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Updated the BS1 music code to use the new looping feature in our MP3/Ogg decoders

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svn-id: r25943
This commit is contained in:
Max Horn 2007-03-02 23:53:20 +00:00
parent 30a7825b84
commit 4a1039fbc6
2 changed files with 85 additions and 111 deletions
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175
engines/sword1/music.cpp
View File

@ -36,132 +36,130 @@
namespace Sword1 {
WaveAudioStream *makeWaveStream(Common::File *source, uint32 size) {
return new WaveAudioStream(source, size);
class WaveAudioStream : public Audio::AudioStream {
public:
WaveAudioStream(Common::SeekableReadStream *source, bool loop);
virtual ~WaveAudioStream();
virtual int readBuffer(int16 *buffer, const int numSamples);
virtual bool isStereo() const { return _isStereo; }
virtual bool endOfData() const { return (_samplesLeft == 0); }
virtual int getRate() const { return _rate; }
private:
Common::SeekableReadStream *_sourceStream;
uint8 *_sampleBuf;
uint32 _rate;
bool _isStereo;
uint32 _samplesLeft;
uint16 _bitsPerSample;
bool _loop;
void rewind();
};
WaveAudioStream::WaveAudioStream(Common::SeekableReadStream *source, bool loop) {
// TODO: honor the loop flag
_sourceStream = source;
_sampleBuf = (uint8*)malloc(SMP_BUFSIZE);
_samplesLeft = 0;
_isStereo = false;
_bitsPerSample = 16;
_rate = 22050;
_loop = loop;
rewind();
if (_samplesLeft == 0)
_loop = false;
}
WaveAudioStream::WaveAudioStream(Common::File *source, uint32 pSize) {
WaveAudioStream::~WaveAudioStream() {
free(_sampleBuf);
}
void WaveAudioStream::rewind() {
int rate, size;
byte flags;
_sourceStream->seek(0);
_sourceFile = source;
_sampleBuf = (uint8*)malloc(SMP_BUFSIZE);
if (_sourceFile->isOpen() && Audio::loadWAVFromStream(*_sourceFile, size, rate, flags)) {
if (Audio::loadWAVFromStream(*_sourceStream, size, rate, flags)) {
_isStereo = (flags & Audio::Mixer::FLAG_STEREO) != 0;
_rate = rate;
if (pSize && (int)pSize < size)
size = pSize;
assert((uint32)size <= (source->size() - source->pos()));
assert((uint)size <= (_sourceStream->size() - _sourceStream->pos()));
_bitsPerSample = ((flags & Audio::Mixer::FLAG_16BITS) != 0) ? 16 : 8;
_samplesLeft = (size * 8) / _bitsPerSample;
if ((_bitsPerSample != 16) && (_bitsPerSample != 8))
error("WaveAudioStream: unknown wave type");
} else {
_samplesLeft = 0;
_isStereo = false;
_bitsPerSample = 16;
_rate = 22050;
}
}
WaveAudioStream::~WaveAudioStream(void) {
free(_sampleBuf);
}
int WaveAudioStream::readBuffer(int16 *buffer, const int numSamples) {
int samples = MIN((int)_samplesLeft, numSamples);
int retVal = samples;
int retVal = 0;
while (samples > 0) {
int readBytes = MIN(samples * (_bitsPerSample >> 3), SMP_BUFSIZE);
_sourceFile->read(_sampleBuf, readBytes);
if (_bitsPerSample == 16) {
readBytes >>= 1;
samples -= readBytes;
int16 *src = (int16*)_sampleBuf;
while (readBytes--)
*buffer++ = (int16)READ_LE_UINT16(src++);
} else {
samples -= readBytes;
int8 *src = (int8*)_sampleBuf;
while (readBytes--)
*buffer++ = (int16)*src++ << 8;
while (retVal < numSamples && _samplesLeft > 0) {
int samples = MIN((int)_samplesLeft, numSamples);
retVal += samples;
_samplesLeft -= samples;
while (samples > 0) {
int readBytes = MIN(samples * (_bitsPerSample >> 3), SMP_BUFSIZE);
_sourceStream->read(_sampleBuf, readBytes);
if (_bitsPerSample == 16) {
readBytes >>= 1;
samples -= readBytes;
int16 *src = (int16*)_sampleBuf;
while (readBytes--)
*buffer++ = (int16)READ_LE_UINT16(src++);
} else {
samples -= readBytes;
int8 *src = (int8*)_sampleBuf;
while (readBytes--)
*buffer++ = (int16)*src++ << 8;
}
}
if (!_samplesLeft && _loop) {
rewind();
}
}
_samplesLeft -= retVal;
return retVal;
}
bool WaveAudioStream::endOfData(void) const {
if (_samplesLeft == 0)
return true;
else
return false;
}
// This means fading takes 3 seconds.
#define FADE_LENGTH 3
// These functions are only called from Music, so I'm just going to
// assume that if locking is needed it has already been taken care of.
Audio::AudioStream *MusicHandle::createAudioSource(void) {
_file.seek(0);
switch (_musicMode) {
#ifdef USE_MAD
case MusicMp3:
return Audio::makeMP3Stream(&_file, _file.size());
#endif
#ifdef USE_VORBIS
case MusicVorbis:
return Audio::makeVorbisStream(&_file, _file.size());
#endif
case MusicWave:
return makeWaveStream(&_file, 0);
case MusicNone: // shouldn't happen
warning("createAudioSource ran into null create\n");
return NULL;
default:
error("MusicHandle::createAudioSource: called with illegal MusicMode");
}
return NULL; // never reached
}
bool MusicHandle::play(const char *fileBase, bool loop) {
/*
TODO/FIXME: This should be rewritten to make use of the new audio stream factories.
In particular, it should take advantage of the looping capabilities; and it
should avoid reading all the data into memory (by not using the old factory functions).
Essentially, it seems to me as if we could get rid of createAudioSource().
Maybe it could even be change to use AudioStream::openStreamFile.
*/
char fileName[30];
stop();
_musicMode = MusicNone;
#ifdef USE_VORBIS
if (!_file.isOpen()) {
if (!_audioSource) {
sprintf(fileName, "%s.ogg", fileBase);
if (_file.open(fileName))
_musicMode = MusicVorbis;
_audioSource = Audio::makeVorbisStream(&_file, false, 0, 0, loop ? 0 : 1);
}
#endif
#ifdef USE_MAD
if (!_file.isOpen()) {
if (!_audioSource) {
sprintf(fileName, "%s.mp3", fileBase);
if (_file.open(fileName))
_musicMode = MusicMp3;
_audioSource = Audio::makeMP3Stream(&_file, false, 0, 0, loop ? 0 : 1);
}
#endif
if (!_file.isOpen()) {
if (!_audioSource) {
sprintf(fileName, "%s.wav", fileBase);
if (_file.open(fileName))
_musicMode = MusicWave;
else
return false;
_audioSource = new WaveAudioStream(&_file, loop);
}
_audioSource = createAudioSource();
_looping = loop;
if (!_audioSource)
return false;
fadeUp();
return true;
}
@ -217,18 +215,14 @@ int MusicHandle::readBuffer(int16 *buffer, const int numSamples) {
expectedSamples -= samplesReturned;
if ((expectedSamples > 0) && _audioSource->endOfData()) {
debug(2, "Music reached EOF");
if (_looping) {
delete _audioSource; // recreate same source.
_audioSource = createAudioSource();
}
if ((!_looping) || (!_audioSource))
stop();
stop();
}
}
// buffer was filled, now do the fading (if necessary)
int samplePos = 0;
while ((_fading > 0) && (samplePos < totalSamples)) { // fade down
bufStart[samplePos] = (bufStart[samplePos] * --_fading) / _fadeSamples;
--_fading;
bufStart[samplePos] = (bufStart[samplePos] * _fading) / _fadeSamples;
samplePos++;
if (_fading == 0) {
stop();
@ -253,7 +247,6 @@ void MusicHandle::stop() {
if (_file.isOpen())
_file.close();
_fading = 0;
_looping = false;
}
Music::Music(Audio::Mixer *pMixer) {

21
engines/sword1/music.h
View File

@ -41,34 +41,15 @@ enum MusicMode {
MusicVorbis
};
class WaveAudioStream : public Audio::AudioStream {
public:
WaveAudioStream(Common::File *source, uint32 pSize);
virtual ~WaveAudioStream();
virtual int readBuffer(int16 *buffer, const int numSamples);
virtual bool isStereo(void) const { return _isStereo; };
virtual bool endOfData(void) const;
virtual int getRate(void) const { return _rate; };
private:
Common::File *_sourceFile;
uint8 *_sampleBuf;
uint32 _rate;
bool _isStereo;
uint32 _samplesLeft;
uint16 _bitsPerSample;
};
class MusicHandle : public Audio::AudioStream {
private:
Common::File _file;
bool _looping;
int32 _fading;
int32 _fadeSamples;
MusicMode _musicMode;
Audio::AudioStream *_audioSource;
Audio::AudioStream *createAudioSource(void);
public:
MusicHandle() : _looping(false), _fading(0), _audioSource(NULL) {}
MusicHandle() : _fading(0), _audioSource(NULL) {}
virtual int readBuffer(int16 *buffer, const int numSamples);
bool play(const char *filename, bool loop);
void stop();