/*

 * 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;

        uint8_t sub0, sub1, sub2, sub3;

        wavfile = Fopen(filename, "rb");

        fread(&WAVHeader.ChunkID, sizeof(uint32_t), 1, wavfile);                /*RIFF*/

        fread(&WAVHeader.ChunkSize, sizeof(uint32_t), 1, wavfile);

        fread(&WAVHeader.Format, sizeof(uint32_t), 1, wavfile);                 /*WAVE*/

#ifdef DEBUG

        fprintf(stderr, "ChunkID: %X\n", ENDIAN_BIG_32(WAVHeader.ChunkID));

        fprintf(stderr, "ChunkSize: %d\n", ENDIAN_LITTLE_32(WAVHeader.ChunkSize));

        fprintf(stderr, "Format: %X\n", ENDIAN_BIG_32(WAVHeader.Format));

#endif

        fread(&WAVHeader.Subchunk1ID, sizeof(uint32_t), 1, wavfile);            /*fmt */

        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", ENDIAN_BIG_32(WAVHeader.Subchunk1ID));

        fprintf(stderr, "Subchunk1Size: %d\n", ENDIAN_LITTLE_32(WAVHeader.Subchunk1Size));

        fprintf(stderr, "AudioFormat: %d\n", ENDIAN_LITTLE_16(WAVHeader.AudioFormat));

        fprintf(stderr, "NumChannels: %d\n", ENDIAN_LITTLE_16(WAVHeader.NumChannels));

        fprintf(stderr, "SampleRate: %d\n", ENDIAN_LITTLE_32(WAVHeader.SampleRate));

        fprintf(stderr, "ByteRate: %d\n", ENDIAN_LITTLE_32(WAVHeader.ByteRate));

        fprintf(stderr, "BlockAlign: %d\n", ENDIAN_LITTLE_16(WAVHeader.BlockAlign));

        fprintf(stderr, "BitsPerSample: %d\n", ENDIAN_LITTLE_16(WAVHeader.BitsPerSample));

#endif

        /*remove useless header chunks by looking for the WAV_HEADER_SUBCHUNK2ID integer */

        do {

                t = fread(&sub0, sizeof(uint8_t), 1, wavfile);

                if(sub0 == 0x64) {

                        t = fread(&sub1, sizeof(uint8_t), 1, wavfile);

                        if(sub1 == 0x61) {

                                t = fread(&sub2, sizeof(uint8_t), 1, wavfile);

                                if(sub2 == 0x74) {

                                        t = fread(&sub3, sizeof(uint8_t), 1, wavfile);

                                        if(sub3 == 0x61) {

                                                WAVHeader.Subchunk2ID = WAV_HEADER_SUBCHUNK2ID;

                                                break;                                                

                                        } 

                                }       

                        }

                }

                if (t <= 0) { 

                        /*eof*/

                        errno = EILSEQ;

                        err_ret("(%s) ERROR - wrong WAV header", prog);

                        return AL_FALSE;

                }

        } while (1);

        fread(&WAVHeader.Subchunk2Size, sizeof(uint32_t), 1, wavfile);

#ifdef DEBUG

        fprintf(stderr, "Subchunk2ID: %X\n", ENDIAN_LITTLE_32(WAVHeader.Subchunk2ID));

        fprintf(stderr, "Subchunk2Size: %d\n", ENDIAN_LITTLE_32(WAVHeader.Subchunk2Size));

#endif

        *data = (char*) Malloc (sizeof(char) * ENDIAN_LITTLE_32(WAVHeader.Subchunk2Size));

        /*read the actual sound data*/

        do {

                n += fread(&((*data)[n]), sizeof(uint8_t), 4, wavfile);

        } while (n < ENDIAN_LITTLE_32(WAVHeader.Subchunk2Size));

        fclose(wavfile);	

#ifdef DEBUG

#endif

        /*set parameters for OpenAL*/

        /*Valid formats are AL_FORMAT_MONO8, AL_FORMAT_MONO16, AL_FORMAT_STEREO8, and AL_FORMAT_STEREO16*/

        if (ENDIAN_LITTLE_16(WAVHeader.NumChannels) == 1) {

            if (ENDIAN_LITTLE_16(WAVHeader.BitsPerSample) == 8)

                *format = AL_FORMAT_MONO8;

            else {

                if (ENDIAN_LITTLE_16(WAVHeader.BitsPerSample) == 16)

                    *format = AL_FORMAT_MONO16;

                else {

                }

            } 

        } else {

            if (ENDIAN_LITTLE_16(WAVHeader.NumChannels) == 2) {

                if (ENDIAN_LITTLE_16(WAVHeader.BitsPerSample) == 8)

                    *format = AL_FORMAT_STEREO8;

                else {

                    if (ENDIAN_LITTLE_16(WAVHeader.BitsPerSample) == 16)

                        *format = AL_FORMAT_STEREO16;

                    else {

                        return AL_FALSE;

                    }				

                }

            } else {

            }

        }

        *bitsize = ENDIAN_LITTLE_32(WAVHeader.Subchunk2Size);

        *freq    = ENDIAN_LITTLE_32(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, endianness;

#ifdef DEBUG

            int i;

            vorbis_comment *vorbisComment; /*other less useful data*/

#endif

	result = ov_fopen((char*) filename, &oggStream);

	if (result < 0) {

                ov_clear(&oggStream);

		return AL_FALSE;

	}

        /*load OGG header and determine the decoded data size*/

        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 {

                            errno = EILSEQ;

                            err_ret("(%s) ERROR - wrong OGG header [channel %d]", prog, vorbisInfo->channels);

                            ov_clear(&oggStream);

                            return AL_FALSE;

                    }

            }

            size = 0;

#ifdef __LITTLE_ENDIAN__

            endianness = 0;

#elif __BIG_ENDIAN__

            endianness = 1;

#endif

        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, endianness, 2, 1, &section);

            if (result > 0) {

                size += result;

            } else {

                if (result == 0)

                    break;

                else { 

                    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