annotate audioio/AudioCallbackPlaySource.h @ 25:e74f508db18c

* Add setRatio method to the time stretcher, and make it possible to change
  the ratio without having to construct and replace the time stretcher.  This
  means we can do it seamlessly.  Add a lot more ratios to the
  time stretch control in the main window

/* -*- 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/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;

/**
 * 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);

    void setSlowdownFactor(float factor, bool sharpen);

signals:
    void modelReplaced();

    void playStatusChanged(bool isPlaying);

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

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

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;

    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);

    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;
};

#endif