/**********************************************************************
** Copyright (C) 2000 Trolltech AS.  All rights reserved.
**
** This file is part of Qtopia Environment.
**
** This file may be distributed and/or modified under the terms of the
** GNU General Public License version 2 as published by the Free Software
** Foundation and appearing in the file LICENSE.GPL included in the
** packaging of this file.
**
** This file is provided AS IS with NO WARRANTY OF ANY KIND, INCLUDING THE
** WARRANTY OF DESIGN, MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE.
**
** See http://www.trolltech.com/gpl/ for GPL licensing information.
**
** Contact info@trolltech.com if any conditions of this licensing are
** not clear to you.
**
**********************************************************************/
#include <stdio.h>
#include <stdarg.h>
#include <stdlib.h>
#include <errno.h>
#include <unistd.h>
#include <qfile.h>
#include "wavplugin.h"


//#define debugMsg(a)	    qDebug(a)
#define debugMsg(a)


struct RiffChunk {
    char id[4];
    Q_UINT32 size;
    char data[4];
};


struct ChunkData {
    Q_INT16 formatTag;
    Q_INT16 channels;
    Q_INT32 samplesPerSec;
    Q_INT32 avgBytesPerSec;
    Q_INT16 blockAlign;
    Q_INT16 wBitsPerSample;
};


const int sound_buffer_size = 4096;


class WavPluginData {
public:
    QFile *input;

    int wavedata_remaining;
    ChunkData chunkdata;
    RiffChunk chunk;
    uchar data[sound_buffer_size+32]; // +32 to handle badly aligned input data
    int out,max;
    int samples_due;
    int samples;

    WavPluginData() {
	max = out = sound_buffer_size;
	wavedata_remaining = 0;
	samples_due = 0;
	samples = -1;
    }

    // expands out samples to the frequency of 44kHz
    bool add( short *output, long count, long& done, bool stereo )
    {
	done = 0;

	if ( input == 0 ) {
	    qDebug("no input");
	    return FALSE;
	}

        while ( count ) {
            int l,r;
            if ( getSample(l, r) == FALSE ) {
		qDebug("didn't get sample");
		return FALSE;
	    }
            samples_due += 44100;
            while ( count && (samples_due > chunkdata.samplesPerSec) ) {
		*output++ = l;
		if ( stereo )
		    *output++ = r;
                samples_due -= chunkdata.samplesPerSec;
                count--;
		done++;
            }
        }

        return TRUE;
    }

    bool initialise() {
	if ( input == 0 )
	    return FALSE;

	wavedata_remaining = -1;

	while ( wavedata_remaining == -1 ) {
	    // Keep reading chunks...
	    const int n = sizeof(chunk) - sizeof(chunk.data);
	    int t = input->readBlock( (char*)&chunk, n );
	    if ( t != n ) {
		if ( t == -1 )
		    return FALSE;
		return TRUE;
	    }
	    if ( qstrncmp(chunk.id,"data",4) == 0 ) {
		samples = wavedata_remaining = chunk.size;
	    } else if ( qstrncmp(chunk.id,"RIFF",4) == 0 ) {
		char d[4];
		if ( input->readBlock(d,4) != 4 ) {
		    return FALSE;
		}
		if ( qstrncmp(d,"WAVE",4) != 0 ) {
		    // skip
		    if ( chunk.size > 1000000000 || !input->at(input->at()+chunk.size-4) ) {
			return FALSE;
		    }
		}
	    } else if ( qstrncmp(chunk.id,"fmt ",4) == 0 ) {
		if ( input->readBlock((char*)&chunkdata,sizeof(chunkdata)) != sizeof(chunkdata) ) {
		    return FALSE;
		}
#define WAVE_FORMAT_PCM 1
		if ( chunkdata.formatTag != WAVE_FORMAT_PCM ) {
		    qDebug("WAV file: UNSUPPORTED FORMAT %d",chunkdata.formatTag);
		    return FALSE;
		}
	    } else {
		// ignored chunk
		if ( chunk.size > 1000000000 || !input->at(input->at()+chunk.size) ) {
		    return FALSE;
		}
	    }
	} // while

	return TRUE;
    }


    // gets a sample from the file
    bool getSample(int& l, int& r)
    {
	l = r = 0;

	if ( input == 0 )
	    return FALSE;

	if ( (wavedata_remaining < 0) || !max )
	    return FALSE;

	if ( out >= max ) {
	    max = input->readBlock( (char*)data, (uint)QMIN(sound_buffer_size,wavedata_remaining) );

	    wavedata_remaining -= max;

	    out = 0;
	    if ( max <= 0 ) {
		max = 0;
		return TRUE;
	    }
	}
	if ( chunkdata.wBitsPerSample == 8 ) {
	    l = (data[out++] - 128) * 128;
	} else {
	    l = ((short*)data)[out/2];
	    out += 2;
	}
	if ( chunkdata.channels == 1 ) {
	    r = l;
	} else {
	    if ( chunkdata.wBitsPerSample == 8 ) {
		r = (data[out++] - 128) * 128;
	    } else {
		r = ((short*)data)[out/2];
		out += 2;
	    }
	}
	return TRUE;
    } // getSample

};


WavPlugin::WavPlugin() {
    d = new WavPluginData;
    d->input = 0;
}


WavPlugin::~WavPlugin() {
    close();
    delete d;
}


bool WavPlugin::isFileSupported( const QString& path ) {
    debugMsg( "WavPlugin::isFileSupported" );

    char *ext = strrchr( path.latin1(), '.' );

    // Test file extension
    if ( ext ) {
	if ( strncasecmp(ext, ".raw", 4) == 0 )
	    return TRUE;
	if ( strncasecmp(ext, ".wav", 4) == 0 )
	    return TRUE;
	if ( strncasecmp(ext, ".wave", 4) == 0 )
	    return TRUE;
    }

    return FALSE;
}


bool WavPlugin::open( const QString& path ) {
    debugMsg( "WavPlugin::open" );

    d->max = d->out = sound_buffer_size;
    d->wavedata_remaining = 0;
    d->samples_due = 0;

    d->input = new QFile( path );
    if ( d->input->open(IO_ReadOnly) == FALSE ) {
	qDebug("couldn't open file");
	delete d->input;
	d->input = 0;
	return FALSE;
    }

    d->initialise();

    return TRUE;
}


bool WavPlugin::close() {
    debugMsg( "WavPlugin::close" );

    d->input->close();
    delete d->input;
    d->input = 0;
    return TRUE;
}


bool WavPlugin::isOpen() {
    debugMsg( "WavPlugin::isOpen" );
    return ( d->input != 0 );
}


int WavPlugin::audioStreams() {
    debugMsg( "WavPlugin::audioStreams" );
    return 1;
}


int WavPlugin::audioChannels( int ) {
    debugMsg( "WavPlugin::audioChannels" );
    return 2; // ### Always scale audio to stereo samples
}


int WavPlugin::audioFrequency( int ) {
    debugMsg( "WavPlugin::audioFrequency" );
    return 44100; // ### Always scale to frequency of 44100
}


int WavPlugin::audioSamples( int ) {
    debugMsg( "WavPlugin::audioSamples" );
    return d->samples * 2 / d->chunkdata.channels; // ### Scaled samples will be made stereo,
				// Therefore if source is mono we will double the number of samples
}


bool WavPlugin::audioSetSample( long, int ) {
    debugMsg( "WavPlugin::audioSetSample" );
    return FALSE;
}


long WavPlugin::audioGetSample( int ) {
    debugMsg( "WavPlugin::audioGetSample" );
    return 0;
}

/*
bool WavPlugin::audioReadSamples( short *, int, long, int ) {
    debugMsg( "WavPlugin::audioReadSamples" );
    return FALSE;
}


bool WavPlugin::audioReReadSamples( short *, int, long, int ) {
    debugMsg( "WavPlugin::audioReReadSamples" );
    return FALSE;
}


bool WavPlugin::audioReadMonoSamples( short *output, long samples, long& samplesMade, int ) {
    debugMsg( "WavPlugin::audioReadMonoSamples" );
    return !d->add( output, samples, samplesMade, FALSE );
}


bool WavPlugin::audioReadStereoSamples( short *output, long samples, long& samplesMade, int ) {
    debugMsg( "WavPlugin::audioReadStereoSamples" );
    return !d->add( output, samples, samplesMade, TRUE );
}
*/

bool WavPlugin::audioReadSamples( short *output, int channels, long samples, long& samplesMade, int ) {
    debugMsg( "WavPlugin::audioReadSamples" );
    return !d->add( output, samples, samplesMade, channels != 1 );
}

double WavPlugin::getTime() {
    debugMsg( "WavPlugin::getTime" );
    return 0.0;
}