mirror of
https://github.com/libretro/scummvm.git
synced 2024-12-14 05:38:56 +00:00
67b311713d
svn-id: r16330
260 lines
5.6 KiB
C++
260 lines
5.6 KiB
C++
/* ScummVM - Scumm Interpreter
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* Copyright (C) 2004 The ScummVM project
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*
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* The ReInherit Engine is (C)2000-2003 by Daniel Balsom.
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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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#include "saga/saga.h"
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#include "saga/sound.h"
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#include "sound/audiostream.h"
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#include "sound/mixer.h"
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namespace Saga {
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#define BUFFER_SIZE 4096
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// Routines to convert 12 bit linear samples to the
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// Dialogic or Oki ADPCM coding format.
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class VOXInputStream : public AudioStream {
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private:
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const byte *_buf;
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uint32 _pos;
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uint32 _inputLen;
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bool _evenPos;
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struct adpcmStatus {
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int16 last;
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int16 stepIndex;
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} _status;
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int16 stepAdjust(byte);
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int16 adpcmDecode(byte);
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public:
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VOXInputStream(const byte *input, int inputLen);
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~VOXInputStream() {};
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int readBuffer(int16 *buffer, const int numSamples);
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bool endOfData() const { return _pos >= _inputLen; }
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bool isStereo() const { return false; }
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int getRate() const { return 22050; }
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};
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VOXInputStream::VOXInputStream(const byte *input, int inputLen)
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: _buf(input), _pos(0), _inputLen(inputLen), _evenPos(true) {
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_status.last = 0;
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_status.stepIndex = 0;
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}
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int VOXInputStream::readBuffer(int16 *buffer, const int numSamples) {
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int samples = 0;
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while (samples < numSamples && !endOfData()) {
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const int len = MIN(numSamples - samples, (int) (_inputLen - _pos));
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// * 16 effectively converts 12-bit input to 16-bit output
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for (int i = 0; i < len; i++) {
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if (_evenPos)
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buffer[i] = adpcmDecode((_buf[_pos] >> 4) & 0x0f) * 16;
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else {
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buffer[i] = adpcmDecode(_buf[_pos] & 0x0f) * 16;
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_pos++;
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}
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_evenPos = !_evenPos;
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}
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samples += len;
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}
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return samples;
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}
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// adjust the step for use on the next sample.
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int16 VOXInputStream::stepAdjust(byte code) {
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static int16 adjusts[] = {-1, -1, -1, -1, 2, 4, 6, 8};
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return adjusts[code & 0x07];
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}
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static int16 stepSize[49] = { 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41,
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45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173,
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190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494, 544, 598, 658,
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724, 796, 876, 963, 1060, 1166, 1282, 1408, 1552 };
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// Decode Linear to ADPCM
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int16 VOXInputStream::adpcmDecode(byte code) {
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int16 diff, E, SS, samp;
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SS = stepSize[_status.stepIndex];
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E = SS/8;
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if (code & 0x01)
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E += SS/4;
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if (code & 0x02)
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E += SS/2;
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if (code & 0x04)
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E += SS;
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diff = (code & 0x08) ? -E : E;
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samp = _status.last + diff;
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// Clip the values to +/- 2^11 (supposed to be 12 bits)
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if(samp > 2048)
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samp = 2048;
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if(samp < -2048)
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samp = -2048;
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_status.last = samp;
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_status.stepIndex += stepAdjust(code);
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if(_status.stepIndex < 0)
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_status.stepIndex = 0;
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if(_status.stepIndex > 48)
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_status.stepIndex = 48;
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return samp;
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}
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AudioStream *makeVOXStream(const byte *input, int size) {
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AudioStream *audioStream = new VOXInputStream(input, size);
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return audioStream;
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}
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Sound::Sound(SagaEngine *vm, SoundMixer *mixer, int enabled) :
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_vm(vm), _mixer(mixer), _enabled(enabled) {
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_soundInitialized = 1;
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return;
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}
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Sound::~Sound() {
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if (!_soundInitialized) {
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return;
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}
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_soundInitialized = 0;
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}
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int Sound::playSoundBuffer(PlayingSoundHandle *handle, SOUNDBUFFER *buf, int volume, bool loop) {
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byte flags;
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if (!_soundInitialized) {
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return FAILURE;
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}
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flags = SoundMixer::FLAG_AUTOFREE;
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if (loop)
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flags |= SoundMixer::FLAG_LOOP;
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if (buf->s_samplebits == 16) {
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flags |= SoundMixer::FLAG_16BITS;
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if (!(_vm->_features & GF_BIG_ENDIAN_DATA))
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flags |= SoundMixer::FLAG_LITTLE_ENDIAN;
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}
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if (buf->s_stereo)
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flags |= SoundMixer::FLAG_STEREO;
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if (!buf->s_signed)
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flags |= SoundMixer::FLAG_UNSIGNED;
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_mixer->playRaw(handle, buf->s_buf, buf->s_buf_len, buf->s_freq, flags, -1, volume);
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return SUCCESS;
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}
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int Sound::playSound(SOUNDBUFFER *buf, int volume, bool loop) {
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return playSoundBuffer(&_effectHandle, buf, 2 * volume, loop);
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}
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int Sound::pauseSound() {
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if (!_soundInitialized) {
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return FAILURE;
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}
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_mixer->pauseHandle(_effectHandle, true);
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return SUCCESS;
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}
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int Sound::resumeSound() {
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if (!_soundInitialized) {
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return FAILURE;
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}
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_mixer->pauseHandle(_effectHandle, false);
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return SUCCESS;
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}
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int Sound::stopSound() {
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if (!_soundInitialized) {
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return FAILURE;
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}
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_mixer->stopHandle(_effectHandle);
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return SUCCESS;
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}
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int Sound::playVoice(SOUNDBUFFER *buf) {
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return playSoundBuffer(&_voiceHandle, buf, 255, false);
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}
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int Sound::playVoxVoice(SOUNDBUFFER *buf) {
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AudioStream *audioStream;
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audioStream = makeVOXStream(buf->s_buf, buf->s_buf_len);
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_mixer->playInputStream(SoundMixer::kSFXAudioDataType, &_voiceHandle, audioStream);
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return SUCCESS;
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}
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int Sound::pauseVoice() {
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if (!_soundInitialized) {
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return FAILURE;
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}
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_mixer->pauseHandle(_voiceHandle, true);
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return SUCCESS;
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}
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int Sound::resumeVoice() {
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if (!_soundInitialized) {
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return FAILURE;
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}
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_mixer->pauseHandle(_voiceHandle, false);
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return SUCCESS;
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}
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int Sound::stopVoice() {
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if (!_soundInitialized) {
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return FAILURE;
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}
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_mixer->stopHandle(_voiceHandle);
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return SUCCESS;
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}
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} // End of namespace Saga
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