view audioio/AudioCallbackPlaySource.h @ 61:963e3187d920

* Fix slowness in serving FFT values to feature extraction plugin transform
  (failure to call resume() on FFT model)
* Fix failure to update completion from time/value model

/* -*- 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_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 <samplerate.h>

#include <set>
#include <map>

class Model;
class ViewManager;
class AudioGenerator;
class PlayParameters;
class PhaseVocoderTimeStretcher;
class RealTimePluginInstance;

/**
 * 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 virtual QObject,
				public AudioPlaySource
{
    Q_OBJECT

public:
    AudioCallbackPlaySource(ViewManager *);
    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(size_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 size_t getCurrentPlayingFrame();

    /**
     * Set the block size of the target audio device.  This should
     * be called by the target class.
     */
    void setTargetBlockSize(size_t);

    /**
     * Get the block size of the target audio device.
     */
    size_t 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(size_t);

    /**
     * Get the playback latency of the target audio device.
     */
    size_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(size_t);

    /**
     * Return the sample rate set by the target audio device (or the
     * source sample rate if the target hasn't set one).
     */
    virtual size_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.
     */
    size_t 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.
     */
    size_t 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.
     */
    size_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.
     */
    size_t getSourceSamples(size_t count, float **buffer);

    /**
     * Set the time stretcher factor (i.e. playback speed).  Also
     * specify whether the time stretcher will be variable rate
     * (sharpening transients), and whether time stretching will be
     * carried out on data mixed down to mono for speed.
     */
    void setTimeStretch(float factor, bool sharpen, bool mono);

    /**
     * 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
     * setAuditioningPlugin (depending on real-time constraints).
     *
     * Pass a null pointer to remove the current auditioning plugin,
     * if any.
     */
    void setAuditioningPlugin(RealTimePluginInstance *plugin);

signals:
    void modelReplaced();

    void playStatusChanged(bool isPlaying);

    void sampleRateMismatch(size_t requested, size_t available, bool willResample);

    void audioOverloadPluginDisabled();

public slots:
    void audioProcessingOverload();

protected slots:
    void selectionChanged();
    void playLoopModeChanged();
    void playSelectionModeChanged();
    void playParametersChanged(PlayParameters *);
    void preferenceChanged(PropertyContainer::PropertyName);

protected:
    ViewManager                     *m_viewManager;
    AudioGenerator                  *m_audioGenerator;

    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;
    size_t                            m_readBufferFill;
    size_t                            m_writeBufferFill;
    Scavenger<RingBufferVector>       m_bufferScavenger;
    size_t                            m_sourceChannelCount;
    size_t                            m_blockSize;
    size_t                            m_sourceSampleRate;
    size_t                            m_targetSampleRate;
    size_t                            m_playLatency;
    bool                              m_playing;
    bool                              m_exiting;
    size_t                            m_lastModelEndFrame;
    static const size_t               m_ringBufferSize;
    float                             m_outputLeft;
    float                             m_outputRight;
    RealTimePluginInstance           *m_auditioningPlugin;
    bool                              m_auditioningPluginBypassed;
    Scavenger<RealTimePluginInstance> m_pluginScavenger;

    RingBuffer<float> *getWriteRingBuffer(size_t c) {
	if (m_writeBuffers && c < m_writeBuffers->size()) {
	    return (*m_writeBuffers)[c];
	} else {
	    return 0;
	}
    }

    RingBuffer<float> *getReadRingBuffer(size_t c) {
	RingBufferVector *rb = m_readBuffers;
	if (rb && c < rb->size()) {
	    return (*rb)[c];
	} else {
	    return 0;
	}
    }

    void clearRingBuffers(bool haveLock = false, size_t count = 0);
    void unifyRingBuffers();

    PhaseVocoderTimeStretcher *m_timeStretcher;
    Scavenger<PhaseVocoderTimeStretcher> m_timeStretcherScavenger;

    // 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).
    size_t mixModels(size_t &frame, size_t count, float **buffers);

    // Called from getSourceSamples.
    void applyAuditioningEffect(size_t count, float **buffers);

    class AudioCallbackPlaySourceFillThread : public Thread
    {
    public:
	AudioCallbackPlaySourceFillThread(AudioCallbackPlaySource &source) :
            Thread(Thread::NonRTThread),
	    m_source(source) { }

	virtual void run();

    protected:
	AudioCallbackPlaySource &m_source;
    };

    QMutex m_mutex;
    QWaitCondition m_condition;
    AudioCallbackPlaySourceFillThread *m_fillThread;
    SRC_STATE *m_converter;
    SRC_STATE *m_crapConverter; // for use when playing very fast
    int m_resampleQuality;
    void initialiseConverter();
};

#endif