/* ScummVM - Scumm Interpreter
 * 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.
 *
 * $Header$
 *
 */

#include "stdafx.h"
#include "sword1/music.h"
#include "sound/mixer.h"
#include "common/util.h"
#include "common/file.h"
#include "sound/mp3.h"
#include "sound/vorbis.h"
#include "sound/wave.h"

#define SMP_BUFSIZE 8192

namespace Sword1 {

WaveAudioStream *makeWaveStream(File *source, uint32 size) {
	return new WaveAudioStream(source, size);
}

WaveAudioStream::WaveAudioStream(File *source, uint32 pSize) {
	uint32 size;
	uint8 wavHeader[WAVEHEADERSIZE];

	_sourceFile = source;
	_sampleBuf = (uint8*)malloc(SMP_BUFSIZE);
	_sourceFile->incRef();
	if (_sourceFile->isOpen()) {
		// TODO: use loadWAVFromStream to load the WAVE data!
		/*
		int rate, size;
		bye flags;
		const uint32 initialPos = _sourceFile->pos();
		isValidWAV = loadWAVFromStream(*_sourceFile, size, rate, flags);
		*/


		_sourceFile->read(wavHeader, WAVEHEADERSIZE);
		_isStereo = (READ_LE_UINT16(wavHeader + 0x16) == 2);
		_rate = READ_LE_UINT16(wavHeader + 0x18);
		size = ((pSize) ? pSize : READ_LE_UINT32(wavHeader + 0x28));
		assert(size <= (source->size() - source->pos()));
		_bitsPerSample = READ_LE_UINT16(wavHeader + 0x22);
		_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);
	_sourceFile->decRef();
}

int WaveAudioStream::readBuffer(int16 *buffer, const int numSamples) {
	int samples = ((int)_samplesLeft < numSamples) ? (int)_samplesLeft : numSamples;
	int retVal = samples;

	while (samples > 0) {
		int readBytes = (samples * (_bitsPerSample >> 3) > SMP_BUFSIZE) ? SMP_BUFSIZE : samples * (_bitsPerSample >> 3);
		_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;
		}
	}
	_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.

AudioStream *MusicHandle::createAudioSource(void) {
	_file.seek(0);
	switch (_musicMode) {
#ifdef USE_MAD
		case MusicMp3:
			return makeMP3Stream(&_file, _file.size());			
#endif
#ifdef USE_VORBIS
		case MusicVorbis:
			return 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) {
	char fileName[30];
	stop();
	_musicMode = MusicNone;
#ifdef USE_MAD
	sprintf(fileName, "%s.mp3", fileBase);
	if (_file.open(fileName))
		_musicMode = MusicMp3;
#endif
#ifdef USE_VORBIS
	if (!_file.isOpen()) { // mp3 doesn't exist (or not compiled with MAD support)
		sprintf(fileName, "%s.ogg", fileBase);
		if (_file.open(fileName))
			_musicMode = MusicVorbis;
	}
#endif
	if (!_file.isOpen()) {
		sprintf(fileName, "%s.wav", fileBase);
		if (_file.open(fileName))
            _musicMode = MusicWave;
		else {
			warning("Music file %s could not be opened", fileName);
			return false;
		}
	}
	_audioSource = createAudioSource();
	_looping = loop;
	fadeUp();
	return true;
}

void MusicHandle::fadeDown() {
	if (streaming()) {
		if (_fading < 0)
			_fading = -_fading;
		else if (_fading == 0)
			_fading = FADE_LENGTH * getRate();
		_fadeSamples = FADE_LENGTH * getRate();
	}
}

void MusicHandle::fadeUp() {
	if (streaming()) {
		if (_fading > 0)
			_fading = -_fading;
		else if (_fading == 0)
			_fading = -1;
		_fadeSamples = FADE_LENGTH * getRate();
	}
}

bool MusicHandle::endOfData() const {
	return !streaming();
}

// is we don't have an audiosource, return some dummy values.
// shouldn't happen anyways.
bool MusicHandle::streaming(void) const {
	return (_audioSource) ? (!_audioSource->endOfStream()) : false;
}

bool MusicHandle::isStereo(void) const {
	return (_audioSource) ? _audioSource->isStereo() : false;
}

int MusicHandle::getRate(void) const {
	return (_audioSource) ? _audioSource->getRate() : 11025;
}

int MusicHandle::readBuffer(int16 *buffer, const int numSamples) {
	int totalSamples = 0;
	int16 *bufStart = buffer;
	if (!_audioSource)
		return 0;
	int expectedSamples = numSamples;
	while ((expectedSamples > 0) && _audioSource) { // _audioSource becomes NULL if we reach EOF and aren't looping
		int samplesReturned = _audioSource->readBuffer(buffer, expectedSamples);
		buffer += samplesReturned;
		totalSamples += samplesReturned;
        expectedSamples -= samplesReturned;
		if ((expectedSamples > 0) && _audioSource->endOfData()) {
			debug(2, "Music reached EOF");
			_audioSource->endOfData();
			if (_looping) { 
                delete _audioSource; // recreate same source.
				_audioSource = createAudioSource();
			}
			if ((!_looping) || (!_audioSource))
				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;
		samplePos++;
		if (_fading == 0) {
			stop();
			// clear the rest of the buffer
            memset(bufStart + samplePos, 0, (totalSamples - samplePos) * 2);
			return samplePos;
		}
	}
	while ((_fading < 0) && (samplePos < totalSamples)) { // fade up
        bufStart[samplePos] = -(bufStart[samplePos] * --_fading) / _fadeSamples;
		if (_fading <= -_fadeSamples)
			_fading = 0;
	}
	return totalSamples;
}

void MusicHandle::stop() {
	if (_audioSource) {
		delete _audioSource;
		_audioSource = NULL;
	}
	if (_file.isOpen())
		_file.close();
	_fading = 0;
	_looping = false;
}

Music::Music(OSystem *system, SoundMixer *pMixer) {
	_system = system;
	_mixer = pMixer;
	_sampleRate = pMixer->getOutputRate();
	_mutex = _system->createMutex();
	_converter[0] = NULL;
	_converter[1] = NULL;
	_volumeL = _volumeR = 192;
	_mixer->setupPremix(this);
}

Music::~Music() {
	_mixer->setupPremix(0);
	delete _converter[0];
	delete _converter[1];
	if (_mutex)
		_system->deleteMutex(_mutex);
}

void Music::mixer(int16 *buf, uint32 len) {
	Common::StackLock lock(_mutex);
	memset(buf, 0, 2 * len * sizeof(int16));
	for (int i = 0; i < ARRAYSIZE(_handles); i++)
		if (_handles[i].streaming() && _converter[i])
			_converter[i]->flow(_handles[i], buf, len, _volumeL, _volumeR);
}

void Music::setVolume(uint8 volL, uint8 volR) {
	_volumeL = (st_volume_t)volL;
	_volumeR = (st_volume_t)volR;
}

void Music::giveVolume(uint8 *volL, uint8 *volR) {
	*volL = (uint8)_volumeL;
	*volR = (uint8)_volumeR;
}

void Music::startMusic(int32 tuneId, int32 loopFlag) {
	Common::StackLock lock(_mutex);
	if (strlen(_tuneList[tuneId]) > 0) {
		int newStream = 0;
		if (_handles[0].streaming() && _handles[1].streaming()) {
			int streamToStop;
			// Both streams playing - one must be forced to stop.
			if (!_handles[0].fading() && !_handles[1].fading()) {
				// None of them are fading. Shouldn't happen,
				// so it doesn't matter which one we pick.
				streamToStop = 0;
			} else if (_handles[0].fading() && !_handles[1].fading()) {
				// Stream 0 is fading, so pick that one.
				streamToStop = 0;
			} else if (!_handles[0].fading() && _handles[1].fading()) {
				// Stream 1 is fading, so pick that one.
				streamToStop = 1;
			} else {
				// Both streams are fading. Pick the one that
				// is closest to silent.
				if (ABS(_handles[0].fading()) < ABS(_handles[1].fading()))
					streamToStop = 0;
				else
					streamToStop = 1;
			}
			_handles[streamToStop].stop();
		}
		if (_handles[0].streaming()) {
			_handles[0].fadeDown();
			newStream = 1;
		} else if (_handles[1].streaming()) {
			_handles[1].fadeDown();
			newStream = 0;
		}
		if (_handles[newStream].play(_tuneList[tuneId], loopFlag != 0)) {
			delete _converter[newStream];
			_converter[newStream] = makeRateConverter(_handles[newStream].getRate(), _mixer->getOutputRate(), _handles[newStream].isStereo(), false);
		}
	} else {
		if (_handles[0].streaming())
			_handles[0].fadeDown();
		if (_handles[1].streaming())
			_handles[1].fadeDown();
	}
}

void Music::fadeDown() {
	Common::StackLock lock(_mutex);
	for (int i = 0; i < ARRAYSIZE(_handles); i++)
		if (_handles[i].streaming())
			_handles[i].fadeDown();
}

} // End of namespace Sword1