annotate transform/RealTimePluginTransform.cpp @ 52:527598e2fa10

* Handle generator transforms (plugins whose channel count isn't dependent on
  number of audio inputs, as they have none)
* Be less keen to suspend writing FFT data in spectrogram repaint -- only do it
  if we find we actually need to query the FFT data (i.e. we aren't repainting
  an area that hasn't been generated at all yet)

/* -*- 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 "RealTimePluginTransform.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 <iostream>

RealTimePluginTransform::RealTimePluginTransform(Model *inputModel,
                                                 QString pluginId,
                                                 const ExecutionContext &context,
                                                 QString configurationXml,
                                                 QString units,
                                                 int output) :
    PluginTransform(inputModel, context),
    m_plugin(0),
    m_outputNo(output)
{
    if (!m_context.blockSize) m_context.blockSize = 1024;

    std::cerr << "RealTimePluginTransform::RealTimePluginTransform: plugin " << pluginId.toStdString() << ", output " << output << std::endl;

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

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

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

    m_plugin = factory->instantiatePlugin(pluginId, 0, 0,
                                          m_input->getSampleRate(),
                                          m_context.blockSize,
                                          input->getChannelCount());

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

    if (configurationXml != "") {
        PluginXml(m_plugin).setParametersFromXml(configurationXml);
    }

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

    if (m_outputNo == -1) {

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

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

        m_output = model;

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

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

        m_output = model;
    }
}

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

DenseTimeValueModel *
RealTimePluginTransform::getInput()
{
    DenseTimeValueModel *dtvm =
	dynamic_cast<DenseTimeValueModel *>(getInputModel());
    if (!dtvm) {
	std::cerr << "RealTimePluginTransform::getInput: WARNING: Input model is not conformable to DenseTimeValueModel" << std::endl;
    }
    return dtvm;
}

void
RealTimePluginTransform::run()
{
    DenseTimeValueModel *input = getInput();
    if (!input) return;

    SparseTimeValueModel *stvm = dynamic_cast<SparseTimeValueModel *>(m_output);
    WritableWaveFileModel *wwfm = dynamic_cast<WritableWaveFileModel *>(m_output);
    if (!stvm && !wwfm) return;

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

    size_t sampleRate = input->getSampleRate();
    int channelCount = input->getChannelCount();
    if (!wwfm && m_context.channel != -1) channelCount = 1;

    size_t blockSize = m_plugin->getBufferSize();

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

    size_t startFrame = m_input->getStartFrame();
    size_t   endFrame = m_input->getEndFrame();
    size_t blockFrame = startFrame;

    size_t prevCompletion = 0;

    size_t latency = m_plugin->getLatency();

    int i = 0;

    while (blockFrame < endFrame) {

	size_t completion =
	    (((blockFrame - startFrame) / blockSize) * 99) /
	    (   (endFrame - startFrame) / blockSize);

	size_t got = 0;

	if (channelCount == 1) {
            if (buffers && buffers[0]) {
                got = input->getValues
                    (m_context.channel, blockFrame, blockFrame + blockSize, buffers[0]);
                while (got < blockSize) {
                    buffers[0][got++] = 0.0;
                }
                if (m_context.channel == -1 && channelCount > 1) {
                    // use mean instead of sum, as plugin input
                    for (size_t i = 0; i < got; ++i) {
                        buffers[0][i] /= channelCount;
                    }
                }                
            }
	} else {
	    for (size_t ch = 0; ch < channelCount; ++ch) {
                if (buffers && buffers[ch]) {
                    got = input->getValues
                        (ch, blockFrame, blockFrame + blockSize, buffers[ch]);
                    while (got < blockSize) {
                        buffers[ch][got++] = 0.0;
                    }
                }
	    }
	}

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

        if (stvm) {

            float value = m_plugin->getControlOutputValue(m_outputNo);

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

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

        } else if (wwfm) {

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

            if (buffers) {

                //!!! This will fail if any buffers[c] is null or
                //uninitialised.  The plugin instance should ensure
                //that that can't happen -- but it doesn't

                if (blockFrame >= latency) {
                    wwfm->addSamples(buffers, blockSize);
                } else if (blockFrame + blockSize >= latency) {
                    size_t offset = latency - blockFrame;
                    size_t count = blockSize - offset;
                    float **tmp = new float *[channelCount];
                    for (size_t c = 0; c < channelCount; ++c) {
                        tmp[c] = buffers[c] + offset;
                    }
                    wwfm->addSamples(tmp, count);
                    delete[] tmp;
                }
            }
        }

	if (blockFrame == startFrame || completion > prevCompletion) {
	    if (stvm) stvm->setCompletion(completion);
	    prevCompletion = completion;
	}
        
	blockFrame += blockSize;
    }
    
    if (stvm) stvm->setCompletion(100);
    if (wwfm) wwfm->sync();
}