mirror of
https://github.com/libretro/scummvm.git
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f67b126f1d
svn-id: r9588
124 lines
3.4 KiB
C++
124 lines
3.4 KiB
C++
/* ScummVM - Scumm Interpreter
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* Copyright (C) 2001-2003 The ScummVM project
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*
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* $Header$
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*
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*/
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#ifndef AUDIOSTREAM_H
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#define AUDIOSTREAM_H
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#include "stdafx.h"
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#include "common/scummsys.h"
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#include "common/util.h"
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#ifdef USE_MAD
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#include <mad.h>
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#endif
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#ifdef USE_VORBIS
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#include <vorbis/vorbisfile.h>
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#endif
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class File;
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// TODO:
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// * maybe make readIntern return 16.16 or 24.8 fixed point values
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// since MAD (and maybe OggVorbis?) gives us those -> higher quality.
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// The rate converters should be able to deal with those just fine, too.
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// * possibly add MADInputStream and VorbisInputStream
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/**
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* Generic input stream for the resampling code.
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*/
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class AudioInputStream {
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public:
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virtual ~AudioInputStream() {}
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/**
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* Fill the given buffer with up to numSamples samples.
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* Returns the actual number of samples read, or -1 if
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* a critical error occured (note: you *must* check if
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* this value is less than what you requested, this can
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* happend when the stream is fully used up).
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* For stereo stream, buffer will be filled with interleaved
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* left and right channel samples.
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*
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* For maximum efficency, subclasses should always override
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* the default implementation!
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*/
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virtual int readBuffer(int16 *buffer, const int numSamples) {
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int samples;
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for (samples = 0; samples < numSamples && !eos(); samples++) {
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*buffer++ = read();
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}
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return samples;
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}
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/** Read a singel (16 bit signed) sample from the stream. */
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virtual int16 read() = 0;
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/** Is this a stereo stream? */
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virtual bool isStereo() const = 0;
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/* End of stream reached? */
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virtual bool eos() const = 0;
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virtual int getRate() const { return -1; }
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};
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class WrappedAudioInputStream : public AudioInputStream {
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public:
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virtual void append(const byte *data, uint32 len) = 0;
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};
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class ZeroInputStream : public AudioInputStream {
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protected:
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int _len;
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public:
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ZeroInputStream(uint len) : _len(len) { }
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int readBuffer(int16 *buffer, int numSamples) {
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int samples = MIN(_len, numSamples);
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memset(buffer, 0, samples * 2);
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_len -= samples;
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return samples;
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}
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int16 read() { assert(_len > 0); _len--; return 0; }
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int size() const { return _len; }
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bool isStereo() const { return false; }
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bool eos() const { return _len <= 0; }
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};
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class MusicStream : public AudioInputStream {
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public:
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virtual int getRate() const = 0;
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};
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AudioInputStream *makeLinearInputStream(byte _flags, const byte *ptr, uint32 len, uint loopOffset, uint loopLen);
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WrappedAudioInputStream *makeWrappedInputStream(byte _flags, uint32 len);
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#ifdef USE_MAD
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MusicStream *makeMP3Stream(File *file, mad_timer_t duration, uint size = 0);
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#endif
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#ifdef USE_VORBIS
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MusicStream *makeVorbisStream(OggVorbis_File *file, int duration);
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#endif
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#endif
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