Revision 2dd1dd53000918abba04156845fce2f50182657e authored by Sven Hesse on 30 July 2007, 19:54:51 UTC, committed by Sven Hesse on 30 July 2007, 19:54:51 UTC
svn-id: r28338
1 parent 3cddad3
audiostream.h
/* ScummVM - Graphic Adventure Engine
*
* ScummVM is the legal property of its developers, whose names
* are too numerous to list here. Please refer to the COPYRIGHT
* file distributed with this source distribution.
*
* 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.
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*
* $URL$
* $Id$
*
*/
#ifndef SOUND_AUDIOSTREAM_H
#define SOUND_AUDIOSTREAM_H
#include "common/stdafx.h"
#include "common/util.h"
#include "common/scummsys.h"
namespace Audio {
/**
* Generic audio input stream. Subclasses of this are used to feed arbitrary
* sampled audio data into ScummVM's audio mixer.
*/
class AudioStream {
public:
virtual ~AudioStream() {}
/**
* Fill the given buffer with up to numSamples samples.
* Returns the actual number of samples read, or -1 if
* a critical error occured (note: you *must* check if
* this value is less than what you requested, this can
* happen when the stream is fully used up).
*
* Data has to be in native endianess, 16 bit per sample, signed.
* For stereo stream, buffer will be filled with interleaved
* left and right channel samples, starting with a left sample.
* Furthermore, the samples in the left and right are summed up.
* So if you request 4 samples from a stereo stream, you will get
* a total of two left channel and two right channel samples.
*/
virtual int readBuffer(int16 *buffer, const int numSamples) = 0;
/** Is this a stereo stream? */
virtual bool isStereo() const = 0;
/**
* End of data reached? If this returns true, it means that at this
* time there is no data available in the stream. However there may be
* more data in the future.
* This is used by e.g. a rate converter to decide whether to keep on
* converting data or stop.
*/
virtual bool endOfData() const = 0;
/**
* End of stream reached? If this returns true, it means that all data
* in this stream is used up and no additional data will appear in it
* in the future.
* This is used by the mixer to decide whether a given stream shall be
* removed from the list of active streams (and thus be destroyed).
* By default this maps to endOfData()
*/
virtual bool endOfStream() const { return endOfData(); }
/** Sample rate of the stream. */
virtual int getRate() const = 0;
/**
* Tries to load a file by trying all available formats.
* In case of an error, the file handle will be closed, but deleting
* it is still the responsibilty of the caller.
* @param basename a filename without an extension
* @param startTime the (optional) time offset in milliseconds from which to start playback
* @param duration the (optional) time in milliseconds specifying how long to play
* @param numLoops how often the data shall be looped (0 = infinite)
* @return an Audiostream ready to use in case of success;
* NULL in case of an error (e.g. invalid/nonexisting file)
*/
static AudioStream* openStreamFile(const Common::String &basename, uint32 startTime = 0, uint32 duration = 0, uint numLoops = 1);
};
/**
* Factory function for a raw linear AudioStream, which will simply treat all data
* in the buffer described by ptr and len as raw sample data in the specified
* format. It will then simply pass this data directly to the mixer, after converting
* it to the sample format used by the mixer (i.e. 16 bit signed native endian).
* Optionally supports (infinite) looping of a portion of the data.
*/
AudioStream *makeLinearInputStream(const byte *ptr, uint32 len, int rate, byte flags, uint loopStart, uint loopEnd);
/**
* An audio stream to which additional data can be appended on-the-fly.
* Used by SMUSH, iMuseDigital, and the Kyrandia 3 VQA player.
*/
class AppendableAudioStream : public Audio::AudioStream {
public:
/**
* Queue another audio data buffer for playback. The stream
* will playback all queued buffers, in the order they were
* queued. After all data contained in them has been played,
* the buffer will be delete[]'d (so make sure to allocate them
* with new[], not with malloc).
*/
virtual void queueBuffer(byte *data, uint32 size) = 0;
/**
* Mark the stream as finished, that is, signal that no further data
* will be appended to it. Only after this has been done can the
* AppendableAudioStream ever 'end' (
*/
virtual void finish() = 0;
};
/**
* Factory function for an AppendableAudioStream. The rate and flags
* parameters are analog to those used in makeLinearInputStream.
*/
AppendableAudioStream *makeAppendableAudioStream(int rate, byte flags);
} // End of namespace Audio
#endif
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