/*

 * 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 */

        FILE		*oggFile;		/*ogg handle*/

        OggVorbis_File  oggStream;		/*stream handle*/

        vorbis_info	*vorbisInfo;	/*some formatting data*/

        int64_t		pcm_length;		/*length of the decoded data*/

#ifdef DEBUG

        int i;

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

#endif

        oggFile = Fopen(filename, "rb");

        result = ov_open(oggFile, &oggStream, NULL, 0);	/*TODO: check returning value of result*/

        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);

                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");

                    return AL_FALSE;

                }

            }

        }

        /*records the last fields*/

        *bitsize = size;

        *freq    = vorbisInfo->rate;

        ov_clear (&oggStream);

        return AL_TRUE;

    }

#ifdef __CPLUSPLUS

}

#endif