/* ScummVM - Scumm Interpreter
 * Copyright (C) 2003-2006 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
 *
 * $URL$
 * $Id$
 *
 */

#include "common/stdafx.h"
#include "common/endian.h"
#include "common/file.h"
#include "common/util.h"
#include "common/system.h"

#include "sword1/music.h"
#include "sound/mixer.h"
#include "sound/mp3.h"
#include "sound/vorbis.h"
#include "sound/wave.h"

#define SMP_BUFSIZE 8192

namespace Sword1 {

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

WaveAudioStream::WaveAudioStream(Common::File *source, uint32 pSize) {
	int rate, size;
	byte flags;

	_sourceFile = source;
	_sampleBuf = (uint8*)malloc(SMP_BUFSIZE);
	if (_sourceFile->isOpen() && Audio::loadWAVFromStream(*_sourceFile, 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()));
		_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;

	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;
		}
	}
	_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()) {
		sprintf(fileName, "%s.ogg", fileBase);
		if (_file.open(fileName))
			_musicMode = MusicVorbis;
	}
#endif
#ifdef USE_MAD
	if (!_file.isOpen()) {
		sprintf(fileName, "%s.mp3", fileBase);
		if (_file.open(fileName))
			_musicMode = MusicMp3;
	}
#endif
	if (!_file.isOpen()) {
		sprintf(fileName, "%s.wav", fileBase);
		if (_file.open(fileName))
			_musicMode = MusicWave;
		else
			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");
			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(Audio::Mixer *pMixer) {
	_mixer = pMixer;
	_sampleRate = pMixer->getOutputRate();
	_converter[0] = NULL;
	_converter[1] = NULL;
	_volumeL = _volumeR = 192;
	_mixer->playInputStream(Audio::Mixer::kPlainSoundType, &_soundHandle, this, -1, Audio::Mixer::kMaxChannelVolume, 0, false, true);
}

Music::~Music() {
	_mixer->stopHandle(_soundHandle);
	delete _converter[0];
	delete _converter[1];
}

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 = (Audio::st_volume_t)volL;
	_volumeR = (Audio::st_volume_t)volR;
}

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

void Music::startMusic(int32 tuneId, int32 loopFlag) {
	if (strlen(_tuneList[tuneId]) > 0) {
		int newStream = 0;
		_mutex.lock();
		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;
		}
		delete _converter[newStream];
		_converter[newStream] = NULL;
		_mutex.unlock();

		/* The handle will load the music file now. It can take a while, so unlock
		   the mutex before, to have the soundthread playing normally.
		   As the corresponding _converter is NULL, the handle will be ignored by the playing thread */
		if (_handles[newStream].play(_tuneList[tuneId], loopFlag != 0)) {
			_mutex.lock();
			_converter[newStream] = Audio::makeRateConverter(_handles[newStream].getRate(), _mixer->getOutputRate(), _handles[newStream].isStereo(), false);
			_mutex.unlock();
		} else {
			if (tuneId != 81) // file 81 was apparently removed from BS.
				warning("Can't find music file %s", _tuneList[tuneId]);
		}
	} else {
		_mutex.lock();
		if (_handles[0].streaming())
			_handles[0].fadeDown();
		if (_handles[1].streaming())
			_handles[1].fadeDown();
		_mutex.unlock();
	}
}

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