mirror of
https://github.com/libretro/scummvm.git
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e22230e223
Add some more HE differences svn-id: r15188
197 lines
6.1 KiB
C++
197 lines
6.1 KiB
C++
/* ScummVM - Scumm Interpreter
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* Copyright (C) 2001 Ludvig Strigeus
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* Copyright (C) 2001-2004 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 SOUND_MIXER_H
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#define SOUND_MIXER_H
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#include "stdafx.h"
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#include "common/scummsys.h"
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#include "common/system.h"
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class AudioStream;
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class Channel;
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class File;
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class PlayingSoundHandle {
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friend class Channel;
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friend class SoundMixer;
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int val;
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int getIndex() const { return val - 1; }
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void setIndex(int i) { val = i + 1; }
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void resetIndex() { val = 0; }
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public:
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PlayingSoundHandle() { resetIndex(); }
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bool isActive() const { return val > 0; }
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};
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class SoundMixer {
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public:
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typedef void PremixProc (void *param, int16 *data, uint len);
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enum {
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NUM_CHANNELS = 16
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};
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enum {
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FLAG_UNSIGNED = 1 << 0, /** unsigned samples (default: signed) */
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FLAG_16BITS = 1 << 1, /** sound is 16 bits wide (default: 8bit) */
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FLAG_LITTLE_ENDIAN = 1 << 2, /** sample is little endian (default: big endian) */
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FLAG_STEREO = 1 << 3, /** sound is in stereo (default: mono) */
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FLAG_REVERSE_STEREO = 1 << 4, /** reverse the left and right stereo channel */
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FLAG_AUTOFREE = 1 << 5, /** sound buffer is freed automagically at the end of playing */
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FLAG_LOOP = 1 << 6 /** loop the audio */
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};
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private:
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OSystem *_syst;
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OSystem::MutexRef _mutex;
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void *_premixParam;
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PremixProc *_premixProc;
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uint _outputRate;
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int _globalVolume;
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int _musicVolume;
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bool _paused;
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Channel *_channels[NUM_CHANNELS];
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bool _mixerReady;
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public:
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SoundMixer();
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~SoundMixer();
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/**
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* Is the mixer ready and setup? This may not be the case on systems which
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* don't support digital sound output. In that case, the mixer proc may
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* never be called. That in turn can cause breakage in games which use the
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* premix callback for syncing. In particular, the Adlib MIDI emulation...
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*/
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bool isReady() const { return _mixerReady; };
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/**
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* Set the premix procedure. This is mainly used for the adlib music, but
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* is not limited to it. The premix proc is invoked by the mixer whenever
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* it needs to generate any data, before any other mixing takes place. The
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* premixer than has a chanve to fill the mix buffer with data (usually
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* music samples). It should generate the specified number of 16bit stereo
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* samples (i.e. len * 4 bytes). The endianness of these samples shall be
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* the native endianness.
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*/
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void setupPremix(PremixProc *proc, void *param);
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// start playing a raw sound
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void playRaw(PlayingSoundHandle *handle, void *sound, uint32 size, uint rate, byte flags,
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int id = -1, byte volume = 255, int8 balance = 0, uint32 loopStart = 0, uint32 loopEnd = 0);
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#ifdef USE_MAD
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void playMP3(PlayingSoundHandle *handle, File *file, uint32 size, byte volume = 255, int8 balance = 0, int id = -1);
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#endif
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#ifdef USE_VORBIS
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void playVorbis(PlayingSoundHandle *handle, File *file, uint32 size, byte volume = 255, int8 balance = 0, int id = -1);
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#endif
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#ifdef USE_FLAC
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void playFlac(PlayingSoundHandle *handle, File *file, uint32 size, byte volume = 255, int8 balance = 0, int id = -1);
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#endif
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void playInputStream(PlayingSoundHandle *handle, AudioStream *input, bool isMusic, byte volume = 255, int8 balance = 0, int id = -1, bool autofreeStream = true);
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/** Start a new stream. */
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void newStream(PlayingSoundHandle *handle, uint rate, byte flags, uint32 buffer_size, byte volume = 255, int8 balance = 0);
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/** Append to an existing stream. */
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void appendStream(PlayingSoundHandle handle, void *sound, uint32 size);
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/**
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* Mark a stream as finished.
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* Where stopHandle() would stop the sound immediately, when using this
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* method, the stream will first finish playing all its data before it
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* finally stops.
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*/
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void endStream(PlayingSoundHandle handle);
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/** stop all currently playing sounds */
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void stopAll();
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/** stop playing the sound with given ID */
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void stopID(int id);
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/** stop playing the channel for the given handle */
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void stopHandle(PlayingSoundHandle handle);
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/** pause/unpause all channels */
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void pauseAll(bool paused);
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/** check if sound ID is active */
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bool isSoundIDActive(int id);
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/** check if mixer is paused */
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bool isPaused();
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/** pause/unpause the sound with the given ID */
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void pauseID(int id, bool paused);
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/** pause/unpause the channel for the given handle */
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void pauseHandle(PlayingSoundHandle handle, bool paused);
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/** set the channel volume for the given handle (0 - 255) */
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void setChannelVolume(PlayingSoundHandle handle, byte volume);
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/** set the channel balance for the given handle (-127 ... 0 ... 127) (left ... center ... right)*/
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void setChannelBalance(PlayingSoundHandle handle, int8 balance);
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/** get approximation of for how long the channel has been playing */
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uint32 getChannelElapsedTime(PlayingSoundHandle handle);
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/** Check whether any SFX channel is active.*/
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bool hasActiveSFXChannel();
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/** set the global volume, 0-256 */
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void setVolume(int volume);
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/** query the global volume, 0-256 */
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int getVolume() const { return _globalVolume; }
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/** set the music volume, 0-256 */
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void setMusicVolume(int volume);
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/** query the music volume, 0-256 */
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int getMusicVolume() const { return _musicVolume; }
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/** query the output rate in kHz */
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uint getOutputRate() const { return _outputRate; }
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private:
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void insertChannel(PlayingSoundHandle *handle, Channel *chan);
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/** main mixer method */
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void mix(int16 * buf, uint len);
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static void mixCallback(void *s, byte *samples, int len);
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};
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#endif
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