annotate base/AudioPlaySource.h @ 1520:954d0cf29ca7 import-audio-data

Switch the normalisation option in WritableWaveFileModel from normalising on read to normalising on write, so that the saved file is already normalised and therefore can be read again without having to remember to normalise it

/* -*- 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 SV_AUDIO_PLAY_SOURCE_H
#define SV_AUDIO_PLAY_SOURCE_H

#include "BaseTypes.h"

struct Auditionable {
    virtual ~Auditionable() { }
};

/**
 * Simple interface for audio playback.  This should be all that the
 * ViewManager needs to know about to synchronise with playback by
 * sample frame, but it doesn't provide enough to determine what is
 * actually being played or how.  See the audioio directory for a
 * concrete subclass.
 */

class AudioPlaySource
{
public:
    virtual ~AudioPlaySource() { }

    /**
     * Start playing from the given frame.  If playback is already
     * under way, reseek to the given frame and continue.
     */
    virtual void play(sv_frame_t startFrame) = 0;

    /**
     * Stop playback.
     */
    virtual void stop() = 0;

    /**
     * Return whether playback is currently supposed to be happening.
     */
    virtual bool isPlaying() const = 0;

    /**
     * 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() = 0;

    /**
     * Return the current (or thereabouts) output levels in the range
     * 0.0 -> 1.0, for metering purposes.  The values returned are
     * peak values since the last call to this function was made
     * (i.e. calling this function also resets them).
     */
    virtual bool getOutputLevels(float &left, float &right) = 0;

    /**
     * Return the sample rate of the source material -- any material
     * that wants to play at a different rate will sound wrong.
     */
    virtual sv_samplerate_t getSourceSampleRate() const = 0;

    /**
     * Return the sample rate set by the target audio device (or 0 if
     * the target hasn't told us yet).  If the source and target
     * sample rates differ, resampling will occur.
     *
     * Note that we don't actually do any processing at the device
     * sample rate. All processing happens at the source sample rate,
     * and then a resampler is applied if necessary at the interface
     * between application and driver layer.
     */
    virtual sv_samplerate_t getDeviceSampleRate() const = 0;

    /**
     * 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.
     */
    virtual int getTargetBlockSize() const = 0;

    /**
     * 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.
     */
    virtual int getTargetChannelCount() const = 0;

    /**
     * Set a plugin or other subclass of Auditionable as an
     * auditioning effect.
     */
    virtual void setAuditioningEffect(Auditionable *) = 0;

};

#endif