Mercurial > hg > svcore
view base/AudioPlaySource.h @ 537:3cc4b7cd2aa5
* Merge from one-fftdataserver-per-fftmodel branch. This bit of
reworking (which is not described very accurately by the title of
the branch) turns the MatrixFile object into something that either
reads or writes, but not both, and separates the FFT file cache
reader and writer implementations separately. This allows the
FFT data server to have a single thread owning writers and one reader
per "customer" thread, and for all locking to be vastly simplified
and concentrated in the data server alone (because none of the
classes it makes use of is used in more than one thread at a time).
The result is faster and more trustworthy code.
author | Chris Cannam |
---|---|
date | Tue, 27 Jan 2009 13:25:10 +0000 |
parents | a1b6d2e33cab |
children | cb5eb032c28b 6a94bb528e9d |
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/* -*- 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_PLAY_SOURCE_H_ #define _AUDIO_PLAY_SOURCE_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(size_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 size_t getCurrentPlayingFrame() = 0; /** * Return the current (or thereabouts) output levels in the range * 0.0 -> 1.0, for metering purposes. */ 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 size_t getSourceSampleRate() const = 0; /** * Return the sample rate set by the target audio device (or the * source sample rate if the target hasn't set one). If the * source and target sample rates differ, resampling will occur. */ virtual size_t getTargetSampleRate() 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 size_t 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 size_t getTargetChannelCount() const = 0; /** * Set a plugin or other subclass of Auditionable as an * auditioning effect. */ virtual void setAuditioningEffect(Auditionable *) = 0; }; #endif