/*

 * 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 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 General Public License for more details.

 *

 * You should have received a copy of the GNU 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*/

		int size = 0;

		int section, result;

#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: %ld\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;

		return AL_TRUE;

	}

#ifdef __CPLUSPLUS

}

#endif