/*
 * OpenAL Bridge - a simple portable library for OpenAL interface
 * Copyright (c) 2009 Vittorio Giovara <vittorio.giovara@gmail.com>
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU Lesser General Public License as published by
 * the Free Software Foundation; version 2 of the License
 *
 * 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 Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser 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
 */

#include "loaders.h"

#ifdef __CPLUSPLUS
extern "C" {
#endif 
    
    int load_wavpcm (const char *filename, ALenum *format, char ** data, ALsizei *bitsize, ALsizei *freq) {
        WAV_header_t WAVHeader;
        FILE *wavfile;
        int32_t t;
        uint32_t n = 0;
        
        wavfile = Fopen(filename, "rb");
        
        fread(&WAVHeader.ChunkID, sizeof(uint32_t), 1, wavfile);
        fread(&WAVHeader.ChunkSize, sizeof(uint32_t), 1, wavfile);
        fread(&WAVHeader.Format, sizeof(uint32_t), 1, wavfile);
        
#ifdef DEBUG
        fprintf(stderr, "ChunkID: %X\n", invert_endianness(WAVHeader.ChunkID));
        fprintf(stderr, "ChunkSize: %d\n", WAVHeader.ChunkSize);
        fprintf(stderr, "Format: %X\n", invert_endianness(WAVHeader.Format));
#endif
        
        fread(&WAVHeader.Subchunk1ID, sizeof(uint32_t), 1, wavfile);
        fread(&WAVHeader.Subchunk1Size, sizeof(uint32_t), 1, wavfile);
        fread(&WAVHeader.AudioFormat, sizeof(uint16_t), 1, wavfile);
        fread(&WAVHeader.NumChannels, sizeof(uint16_t), 1, wavfile);
        fread(&WAVHeader.SampleRate, sizeof(uint32_t), 1, wavfile);
        fread(&WAVHeader.ByteRate, sizeof(uint32_t), 1, wavfile);
        fread(&WAVHeader.BlockAlign, sizeof(uint16_t), 1, wavfile);
        fread(&WAVHeader.BitsPerSample, sizeof(uint16_t), 1, wavfile);
        
#ifdef DEBUG
        fprintf(stderr, "Subchunk1ID: %X\n", invert_endianness(WAVHeader.Subchunk1ID));
        fprintf(stderr, "Subchunk1Size: %d\n", WAVHeader.Subchunk1Size);
        fprintf(stderr, "AudioFormat: %d\n", WAVHeader.AudioFormat);
        fprintf(stderr, "NumChannels: %d\n", WAVHeader.NumChannels);
        fprintf(stderr, "SampleRate: %d\n", WAVHeader.SampleRate);
        fprintf(stderr, "ByteRate: %d\n", WAVHeader.ByteRate);
        fprintf(stderr, "BlockAlign: %d\n", WAVHeader.BlockAlign);
        fprintf(stderr, "BitsPerSample: %d\n", WAVHeader.BitsPerSample);
#endif
        
        do { /*remove useless header chunks (plenty room for improvements)*/
            t = fread(&WAVHeader.Subchunk2ID, sizeof(uint32_t), 1, wavfile);
            if (invert_endianness(WAVHeader.Subchunk2ID) == 0x64617461)
                break;
            if (t <= 0) { /*eof*/
                fprintf(stderr, "ERROR: wrong WAV header\n");
                return AL_FALSE;
            }
        } while (1);
        fread(&WAVHeader.Subchunk2Size, sizeof(uint32_t), 1, wavfile);
        
#ifdef DEBUG
        fprintf(stderr, "Subchunk2ID: %X\n", invert_endianness(WAVHeader.Subchunk2ID));
        fprintf(stderr, "Subchunk2Size: %d\n", WAVHeader.Subchunk2Size);
#endif
        
        *data = (char*) Malloc (sizeof(char) * WAVHeader.Subchunk2Size);
        
        /*this could be improved*/
        do {
            n += fread(&((*data)[n]), sizeof(uint8_t), 1, wavfile);
        } while (n < WAVHeader.Subchunk2Size);
        
        fclose(wavfile);	
        
#ifdef DEBUG
        fprintf(stderr, "Last two bytes of data: %X%X\n", (*data)[n-2], (*data)[n-1]);
#endif
        
        /*remaining parameters*/
        /*Valid formats are AL_FORMAT_MONO8, AL_FORMAT_MONO16, AL_FORMAT_STEREO8, and AL_FORMAT_STEREO16*/
        if (WAVHeader.NumChannels == 1) {
            if (WAVHeader.BitsPerSample == 8)
                *format = AL_FORMAT_MONO8;
            else {
                if (WAVHeader.BitsPerSample == 16)
                    *format = AL_FORMAT_MONO16;
                else {
                    fprintf(stderr, "ERROR: wrong WAV header - bitsample value\n");
                    return AL_FALSE;
                }
            } 
        } else {
            if (WAVHeader.NumChannels == 2) {
                if (WAVHeader.BitsPerSample == 8)
                    *format = AL_FORMAT_STEREO8;
                else {
                    if (WAVHeader.BitsPerSample == 16)
                        *format = AL_FORMAT_STEREO16;
                    else {
                        fprintf(stderr, "ERROR: wrong WAV header - bitsample value\n");
                        return AL_FALSE;
                    }				
                }
            } else {
                fprintf(stderr, "ERROR: wrong WAV header - format value\n");
                return AL_FALSE;
            }
        }
        
        *bitsize = WAVHeader.Subchunk2Size;
        *freq = WAVHeader.SampleRate;
        return AL_TRUE;
    }
    

    int load_oggvorbis (const char *filename, ALenum *format, char **data, ALsizei *bitsize, ALsizei *freq) {
        /*implementation inspired from http://www.devmaster.net/forums/showthread.php?t=1153 */
        OggVorbis_File  oggStream;	/*stream handle*/
        vorbis_info	*vorbisInfo;	/*some formatting data*/
        int64_t		pcm_length;	/*length of the decoded data*/
        int             section, result, size = 0;
#ifdef DEBUG
        int i;
        vorbis_comment	*vorbisComment;	/*other less useful data*/
#endif

	result = ov_fopen((char*) filename, &oggStream);
	if (result < 0) {
		fprintf (stderr, "ERROR: ov_open_callbacks failed with %X", result);
                ov_clear(&oggStream);
		return -1;
	}

        vorbisInfo = ov_info(&oggStream, -1);
        pcm_length = ov_pcm_total(&oggStream, -1) << vorbisInfo->channels;	
        
#ifdef DEBUG
        vorbisComment = ov_comment(&oggStream, -1);
        fprintf(stderr, "Version: %d\n", vorbisInfo->version);
        fprintf(stderr, "Channels: %d\n", vorbisInfo->channels);
        fprintf(stderr, "Rate (Hz): %ld\n", vorbisInfo->rate);
        fprintf(stderr, "Bitrate Upper: %ld\n", vorbisInfo->bitrate_upper);
        fprintf(stderr, "Bitrate Nominal: %ld\n", vorbisInfo->bitrate_nominal);
        fprintf(stderr, "Bitrate Lower: %ld\n", vorbisInfo->bitrate_lower);
        fprintf(stderr, "Bitrate Windows: %ld\n", vorbisInfo->bitrate_window);
        fprintf(stderr, "Vendor: %s\n", vorbisComment->vendor);
        fprintf(stderr, "PCM data size: %lld\n", pcm_length);
        fprintf(stderr, "# comment: %d\n", vorbisComment->comments);
        for (i = 0; i < vorbisComment->comments; i++)
            fprintf(stderr, "\tComment %d: %s\n", i, vorbisComment->user_comments[i]);
#endif
        
        /*allocates enough room for the decoded data*/
        *data = (char*) Malloc (sizeof(char) * pcm_length);
        
        /*there *should* not be ogg at 8 bits*/
        if (vorbisInfo->channels == 1)
            *format = AL_FORMAT_MONO16;
        else {
            if (vorbisInfo->channels == 2)
                *format = AL_FORMAT_STEREO16;
            else {
                fprintf(stderr, "ERROR: wrong OGG header - channel value (%d)\n", vorbisInfo->channels);
                ov_clear(&oggStream);
                return AL_FALSE;
            }
        }
        
        while (size < pcm_length) {
            /*ov_read decodes the ogg stream and storse the pcm in data*/
            result = ov_read (&oggStream, *data + size, pcm_length - size, 0, 2, 1, &section);
            if (result > 0) {
                size += result;
            } else {
                if (result == 0)
                    break;
                else { 
                    fprintf(stderr, "ERROR: end of file from OGG stream\n");
                    ov_clear(&oggStream);
                    return AL_FALSE;
                }
            }
        }
        
        /*set the last fields*/
        *bitsize = size;
        *freq    = vorbisInfo->rate;
        
	/*cleaning time*/
        ov_clear(&oggStream);

        return AL_TRUE;
    }
    
#ifdef __CPLUSPLUS
}
#endif