mirror of
https://github.com/libretro/scummvm.git
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672 lines
20 KiB
C++
672 lines
20 KiB
C++
/* ScummVM - Graphic Adventure Engine
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*
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* ScummVM is the legal property of its developers, whose names
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* are too numerous to list here. Please refer to the COPYRIGHT
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* file distributed with this source distribution.
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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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
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*
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*/
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#include "common/endian.h"
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#include "common/util.h"
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#include "common/memstream.h"
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#include "common/textconsole.h"
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#include "sword1/sound.h"
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#include "sword1/resman.h"
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#include "sword1/logic.h"
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#include "sword1/sword1.h"
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#include "audio/audiostream.h"
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#include "audio/decoders/flac.h"
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#include "audio/decoders/mp3.h"
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#include "audio/decoders/raw.h"
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#include "audio/decoders/vorbis.h"
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#include "audio/decoders/wave.h"
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#include "audio/decoders/xa.h"
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namespace Sword1 {
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#define SOUND_SPEECH_ID 1
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#define SPEECH_FLAGS (Audio::FLAG_16BITS | Audio::FLAG_LITTLE_ENDIAN)
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Sound::Sound(const char *searchPath, Audio::Mixer *mixer, ResMan *pResMan)
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: _rnd("sword1sound") {
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strcpy(_filePath, searchPath);
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_mixer = mixer;
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_resMan = pResMan;
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_bigEndianSpeech = false;
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_cowHeader = NULL;
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_endOfQueue = 0;
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_currentCowFile = 0;
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_speechVolL = _speechVolR = _sfxVolL = _sfxVolR = 192;
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}
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Sound::~Sound() {
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// clean up fx queue
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_mixer->stopAll();
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for (uint8 cnt = 0; cnt < _endOfQueue; cnt++)
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if (_fxQueue[cnt].delay == 0)
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_resMan->resClose(getSampleId(_fxQueue[cnt].id));
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_endOfQueue = 0;
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closeCowSystem();
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}
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uint32 Sound::getSampleId(int32 fxNo) {
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byte cluster = _fxList[fxNo].sampleId.cluster;
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byte id;
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if (SwordEngine::_systemVars.isDemo && SwordEngine::_systemVars.platform == Common::kPlatformWindows) {
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id = _fxList[fxNo].sampleId.idWinDemo;
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} else {
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id = _fxList[fxNo].sampleId.idStd;
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}
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return (cluster << 24) | id;
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}
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void Sound::checkSpeechFileEndianness() {
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// Some mac versions (not all of them) use big endian wav, although
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// the wav header doesn't indicate it.
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// Use heuristic to determine endianness of speech.
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// The heuristic consist in computing the sum of the absolute difference for
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// every two consecutive samples. This is done both with a big endian and a
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// little endian assumption. The one with the smallest sum should be the
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// correct one (the sound wave is supposed to be relatively smooth).
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// It needs at least 1000 samples to get stable result (the code below is
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// using the first 2000 samples of the wav sound).
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// Init speech file if not already done.
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if (!_currentCowFile) {
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// Open one of the speech files. It uses SwordEngine::_systemVars.currentCD
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// to decide which file to open, therefore if it is currently set to zero
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// we have to set it to either 1 or 2 (I decided to set it to 1 as this is
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// more likely to be the first file that will be needed).
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bool no_current_cd = false;
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if (SwordEngine::_systemVars.currentCD == 0) {
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SwordEngine::_systemVars.currentCD = 1;
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no_current_cd = true;
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}
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initCowSystem();
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if (no_current_cd) {
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// In case it fails with CD1 retry with CD2
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if (!_currentCowFile) {
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SwordEngine::_systemVars.currentCD = 2;
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initCowSystem();
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}
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// Reset currentCD flag
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SwordEngine::_systemVars.currentCD = 0;
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}
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}
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// Testing for endianness makes sense only if using the uncompressed files.
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if (_cowHeader == NULL || (_cowMode != CowWave && _cowMode != CowDemo))
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return;
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// I picked the sample to use randomly (I just made sure it is long enough so that there is
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// a fair change of the heuristic to have a stable result and work for every language).
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int roomNo = _currentCowFile == 1 ? 1 : 129;
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int localNo = _currentCowFile == 1 ? 2 : 933;
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// Get the speech data and apply the heuristic
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uint32 locIndex = _cowHeader[roomNo] >> 2;
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uint32 sampleSize = _cowHeader[locIndex + (localNo * 2)];
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uint32 index = _cowHeader[locIndex + (localNo * 2) - 1];
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if (sampleSize) {
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uint32 size;
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double be_diff_sum = 0., le_diff_sum = 0.;
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_bigEndianSpeech = false;
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int16 *data = uncompressSpeech(index + _cowHeaderSize, sampleSize, &size);
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// Compute average of difference between two consecutive samples for both BE and LE
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if (data) {
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if (size > 4000)
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size = 2000;
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else
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size /= 2;
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int16 prev_be_value = (int16)SWAP_BYTES_16(*((uint16 *)(data)));
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for (uint32 i = 1; i < size; ++i) {
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le_diff_sum += fabs((double)(data[i] - data[i - 1]));
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int16 be_value = (int16)SWAP_BYTES_16(*((uint16 *)(data + i)));
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be_diff_sum += fabs((double)(be_value - prev_be_value));
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prev_be_value = be_value;
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}
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delete[] data;
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}
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// Set the big endian flag
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_bigEndianSpeech = (be_diff_sum < le_diff_sum);
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if (_bigEndianSpeech)
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debug(6, "Mac version: using big endian speech file");
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else
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debug(6, "Mac version: using little endian speech file");
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debug(8, "Speech endianness heuristic: average = %f for BE and %f for LE, computed on %d samples)", be_diff_sum / (size - 1), le_diff_sum / (size - 1), size);
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}
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}
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int Sound::addToQueue(int32 fxNo) {
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bool alreadyInQueue = false;
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for (uint8 cnt = 0; (cnt < _endOfQueue) && (!alreadyInQueue); cnt++)
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if (_fxQueue[cnt].id == (uint32)fxNo)
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alreadyInQueue = true;
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if (!alreadyInQueue) {
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if (_endOfQueue == MAX_FXQ_LENGTH) {
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warning("Sound queue overflow");
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return 0;
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}
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uint32 sampleId = getSampleId(fxNo);
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if ((sampleId & 0xFF) != 0xFF) {
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_resMan->resOpen(sampleId);
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_fxQueue[_endOfQueue].id = fxNo;
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if (_fxList[fxNo].type == FX_SPOT)
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_fxQueue[_endOfQueue].delay = _fxList[fxNo].delay + 1;
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else
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_fxQueue[_endOfQueue].delay = 1;
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_endOfQueue++;
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return 1;
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}
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return 0;
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}
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return 0;
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}
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void Sound::engine() {
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// first of all, add any random sfx to the queue...
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for (uint16 cnt = 0; cnt < TOTAL_FX_PER_ROOM; cnt++) {
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uint16 fxNo = _roomsFixedFx[Logic::_scriptVars[SCREEN]][cnt];
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if (fxNo) {
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if (_fxList[fxNo].type == FX_RANDOM) {
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if (_rnd.getRandomNumber(_fxList[fxNo].delay) == 0)
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addToQueue(fxNo);
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}
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} else
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break;
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}
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// now process the queue
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for (uint8 cnt2 = 0; cnt2 < _endOfQueue; cnt2++) {
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if (_fxQueue[cnt2].delay > 0) {
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_fxQueue[cnt2].delay--;
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if (_fxQueue[cnt2].delay == 0)
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playSample(&_fxQueue[cnt2]);
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} else {
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if (!_mixer->isSoundHandleActive(_fxQueue[cnt2].handle)) { // sound finished
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_resMan->resClose(getSampleId(_fxQueue[cnt2].id));
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if (cnt2 != _endOfQueue - 1)
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_fxQueue[cnt2] = _fxQueue[_endOfQueue - 1];
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_endOfQueue--;
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}
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}
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}
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}
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void Sound::fnStopFx(int32 fxNo) {
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_mixer->stopID(fxNo);
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for (uint8 cnt = 0; cnt < _endOfQueue; cnt++)
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if (_fxQueue[cnt].id == (uint32)fxNo) {
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if (!_fxQueue[cnt].delay) // sound was started
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_resMan->resClose(getSampleId(_fxQueue[cnt].id));
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if (cnt != _endOfQueue - 1)
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_fxQueue[cnt] = _fxQueue[_endOfQueue - 1];
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_endOfQueue--;
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return;
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}
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debug(8, "fnStopFx: id not found in queue");
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}
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bool Sound::amISpeaking() {
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_waveVolPos++;
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return _waveVolume[_waveVolPos - 1];
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}
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bool Sound::speechFinished() {
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return !_mixer->isSoundHandleActive(_speechHandle);
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}
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void Sound::newScreen(uint32 screen) {
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if (_currentCowFile != SwordEngine::_systemVars.currentCD) {
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if (_cowFile.isOpen())
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closeCowSystem();
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initCowSystem();
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}
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// Start the room's looping sounds.
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for (uint16 cnt = 0; cnt < TOTAL_FX_PER_ROOM; cnt++) {
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uint16 fxNo = _roomsFixedFx[screen][cnt];
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if (fxNo) {
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if (_fxList[fxNo].type == FX_LOOP)
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addToQueue(fxNo);
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} else
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break;
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}
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}
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void Sound::quitScreen() {
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// stop all running SFX
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while (_endOfQueue)
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fnStopFx(_fxQueue[0].id);
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}
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void Sound::playSample(QueueElement *elem) {
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uint8 *sampleData = (uint8 *)_resMan->fetchRes(getSampleId(elem->id));
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for (uint16 cnt = 0; cnt < MAX_ROOMS_PER_FX; cnt++) {
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if (_fxList[elem->id].roomVolList[cnt].roomNo) {
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if ((_fxList[elem->id].roomVolList[cnt].roomNo == (int)Logic::_scriptVars[SCREEN]) ||
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(_fxList[elem->id].roomVolList[cnt].roomNo == -1)) {
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uint8 volL = (_fxList[elem->id].roomVolList[cnt].leftVol * 10 * _sfxVolL) / 255;
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uint8 volR = (_fxList[elem->id].roomVolList[cnt].rightVol * 10 * _sfxVolR) / 255;
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int8 pan = (volR - volL) / 2;
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uint8 volume = (volR + volL) / 2;
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if (SwordEngine::isPsx()) {
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// We ignore FX_LOOP as XA has its own looping mechanism
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uint32 size = READ_LE_UINT32(sampleData);
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Audio::AudioStream *audStream = Audio::makeXAStream(new Common::MemoryReadStream(sampleData + 4, size - 4), 11025);
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_mixer->playStream(Audio::Mixer::kSFXSoundType, &elem->handle, audStream, elem->id, volume, pan);
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} else {
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uint32 size = READ_LE_UINT32(sampleData + 0x28);
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uint8 flags;
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if (READ_LE_UINT16(sampleData + 0x22) == 16)
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flags = Audio::FLAG_16BITS | Audio::FLAG_LITTLE_ENDIAN;
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else
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flags = Audio::FLAG_UNSIGNED;
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if (READ_LE_UINT16(sampleData + 0x16) == 2)
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flags |= Audio::FLAG_STEREO;
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Audio::AudioStream *stream = Audio::makeLoopingAudioStream(
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Audio::makeRawStream(sampleData + 0x2C, size, 11025, flags, DisposeAfterUse::NO),
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(_fxList[elem->id].type == FX_LOOP) ? 0 : 1);
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_mixer->playStream(Audio::Mixer::kSFXSoundType, &elem->handle, stream, elem->id, volume, pan);
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}
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}
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} else
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break;
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}
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}
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bool Sound::startSpeech(uint16 roomNo, uint16 localNo) {
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if (_cowHeader == NULL) {
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warning("Sound::startSpeech: COW file isn't open");
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return false;
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}
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uint32 locIndex = 0xFFFFFFFF;
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uint32 sampleSize = 0;
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uint32 index = 0;
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if (_cowMode == CowPSX) {
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Common::File file;
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uint16 i;
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if (!file.open("speech.lis")) {
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warning("Could not open speech.lis");
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return false;
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}
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for (i = 0; !file.eos() && !file.err(); i++)
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if (file.readUint16LE() == roomNo) {
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locIndex = i;
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break;
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}
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file.close();
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if (locIndex == 0xFFFFFFFF) {
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warning("Could not find room %d in speech.lis", roomNo);
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return false;
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}
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if (!file.open("speech.inf")) {
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warning("Could not open speech.inf");
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return false;
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}
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uint16 numRooms = file.readUint16LE(); // Read number of rooms referenced in this file
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file.seek(locIndex * 4 + 2); // 4 bytes per room, skip first 2 bytes
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uint16 numLines = file.readUint16LE();
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uint16 roomOffset = file.readUint16LE();
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file.seek(2 + numRooms * 4 + roomOffset * 2); // The offset is in terms of uint16's, so multiply by 2. Skip the room indexes too.
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locIndex = 0xFFFFFFFF;
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for (i = 0; i < numLines; i++)
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if (file.readUint16LE() == localNo) {
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locIndex = i;
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break;
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}
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if (locIndex == 0xFFFFFFFF) {
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warning("Could not find local number %d in room %d in speech.inf", roomNo, localNo);
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return false;
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}
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file.close();
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index = _cowHeader[(roomOffset + locIndex) * 2];
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sampleSize = _cowHeader[(roomOffset + locIndex) * 2 + 1];
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} else {
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locIndex = _cowHeader[roomNo] >> 2;
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sampleSize = _cowHeader[locIndex + (localNo * 2)];
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index = _cowHeader[locIndex + (localNo * 2) - 1];
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}
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debug(6, "startSpeech(%d, %d): locIndex %d, sampleSize %d, index %d", roomNo, localNo, locIndex, sampleSize, index);
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Audio::AudioStream *stream = 0;
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if (sampleSize) {
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uint8 speechVol = (_speechVolR + _speechVolL) / 2;
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int8 speechPan = (_speechVolR - _speechVolL) / 2;
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if ((_cowMode == CowWave) || (_cowMode == CowDemo)) {
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uint32 size;
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int16 *data = uncompressSpeech(index + _cowHeaderSize, sampleSize, &size);
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if (data) {
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stream = Audio::makeRawStream((byte *)data, size, 11025, SPEECH_FLAGS);
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_mixer->playStream(Audio::Mixer::kSpeechSoundType, &_speechHandle, stream, SOUND_SPEECH_ID, speechVol, speechPan);
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}
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} else if (_cowMode == CowPSX && sampleSize != 0xffffffff) {
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_cowFile.seek(index * 2048);
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Common::SeekableReadStream *tmp = _cowFile.readStream(sampleSize);
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assert(tmp);
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stream = Audio::makeXAStream(tmp, 11025);
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_mixer->playStream(Audio::Mixer::kSpeechSoundType, &_speechHandle, stream, SOUND_SPEECH_ID, speechVol, speechPan);
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// with compressed audio, we can't calculate the wave volume.
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// so default to talking.
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for (int cnt = 0; cnt < 480; cnt++)
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_waveVolume[cnt] = true;
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_waveVolPos = 0;
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}
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#ifdef USE_FLAC
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else if (_cowMode == CowFLAC) {
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_cowFile.seek(index);
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Common::SeekableReadStream *tmp = _cowFile.readStream(sampleSize);
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assert(tmp);
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stream = Audio::makeFLACStream(tmp, DisposeAfterUse::YES);
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_mixer->playStream(Audio::Mixer::kSpeechSoundType, &_speechHandle, stream, SOUND_SPEECH_ID, speechVol, speechPan);
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// with compressed audio, we can't calculate the wave volume.
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// so default to talking.
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for (int cnt = 0; cnt < 480; cnt++)
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_waveVolume[cnt] = true;
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_waveVolPos = 0;
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}
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#endif
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#ifdef USE_VORBIS
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else if (_cowMode == CowVorbis) {
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_cowFile.seek(index);
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Common::SeekableReadStream *tmp = _cowFile.readStream(sampleSize);
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assert(tmp);
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stream = Audio::makeVorbisStream(tmp, DisposeAfterUse::YES);
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_mixer->playStream(Audio::Mixer::kSpeechSoundType, &_speechHandle, stream, SOUND_SPEECH_ID, speechVol, speechPan);
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// with compressed audio, we can't calculate the wave volume.
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// so default to talking.
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for (int cnt = 0; cnt < 480; cnt++)
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_waveVolume[cnt] = true;
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_waveVolPos = 0;
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}
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#endif
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#ifdef USE_MAD
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else if (_cowMode == CowMP3) {
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_cowFile.seek(index);
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Common::SeekableReadStream *tmp = _cowFile.readStream(sampleSize);
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assert(tmp);
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stream = Audio::makeMP3Stream(tmp, DisposeAfterUse::YES);
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_mixer->playStream(Audio::Mixer::kSpeechSoundType, &_speechHandle, stream, SOUND_SPEECH_ID, speechVol, speechPan);
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// with compressed audio, we can't calculate the wave volume.
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// so default to talking.
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for (int cnt = 0; cnt < 480; cnt++)
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_waveVolume[cnt] = true;
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_waveVolPos = 0;
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}
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#endif
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return true;
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} else
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return false;
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}
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int16 *Sound::uncompressSpeech(uint32 index, uint32 cSize, uint32 *size) {
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uint8 *fBuf = (uint8 *)malloc(cSize);
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_cowFile.seek(index);
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_cowFile.read(fBuf, cSize);
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uint32 headerPos = 0;
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while ((READ_BE_UINT32(fBuf + headerPos) != 'data') && (headerPos < 100))
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headerPos++;
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if (headerPos < 100) {
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int32 resSize;
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int16 *srcData;
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uint32 srcPos;
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int16 length;
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cSize /= 2;
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headerPos += 4; // skip 'data' tag
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if (_cowMode != CowDemo) {
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resSize = READ_LE_UINT32(fBuf + headerPos) >> 1;
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headerPos += 4;
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} else {
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// the demo speech files have the uncompressed size
|
|
// embedded in the compressed stream *sigh*
|
|
//
|
|
// But not always, apparently. See bug #2182450. Is
|
|
// there any way to figure out the size other than
|
|
// decoding the sound in that case?
|
|
|
|
if (fBuf[headerPos + 1] == 0) {
|
|
if (READ_LE_UINT16(fBuf + headerPos) == 1) {
|
|
resSize = READ_LE_UINT16(fBuf + headerPos + 2);
|
|
resSize |= READ_LE_UINT16(fBuf + headerPos + 6) << 16;
|
|
} else
|
|
resSize = READ_LE_UINT32(fBuf + headerPos + 2);
|
|
resSize >>= 1;
|
|
} else {
|
|
resSize = 0;
|
|
srcData = (int16 *)fBuf;
|
|
srcPos = headerPos >> 1;
|
|
while (srcPos < cSize) {
|
|
length = (int16)READ_LE_UINT16(srcData + srcPos);
|
|
srcPos++;
|
|
if (length < 0) {
|
|
resSize -= length;
|
|
srcPos++;
|
|
} else {
|
|
resSize += length;
|
|
srcPos += length;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
assert(!(headerPos & 1));
|
|
srcData = (int16 *)fBuf;
|
|
srcPos = headerPos >> 1;
|
|
uint32 dstPos = 0;
|
|
int16 *dstData = (int16 *)malloc(resSize * 2);
|
|
int32 samplesLeft = resSize;
|
|
while (srcPos < cSize && samplesLeft > 0) {
|
|
length = (int16)(_bigEndianSpeech ? READ_BE_UINT16(srcData + srcPos) : READ_LE_UINT16(srcData + srcPos));
|
|
srcPos++;
|
|
if (length < 0) {
|
|
length = -length;
|
|
if (length > samplesLeft)
|
|
length = samplesLeft;
|
|
int16 value;
|
|
if (_bigEndianSpeech) {
|
|
value = (int16)SWAP_BYTES_16(*((uint16 *)(srcData + srcPos)));
|
|
} else {
|
|
value = srcData[srcPos];
|
|
}
|
|
for (uint16 cnt = 0; cnt < (uint16)length; cnt++)
|
|
dstData[dstPos++] = value;
|
|
srcPos++;
|
|
} else {
|
|
if (length > samplesLeft)
|
|
length = samplesLeft;
|
|
if (_bigEndianSpeech) {
|
|
for (uint16 cnt = 0; cnt < (uint16)length; cnt++)
|
|
dstData[dstPos++] = (int16)SWAP_BYTES_16(*((uint16 *)(srcData + (srcPos++))));
|
|
} else {
|
|
memcpy(dstData + dstPos, srcData + srcPos, length * 2);
|
|
dstPos += length;
|
|
srcPos += length;
|
|
}
|
|
}
|
|
samplesLeft -= length;
|
|
}
|
|
if (samplesLeft > 0) {
|
|
memset(dstData + dstPos, 0, samplesLeft * 2);
|
|
}
|
|
if (_cowMode == CowDemo) // demo has wave output size embedded in the compressed data
|
|
*(uint32 *)dstData = 0;
|
|
free(fBuf);
|
|
*size = resSize * 2;
|
|
calcWaveVolume(dstData, resSize);
|
|
return dstData;
|
|
} else {
|
|
free(fBuf);
|
|
warning("Sound::uncompressSpeech(): DATA tag not found in wave header");
|
|
*size = 0;
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
void Sound::calcWaveVolume(int16 *data, uint32 length) {
|
|
int16 *blkPos = data + 918;
|
|
uint32 cnt;
|
|
for (cnt = 0; cnt < WAVE_VOL_TAB_LENGTH; cnt++)
|
|
_waveVolume[cnt] = false;
|
|
_waveVolPos = 0;
|
|
for (uint32 blkCnt = 1; blkCnt < length / 918; blkCnt++) {
|
|
if (blkCnt >= WAVE_VOL_TAB_LENGTH) {
|
|
warning("Wave vol tab too small");
|
|
return;
|
|
}
|
|
int32 average = 0;
|
|
for (cnt = 0; cnt < 918; cnt++)
|
|
average += blkPos[cnt];
|
|
average /= 918;
|
|
uint32 diff = 0;
|
|
for (cnt = 0; cnt < 918; cnt++) {
|
|
int16 smpDiff = *blkPos - average;
|
|
diff += (uint32)ABS(smpDiff);
|
|
blkPos++;
|
|
}
|
|
if (diff > WAVE_VOL_THRESHOLD)
|
|
_waveVolume[blkCnt - 1] = true;
|
|
}
|
|
}
|
|
|
|
void Sound::stopSpeech() {
|
|
_mixer->stopID(SOUND_SPEECH_ID);
|
|
}
|
|
|
|
void Sound::initCowSystem() {
|
|
if (SwordEngine::_systemVars.currentCD == 0)
|
|
return;
|
|
|
|
char cowName[25];
|
|
/* look for speech1/2.clu in the data dir
|
|
and speech/speech.clu (running from cd or using cd layout)
|
|
*/
|
|
#ifdef USE_FLAC
|
|
if (!_cowFile.isOpen()) {
|
|
sprintf(cowName, "SPEECH%d.CLF", SwordEngine::_systemVars.currentCD);
|
|
_cowFile.open(cowName);
|
|
if (_cowFile.isOpen()) {
|
|
debug(1, "Using FLAC compressed Speech Cluster");
|
|
_cowMode = CowFLAC;
|
|
}
|
|
}
|
|
#endif
|
|
#ifdef USE_VORBIS
|
|
if (!_cowFile.isOpen()) {
|
|
sprintf(cowName, "SPEECH%d.CLV", SwordEngine::_systemVars.currentCD);
|
|
_cowFile.open(cowName);
|
|
if (_cowFile.isOpen()) {
|
|
debug(1, "Using Vorbis compressed Speech Cluster");
|
|
_cowMode = CowVorbis;
|
|
}
|
|
}
|
|
#endif
|
|
#ifdef USE_MAD
|
|
if (!_cowFile.isOpen()) {
|
|
sprintf(cowName, "SPEECH%d.CL3", SwordEngine::_systemVars.currentCD);
|
|
_cowFile.open(cowName);
|
|
if (_cowFile.isOpen()) {
|
|
debug(1, "Using MP3 compressed Speech Cluster");
|
|
_cowMode = CowMP3;
|
|
}
|
|
}
|
|
#endif
|
|
if (!_cowFile.isOpen()) {
|
|
sprintf(cowName, "SPEECH%d.CLU", SwordEngine::_systemVars.currentCD);
|
|
_cowFile.open(cowName);
|
|
if (!_cowFile.isOpen()) {
|
|
_cowFile.open("speech.clu");
|
|
}
|
|
debug(1, "Using uncompressed Speech Cluster");
|
|
_cowMode = CowWave;
|
|
}
|
|
|
|
if (SwordEngine::isPsx()) {
|
|
// There's only one file on the PSX, so set it to the current disc.
|
|
_currentCowFile = SwordEngine::_systemVars.currentCD;
|
|
if (!_cowFile.isOpen()) {
|
|
if (!_cowFile.open("speech.dat"))
|
|
error("Could not open speech.dat");
|
|
_cowMode = CowPSX;
|
|
}
|
|
}
|
|
|
|
if (!_cowFile.isOpen())
|
|
_cowFile.open("speech.clu");
|
|
|
|
if (!_cowFile.isOpen()) {
|
|
_cowFile.open("cows.mad");
|
|
if (_cowFile.isOpen())
|
|
_cowMode = CowDemo;
|
|
}
|
|
|
|
if (_cowFile.isOpen()) {
|
|
if (SwordEngine::isPsx()) {
|
|
// Get data from the external table file
|
|
Common::File tableFile;
|
|
if (!tableFile.open("speech.tab"))
|
|
error("Could not open speech.tab");
|
|
_cowHeaderSize = tableFile.size();
|
|
_cowHeader = (uint32 *)malloc(_cowHeaderSize);
|
|
if (_cowHeaderSize & 3)
|
|
error("Unexpected cow header size %d", _cowHeaderSize);
|
|
for (uint32 cnt = 0; cnt < _cowHeaderSize / 4; cnt++)
|
|
_cowHeader[cnt] = tableFile.readUint32LE();
|
|
} else {
|
|
_cowHeaderSize = _cowFile.readUint32LE();
|
|
_cowHeader = (uint32 *)malloc(_cowHeaderSize);
|
|
if (_cowHeaderSize & 3)
|
|
error("Unexpected cow header size %d", _cowHeaderSize);
|
|
for (uint32 cnt = 0; cnt < (_cowHeaderSize / 4) - 1; cnt++)
|
|
_cowHeader[cnt] = _cowFile.readUint32LE();
|
|
_currentCowFile = SwordEngine::_systemVars.currentCD;
|
|
}
|
|
} else
|
|
warning("Sound::initCowSystem: Can't open SPEECH%d.CLU", SwordEngine::_systemVars.currentCD);
|
|
}
|
|
|
|
void Sound::closeCowSystem() {
|
|
_cowFile.close();
|
|
free(_cowHeader);
|
|
_cowHeader = NULL;
|
|
_currentCowFile = 0;
|
|
}
|
|
|
|
} // End of namespace Sword1
|