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|
/* Copyright (C) 1994-1998 Revolution Software Ltd.
* Copyright (C) 2003-2005 The ScummVM project
*
* 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*
* $Header$
*/
// One feature still missing is the original's DipMusic() function which, as
// far as I can understand, softened the music volume when someone was
// speaking, but only (?) if the music was playing loudly at the time.
//
// All things considered, I think this is more bother than it's worth.
#include "common/stdafx.h"
#include "common/file.h"
#include "common/system.h"
#include "sound/mp3.h"
#include "sound/vorbis.h"
#include "sound/flac.h"
#include "sound/rate.h"
#include "sound/wave.h"
#include "sword2/sword2.h"
#include "sword2/resman.h"
#include "sword2/sound.h"
namespace Sword2 {
static AudioStream *makeCLUStream(File *fp, int size);
static AudioStream *getAudioStream(SoundFileHandle *fh, const char *base, int cd, uint32 id, uint32 *numSamples) {
if (!fh->file->isOpen()) {
struct {
const char *ext;
int mode;
} file_types[] = {
#ifdef USE_MAD
{ "cl3", kMP3Mode },
#endif
#ifdef USE_VORBIS
{ "clg", kVorbisMode },
#endif
#ifdef USE_FLAC
{ "clf", kFlacMode },
#endif
{ "clu", kCLUMode }
};
int soundMode = -1;
char filename[20];
for (int i = 0; i < ARRAYSIZE(file_types); i++) {
File f;
sprintf(filename, "%s%d.%s", base, cd, file_types[i].ext);
if (f.open(filename)) {
soundMode = file_types[i].mode;
break;
}
sprintf(filename, "%s.%s", base, file_types[i].ext);
if (f.open(filename)) {
soundMode = file_types[i].mode;
break;
}
}
if (soundMode == 0)
return NULL;
fh->file->open(filename);
fh->fileType = soundMode;
if (!fh->file->isOpen()) {
warning("Very strange fopen error");
return NULL;
}
if (fh->fileSize != fh->file->size()) {
if (fh->idxTab) {
free(fh->idxTab);
fh->idxTab = NULL;
}
}
}
uint32 entrySize = (fh->fileType == kCLUMode) ? 2 : 3;
if (!fh->idxTab) {
fh->file->seek(0);
fh->idxLen = fh->file->readUint32LE();
fh->file->seek(entrySize * 4);
fh->idxTab = (uint32*)malloc(fh->idxLen * 3 * sizeof(uint32));
for (uint32 cnt = 0; cnt < fh->idxLen; cnt++) {
fh->idxTab[cnt * 3 + 0] = fh->file->readUint32LE();
fh->idxTab[cnt * 3 + 1] = fh->file->readUint32LE();
if (fh->fileType == kCLUMode)
fh->idxTab[cnt * 3 + 2] = fh->idxTab[cnt * 3 + 1] + 1;
else
fh->idxTab[cnt * 3 + 2] = fh->file->readUint32LE();
}
}
uint32 pos = fh->idxTab[id * 3 + 0];
uint32 len = fh->idxTab[id * 3 + 1];
uint32 enc_len = fh->idxTab[id * 3 + 2];
if (numSamples)
*numSamples = len;
if (!pos || !len) {
fh->file->close();
return NULL;
}
fh->file->seek(pos, SEEK_SET);
switch (fh->fileType) {
case kCLUMode:
return makeCLUStream(fh->file, enc_len);
#ifdef USE_MAD
case kMP3Mode:
return makeMP3Stream(fh->file, enc_len);
#endif
#ifdef USE_VORBIS
case kVorbisMode:
return makeVorbisStream(fh->file, enc_len);
#endif
#ifdef USE_FLAC
case kFlacMode:
return makeFlacStream(fh->file, enc_len);
#endif
default:
return NULL;
}
}
// ----------------------------------------------------------------------------
// Custom AudioStream class to handle Broken Sword II's audio compression.
// ----------------------------------------------------------------------------
#define GetCompressedShift(n) ((n) >> 4)
#define GetCompressedSign(n) (((n) >> 3) & 1)
#define GetCompressedAmplitude(n) ((n) & 7)
CLUInputStream::CLUInputStream(File *file, int size)
: _file(file), _firstTime(true), _bufferEnd(_outbuf + BUFFER_SIZE) {
_file->incRef();
// Determine the end position.
_file_pos = _file->pos();
_end_pos = _file_pos + size;
// Read in initial data
refill();
}
CLUInputStream::~CLUInputStream() {
_file->decRef();
}
int CLUInputStream::readBuffer(int16 *buffer, const int numSamples) {
int samples = 0;
while (samples < numSamples && !eosIntern()) {
const int len = MIN(numSamples - samples, (int) (_bufferEnd - _pos));
memcpy(buffer, _pos, len * 2);
buffer += len;
_pos += len;
samples += len;
if (_pos >= _bufferEnd) {
refill();
}
}
return samples;
}
void CLUInputStream::refill() {
byte *in = _inbuf;
int16 *out = _outbuf;
_file->seek(_file_pos, SEEK_SET);
uint len_left = _file->read(in, MIN((uint32) BUFFER_SIZE, _end_pos - _file->pos()));
_file_pos = _file->pos();
while (len_left > 0) {
uint16 sample;
if (_firstTime) {
_firstTime = false;
_prev = READ_LE_UINT16(in);
sample = _prev;
len_left -= 2;
in += 2;
} else {
uint16 delta = GetCompressedAmplitude(*in) << GetCompressedShift(*in);
if (GetCompressedSign(*in))
sample = _prev - delta;
else
sample = _prev + delta;
_prev = sample;
len_left--;
in++;
}
*out++ = sample;
}
_pos = _outbuf;
_bufferEnd = out;
}
AudioStream *makeCLUStream(File *file, int size) {
return new CLUInputStream(file, size);
}
// ----------------------------------------------------------------------------
// Another custom AudioStream class, to wrap around the various AudioStream
// classes used for music decompression, and to add looping, fading, etc.
// ----------------------------------------------------------------------------
// The length of a fade-in/out, in milliseconds.
#define FADE_LENGTH 3000
MusicInputStream::MusicInputStream(int cd, SoundFileHandle *fh, uint32 musicId, bool looping) {
_cd = cd;
_fh = fh;
_musicId = musicId;
_looping = looping;
_bufferEnd = _buffer + BUFFER_SIZE;
_remove = false;
_fading = 0;
_decoder = getAudioStream(_fh, "music", _cd, _musicId, &_numSamples);
if (_decoder) {
_samplesLeft = _numSamples;
_fadeSamples = (getRate() * FADE_LENGTH) / 1000;
fadeUp();
// Read in initial data
refill();
}
}
MusicInputStream::~MusicInputStream() {
delete _decoder;
}
int MusicInputStream::readBuffer(int16 *buffer, const int numSamples) {
if (!_decoder)
return 0;
int samples = 0;
while (samples < numSamples && !eosIntern()) {
const int len = MIN(numSamples - samples, (int) (_bufferEnd - _pos));
memcpy(buffer, _pos, len * 2);
buffer += len;
_pos += len;
samples += len;
if (_pos >= _bufferEnd) {
refill();
}
}
return samples;
}
void MusicInputStream::refill() {
int16 *buf = _buffer;
uint32 numSamples = 0;
uint32 len_left;
bool endFade = false;
len_left = BUFFER_SIZE;
if (_fading > 0 && (uint32) _fading < len_left)
len_left = _fading;
if (_samplesLeft < len_left)
len_left = _samplesLeft;
if (!_looping) {
// None-looping music is faded out at the end. If this fade
// out would have started somewhere within the len_left samples
// to read, we only read up to that point. This way, we can
// treat this fade as any other.
if (!_fading) {
uint32 currentlyAt = _numSamples - _samplesLeft;
uint32 fadeOutAt = _numSamples - _fadeSamples;
uint32 readTo = currentlyAt + len_left;
if (fadeOutAt == currentlyAt)
fadeDown();
else if (fadeOutAt > currentlyAt && fadeOutAt <= readTo) {
len_left = fadeOutAt - currentlyAt;
endFade = true;
}
}
}
int desired = len_left - numSamples;
int len = _decoder->readBuffer(buf, desired);
// Shouldn't happen, but if it does it could cause an infinite loop.
// Of course there were bugs that caused it to happen several times
// during development. :-)
if (len < desired) {
warning("Expected %d samples, but got none", desired);
if (!len)
return;
}
buf += len;
numSamples += len;
len_left -= len;
_samplesLeft -= len;
int16 *ptr;
if (_fading > 0) {
// Fade down
for (ptr = _buffer; ptr < buf; ptr++) {
if (_fading > 0) {
_fading--;
*ptr = (*ptr * _fading) / _fadeSamples;
}
if (_fading == 0) {
_looping = false;
_remove = true;
*ptr = 0;
}
}
} else if (_fading < 0) {
// Fade up
for (ptr = _buffer; ptr < buf; ptr++) {
_fading--;
*ptr = -(*ptr * _fading) / _fadeSamples;
if (_fading <= -_fadeSamples) {
_fading = 0;
break;
}
}
}
if (endFade)
fadeDown();
if (!_samplesLeft) {
if (_looping) {
delete _decoder;
_decoder = getAudioStream(_fh, "music", _cd, _musicId, &_numSamples);
_samplesLeft = _numSamples;
} else
_remove = true;
}
_pos = _buffer;
_bufferEnd = buf;
}
void MusicInputStream::fadeUp() {
if (_fading > 0)
_fading = -_fading;
else if (_fading == 0)
_fading = -1;
}
void MusicInputStream::fadeDown() {
if (_fading < 0)
_fading = -_fading;
else if (_fading == 0)
_fading = _fadeSamples;
}
bool MusicInputStream::readyToRemove() {
return _remove;
}
int32 MusicInputStream::getTimeRemaining() {
// This is far from exact, but it doesn't have to be.
return (_samplesLeft + BUFFER_SIZE) / getRate();
}
// ----------------------------------------------------------------------------
// Main sound class
// ----------------------------------------------------------------------------
// AudioStream API
int Sound::readBuffer(int16 *buffer, const int numSamples) {
Common::StackLock lock(_mutex);
int i;
if (_musicPaused)
return 0;
for (i = 0; i < MAXMUS; i++) {
if (_music[i] && _music[i]->readyToRemove()) {
delete _music[i];
_music[i] = NULL;
}
}
memset(buffer, 0, 2 * numSamples);
if (!_mixBuffer || numSamples > _mixBufferLen) {
if (_mixBuffer)
_mixBuffer = (int16 *) realloc(_mixBuffer, 2 * numSamples);
else
_mixBuffer = (int16 *) malloc(2 * numSamples);
_mixBufferLen = numSamples;
}
if (!_mixBuffer)
return 0;
for (i = 0; i < MAXMUS; i++) {
if (!_music[i])
continue;
int len = _music[i]->readBuffer(_mixBuffer, numSamples);
if (!_musicMuted) {
for (int j = 0; j < len; j++) {
clampedAdd(buffer[j], _mixBuffer[j]);
}
}
}
bool inUse[MAXMUS];
for (i = 0; i < MAXMUS; i++)
inUse[i] = false;
for (i = 0; i < MAXMUS; i++) {
if (_music[i]) {
if (_music[i]->whichCd() == 1)
inUse[0] = true;
else
inUse[1] = true;
}
}
for (i = 0; i < MAXMUS; i++) {
if (!inUse[i] && !_musicFile[i].inUse && _musicFile[i].file->isOpen())
_musicFile[i].file->close();
}
return numSamples;
}
bool Sound::endOfData() const {
for (int i = 0; i < MAXMUS; i++) {
if (_musicFile[i].file->isOpen())
return false;
}
return true;
}
// ----------------------------------------------------------------------------
// MUSIC
// ----------------------------------------------------------------------------
/**
* Stops the music dead in its tracks. Any music that is currently being
* streamed is paused.
*/
void Sound::pauseMusic() {
Common::StackLock lock(_mutex);
_musicPaused = true;
}
/**
* Restarts the music from where it was stopped.
*/
void Sound::unpauseMusic() {
Common::StackLock lock(_mutex);
_musicPaused = false;
}
/**
* Fades out and stops the music.
*/
void Sound::stopMusic(bool immediately) {
Common::StackLock lock(_mutex);
_loopingMusicId = 0;
for (int i = 0; i < MAXMUS; i++) {
if (_music[i]) {
if (immediately) {
delete _music[i];
_music[i] = NULL;
} else
_music[i]->fadeDown();
}
}
}
/**
* Streams music from a cluster file.
* @param musicId the id of the music to stream
* @param looping true if the music is to loop back to the start
* @return RD_OK or an error code
*/
int32 Sound::streamCompMusic(uint32 musicId, bool loop) {
//Common::StackLock lock(_mutex);
_mutex.lock();
int cd = _vm->_resman->whichCd();
if (loop)
_loopingMusicId = musicId;
else
_loopingMusicId = 0;
int primary = -1;
int secondary = -1;
// If both music streams are active, one of them will have to go.
if (_music[0] && _music[1]) {
int32 fade0 = _music[0]->isFading();
int32 fade1 = _music[1]->isFading();
if (!fade0 && !fade1) {
// Neither is fading. This shouldn't happen, so just
// pick one and be done with it.
primary = 0;
} else if (fade0 && !fade1) {
// Stream 0 is fading, so pick that one.
primary = 0;
} else if (!fade0 && fade1) {
// Stream 1 is fading, so pick that one.
primary = 1;
} else {
// Both streams are fading. Pick the one that is
// closest to silent.
if (ABS(fade0) < ABS(fade1))
primary = 0;
else
primary = 1;
}
delete _music[primary];
_music[primary] = NULL;
}
// Pick the available music stream. If no music is playing it doesn't
// matter which we use.
if (_music[0] || _music[1]) {
if (_music[0]) {
primary = 1;
secondary = 0;
} else {
primary = 0;
secondary = 1;
}
} else
primary = 0;
// Don't start streaming if the volume is off.
if (isMusicMute()) {
_mutex.unlock();
return RD_OK;
}
if (secondary != -1)
_music[secondary]->fadeDown();
SoundFileHandle *fh = (cd == 1) ? &_musicFile[0] : &_musicFile[1];
fh->inUse = true;
_mutex.unlock();
MusicInputStream *tmp = new MusicInputStream(cd, fh, musicId, loop);
if (tmp->isReady()) {
_mutex.lock();
_music[primary] = tmp;
fh->inUse = false;
_mutex.unlock();
return RD_OK;
} else {
_mutex.lock();
fh->inUse = false;
_mutex.unlock();
delete tmp;
return RDERR_INVALIDFILENAME;
}
}
/**
* @return the time left for the current music, in seconds.
*/
int32 Sound::musicTimeRemaining(void) {
Common::StackLock lock(_mutex);
for (int i = 0; i < MAXMUS; i++) {
if (_music[i] && _music[i]->isFading() <= 0)
return _music[i]->getTimeRemaining();
}
return 0;
}
// ----------------------------------------------------------------------------
// SPEECH
// ----------------------------------------------------------------------------
/**
* Mutes/Unmutes the speech.
* @param mute If mute is false, restore the volume to the last set master
* level. Otherwise the speech is muted (volume 0).
*/
void Sound::muteSpeech(bool mute) {
_speechMuted = mute;
if (_soundHandleSpeech.isActive()) {
uint volume = mute ? 0 : SoundMixer::kMaxChannelVolume;
_vm->_mixer->setChannelVolume(_soundHandleSpeech, volume);
}
}
/**
* Stops the speech dead in its tracks.
*/
void Sound::pauseSpeech(void) {
_speechPaused = true;
_vm->_mixer->pauseHandle(_soundHandleSpeech, true);
}
/**
* Restarts the speech from where it was stopped.
*/
void Sound::unpauseSpeech(void) {
_speechPaused = false;
_vm->_mixer->pauseHandle(_soundHandleSpeech, false);
}
/**
* Stops the speech from playing.
*/
int32 Sound::stopSpeech() {
if (_soundHandleSpeech.isActive()) {
_vm->_mixer->stopHandle(_soundHandleSpeech);
return RD_OK;
}
return RDERR_SPEECHNOTPLAYING;
}
/**
* @return Either RDSE_SAMPLEPLAYING or RDSE_SAMPLEFINISHED
*/
int32 Sound::getSpeechStatus() {
return _soundHandleSpeech.isActive() ? RDSE_SAMPLEPLAYING : RDSE_SAMPLEFINISHED;
}
/**
* Returns either RDSE_QUIET or RDSE_SPEAKING
*/
int32 Sound::amISpeaking() {
if (!_speechMuted && !_speechPaused && _soundHandleSpeech.isActive())
return RDSE_SPEAKING;
return RDSE_QUIET;
}
/**
* This function loads and decompresses a list of speech from a cluster, but
* does not play it. This is used for cutscene voice-overs, presumably to
* avoid having to read from more than one file on the CD during playback.
* @param speechId the text line id used to reference the speech
* @param buf a pointer to the buffer that will be allocated for the sound
*/
uint32 Sound::preFetchCompSpeech(uint32 speechId, uint16 **buf) {
int cd = _vm->_resman->whichCd();
uint32 numSamples;
SoundFileHandle *fh = (cd == 1) ? &_speechFile[0] : &_speechFile[1];
AudioStream *input = getAudioStream(fh, "speech", cd, speechId, &numSamples);
if (!input)
return 0;
*buf = NULL;
// Decompress data into speech buffer.
uint32 bufferSize = 2 * numSamples;
*buf = (uint16 *) malloc(bufferSize);
if (!*buf) {
delete input;
fh->file->close();
return 0;
}
uint32 readSamples = input->readBuffer((int16 *) *buf, numSamples);
fh->file->close();
delete input;
return 2 * readSamples;
}
/**
* This function loads, decompresses and plays a line of speech. An error
* occurs if speech is already playing.
* @param speechId the text line id used to reference the speech
* @param vol volume, 0 (minimum) to 16 (maximum)
* @param pan panning, -16 (full left) to 16 (full right)
*/
int32 Sound::playCompSpeech(uint32 speechId, uint8 vol, int8 pan) {
if (_speechMuted)
return RD_OK;
if (getSpeechStatus() == RDERR_SPEECHPLAYING)
return RDERR_SPEECHPLAYING;
int cd = _vm->_resman->whichCd();
SoundFileHandle *fh = (cd == 1) ? &_speechFile[0] : &_speechFile[1];
AudioStream *input = getAudioStream(fh, "speech", cd, speechId, NULL);
if (!input)
return RDERR_INVALIDID;
// Modify the volume according to the master volume
byte volume = _speechMuted ? 0 : vol * SoundMixer::kMaxChannelVolume / 16;
int8 p = (pan * 127) / 16;
if (isReverseStereo())
p = -p;
// Start the speech playing
_vm->_mixer->playInputStream(SoundMixer::kSpeechAudioDataType, &_soundHandleSpeech, input, -1, volume, p);
return RD_OK;
}
// ----------------------------------------------------------------------------
// SOUND EFFECTS
// ----------------------------------------------------------------------------
/**
* Mutes/Unmutes the sound effects.
* @param mute If mute is false, restore the volume to the last set master
* level. Otherwise the sound effects are muted (volume 0).
*/
void Sound::muteFx(bool mute) {
_fxMuted = mute;
// Now update the volume of any fxs playing
for (int i = 0; i < FXQ_LENGTH; i++) {
if (_fxQueue[i].resource) {
_vm->_mixer->setChannelVolume(_fxQueue[i].handle, mute ? 0 : _fxQueue[i].volume);
}
}
}
/**
* Sets the volume and pan of the sample which is currently playing
* @param id the id of the sample
* @param vol volume
* @param pan panning
*/
int32 Sound::setFxIdVolumePan(int32 i, int vol, int pan) {
if (!_fxQueue[i].resource)
return RDERR_FXNOTOPEN;
if (vol > 16)
vol = 16;
_fxQueue[i].volume = (vol * SoundMixer::kMaxChannelVolume) / 16;
if (pan != 255) {
if (isReverseStereo())
pan = -pan;
_fxQueue[i].pan = (pan * 127) / 16;
}
if (!_fxMuted && _fxQueue[i].handle.isActive()) {
_vm->_mixer->setChannelVolume(_fxQueue[i].handle, _fxQueue[i].volume);
if (pan != -1)
_vm->_mixer->setChannelBalance(_fxQueue[i].handle, _fxQueue[i].pan);
}
return RD_OK;
}
void Sound::pauseFx() {
if (_fxPaused)
return;
for (int i = 0; i < FXQ_LENGTH; i++) {
if (_fxQueue[i].resource)
_vm->_mixer->pauseHandle(_fxQueue[i].handle, true);
}
_fxPaused = true;
}
void Sound::unpauseFx() {
if (!_fxPaused)
return;
for (int i = 0; i < FXQ_LENGTH; i++)
if (_fxQueue[i].resource)
_vm->_mixer->pauseHandle(_fxQueue[i].handle, false);
_fxPaused = false;
}
} // End of namespace Sword2
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