = m_transforms[0];
    
    RealTime contextStartRT = transform.getStartTime();
    RealTime contextDurationRT = transform.getDuration();

    sv_frame_t contextStart =
        RealTime::realTime2Frame(contextStartRT, sampleRate);

    sv_frame_t contextDuration =
        RealTime::realTime2Frame(contextDurationRT, sampleRate);

    if (contextStart == 0 || contextStart < startFrame) {
        contextStart = startFrame;
    }

    if (contextDuration == 0) {
        contextDuration = endFrame - contextStart;
    }
    if (contextStart + contextDuration > endFrame) {
        contextDuration = endFrame - contextStart;
    }

    if (wwfm) {
        wwfm->setStartFrame(contextStart);
    }

    sv_frame_t blockFrame = contextStart;

    int prevCompletion = 0;

    sv_frame_t latency = m_plugin->getLatency();

    while (blockFrame < contextStart + contextDuration + latency &&
           !m_abandoned) {

	int completion = int
	    ((((blockFrame - contextStart) / blockSize) * 99) /
             (1 + ((contextDuration) / blockSize)));

	sv_frame_t got = 0;

	if (channelCount == 1) {
            if (inbufs && inbufs[0]) {
                auto data = input->getData
                    (m_input.getChannel(), blockFrame, blockSize);
                got = data.size();
                for (sv_frame_t i = 0; i < got; ++i) {
                    inbufs[0][i] = data[i];
                }
                while (got < blockSize) {
                    inbufs[0][got++] = 0.f;
                }          
                for (int ch = 1; ch < (int)m_plugin->getAudioInputCount(); ++ch) {
                    for (sv_frame_t i = 0; i < blockSize; ++i) {
                        inbufs[ch][i] = inbufs[0][i];
                    }
                }
            }
	} else {
            if (inbufs && inbufs[0]) {
                auto data = input->getMultiChannelData
                    (0, channelCount - 1, blockFrame, blockSize);
                if (!data.empty()) got = data[0].size();
                for (int ch = 0; ch < channelCount; ++ch) {
                    for (sv_frame_t i = 0; i < got; ++i) {
                        inbufs[ch][i] = data[ch][i];
                    }
                }
                while (got < blockSize) {
                    for (int ch = 0; ch < channelCount; ++ch) {
                        inbufs[ch][got] = 0.0;
                    }
                    ++got;
                }
                for (int ch = channelCount; ch < (int)m_plugin->getAudioInputCount(); ++ch) {
                    for (sv_frame_t i = 0; i < blockSize; ++i) {
                        inbufs[ch][i] = inbufs[ch % channelCount][i];
                    }
                }
            }
	}

/*
        cerr << "Input for plugin: " << m_plugin->getAudioInputCount() << " channels "<< endl;

        for (int ch = 0; ch < m_plugin->getAudioInputCount(); ++ch) {
            cerr << "Input channel " << ch << endl;
            for (int i = 0; i < 100; ++i) {
                cerr << inbufs[ch][i] << " ";
                if (isnan(inbufs[ch][i])) {
                    cerr << "\n\nWARNING: NaN in audio input" << endl;
                }
            }
        }
*/

        m_plugin->run(RealTime::frame2RealTime(blockFrame, sampleRate));

        if (stvm) {

            float value = m_plugin->getControlOutputValue(m_outputNo);

            sv_frame_t pointFrame = blockFrame;
            if (pointFrame > latency) pointFrame -= latency;
            else pointFrame = 0;

            stvm->addPoint(SparseTimeValueModel::Point
                           (pointFrame, value, ""));

        } else if (wwfm) {

            float **outbufs = m_plugin->getAudioOutputBuffers();

            if (outbufs) {

                if (blockFrame >= latency) {
                    sv_frame_t writeSize = std::min
                        (blockSize,
                         contextStart + contextDuration + latency - blockFrame);
                    wwfm->addSamples(outbufs, writeSize);
                } else if (blockFrame + blockSize >= latency) {
                    sv_frame_t offset = latency - blockFrame;
                    sv_frame_t count = blockSize - offset;
                    float **tmp = new float *[channelCount];
                    for (int c = 0; c < channelCount; ++c) {
                        tmp[c] = outbufs[c] + offset;
                    }
                    wwfm->addSamples(tmp, count);
                    delete[] tmp;
                }
            }
        }

	if (blockFrame == contextStart || completion > prevCompletion) {
            // This setCompletion is probably misusing the completion
            // terminology, just as it was for WritableWaveFileModel
	    if (stvm) stvm->setCompletion(completion);
	    if (wwfm) wwfm->setWriteProportion(completion);
	    prevCompletion = completion;
	}
        
	blockFrame += blockSize;
    }

    if (m_abandoned) return;
    
    if (stvm) stvm->setCompletion(100);
    if (wwfm) wwfm->writeComplete();
}