* Copyright (c) 2009 Vittorio Giovara <vittorio.giovara@gmail.com>,

 *                    Mario Liebisch <mario.liebisch+hw@googlemail.com>

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);

        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));

        /*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);

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

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

        *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

        err_msg("(%s) INFO - WAV data loaded", prog);

#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 {

                                err_ret("(%s) ERROR - wrong WAV header [bitsample value]", prog);

                } 

        } else {

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

                                *format = AL_FORMAT_STEREO8;

                                        *format = AL_FORMAT_STEREO16;

                                }				

                        }

                        errno = EILSEQ;

                        err_ret("(%s) ERROR - wrong WAV header [format value]", prog); 

                        return AL_FALSE;

        *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 */

        /*stream handle*/

        OggVorbis_File  oggStream; 

        /*some formatting data*/

        vorbis_info *vorbisInfo; 

        /*length of the decoded data*/

        int64_t pcm_length;

        /*other vars*/

        int section, result, size, endianness;

#ifdef DEBUG

        int i;

        /*other less useful data*/

        vorbis_comment *vorbisComment;

#endif

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

        if (result < 0) {

                errno = EINVAL;

                err_ret("(%s) ERROR - ov_fopen() failed with %X", prog, result);

                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;	

        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]);

        /*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);

        }

        size = 0;

#ifdef __LITTLE_ENDIAN__

        endianness = 0;

#elif __BIG_ENDIAN__

        endianness = 1;

        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 { 

                                err_ret("(%s) ERROR - End of file from OGG stream", prog);

        /*set the last fields*/

        *bitsize = size;

        *freq    = vorbisInfo->rate;

        /*cleaning time*/

        ov_clear(&oggStream);

        return AL_TRUE;