openalbridge/loaders.c
author koda
Mon, 29 Jun 2009 23:03:25 +0000
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child 2213 bd51bbf06033
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-openalbridge headers cleaned -static building of lib for non MSVC compilers
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/*
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 * OpenAL Bridge - a simple portable library for OpenAL interface
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 * Copyright (c) 2009 Vittorio Giovara <vittorio.giovara@gmail.com>
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 *
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 * This program is free software; you can redistribute it and/or modify
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 * it under the terms of the GNU General Public License as published by
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 * the Free Software Foundation; version 2 of the License
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 *
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 * This program is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
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 * GNU General Public License for more details.
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 *
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 * You should have received a copy of the GNU General Public License
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 * along with this program; if not, write to the Free Software
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 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA
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 */
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#include "loaders.h"
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#ifdef __CPLUSPLUS
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extern "C" {
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#endif 
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	int load_WavPcm (const char *filename, ALenum *format, uint8_t** data, ALsizei *bitsize, ALsizei *freq) {
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		WAV_header_t WAVHeader;
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		FILE *wavfile;
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		int t, n = 0;
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		wavfile = Fopen(filename, "rb");
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		fread(&WAVHeader.ChunkID, sizeof(uint32_t), 1, wavfile);
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		fread(&WAVHeader.ChunkSize, sizeof(uint32_t), 1, wavfile);
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		fread(&WAVHeader.Format, sizeof(uint32_t), 1, wavfile);
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#ifdef DEBUG
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		fprintf(stderr, "ChunkID: %X\n", invert_endianness(WAVHeader.ChunkID));
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		fprintf(stderr, "ChunkSize: %d\n", WAVHeader.ChunkSize);
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		fprintf(stderr, "Format: %X\n", invert_endianness(WAVHeader.Format));
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#endif
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		fread(&WAVHeader.Subchunk1ID, sizeof(uint32_t), 1, wavfile);
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		fread(&WAVHeader.Subchunk1Size, sizeof(uint32_t), 1, wavfile);
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		fread(&WAVHeader.AudioFormat, sizeof(uint16_t), 1, wavfile);
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		fread(&WAVHeader.NumChannels, sizeof(uint16_t), 1, wavfile);
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		fread(&WAVHeader.SampleRate, sizeof(uint32_t), 1, wavfile);
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		fread(&WAVHeader.ByteRate, sizeof(uint32_t), 1, wavfile);
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		fread(&WAVHeader.BlockAlign, sizeof(uint16_t), 1, wavfile);
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		fread(&WAVHeader.BitsPerSample, sizeof(uint16_t), 1, wavfile);
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#ifdef DEBUG
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		fprintf(stderr, "Subchunk1ID: %X\n", invert_endianness(WAVHeader.Subchunk1ID));
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		fprintf(stderr, "Subchunk1Size: %d\n", WAVHeader.Subchunk1Size);
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		fprintf(stderr, "AudioFormat: %d\n", WAVHeader.AudioFormat);
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		fprintf(stderr, "NumChannels: %d\n", WAVHeader.NumChannels);
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		fprintf(stderr, "SampleRate: %d\n", WAVHeader.SampleRate);
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		fprintf(stderr, "ByteRate: %d\n", WAVHeader.ByteRate);
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		fprintf(stderr, "BlockAlign: %d\n", WAVHeader.BlockAlign);
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		fprintf(stderr, "BitsPerSample: %d\n", WAVHeader.BitsPerSample);
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#endif
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		do { /*remove useless header chunks (plenty room for improvements)*/
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			t = fread(&WAVHeader.Subchunk2ID, sizeof(uint32_t), 1, wavfile);
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			if (invert_endianness(WAVHeader.Subchunk2ID) == 0x64617461)
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				break;
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			if (t <= 0) { /*eof*/
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				fprintf(stderr, "ERROR: wrong WAV header\n");
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				return AL_FALSE;
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			}
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		} while (1);
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		fread(&WAVHeader.Subchunk2Size, sizeof(uint32_t), 1, wavfile);
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#ifdef DEBUG
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		fprintf(stderr, "Subchunk2ID: %X\n", invert_endianness(WAVHeader.Subchunk2ID));
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		fprintf(stderr, "Subchunk2Size: %d\n", WAVHeader.Subchunk2Size);
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#endif
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		*data = (uint8_t*) malloc (sizeof(uint8_t) * WAVHeader.Subchunk2Size);
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		/*this could be improved*/
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		do {
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			n += fread(&((*data)[n]), sizeof(uint8_t), 1, wavfile);
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		} while (n < WAVHeader.Subchunk2Size);
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		fclose(wavfile);	
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#ifdef DEBUG
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		fprintf(stderr, "Last two bytes of data: %X%X\n", (*data)[n-2], (*data)[n-1]);
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#endif
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		/*remaining parameters*/
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		/*Valid formats are AL_FORMAT_MONO8, AL_FORMAT_MONO16, AL_FORMAT_STEREO8, and AL_FORMAT_STEREO16*/
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		if (WAVHeader.NumChannels == 1) {
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			if (WAVHeader.BitsPerSample == 8)
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				*format = AL_FORMAT_MONO8;
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			else {
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				if (WAVHeader.BitsPerSample == 16)
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					*format = AL_FORMAT_MONO16;
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				else {
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					fprintf(stderr, "ERROR: wrong WAV header - bitsample value\n");
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					return AL_FALSE;
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				}
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			} 
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		} else {
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			if (WAVHeader.NumChannels == 2) {
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				if (WAVHeader.BitsPerSample == 8)
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					*format = AL_FORMAT_STEREO8;
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				else {
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					if (WAVHeader.BitsPerSample == 16)
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						*format = AL_FORMAT_STEREO16;
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					else {
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						fprintf(stderr, "ERROR: wrong WAV header - bitsample value\n");
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						return AL_FALSE;
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					}				
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				}
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			} else {
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				fprintf(stderr, "ERROR: wrong WAV header - format value\n");
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				return AL_FALSE;
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			}
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		}
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		*bitsize = WAVHeader.Subchunk2Size;
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		*freq = WAVHeader.SampleRate;
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		return AL_TRUE;
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	}
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	int load_OggVorbis (const char *filename, ALenum *format, uint8_t**data, ALsizei *bitsize, ALsizei *freq) {
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		/*implementation inspired from http://www.devmaster.net/forums/showthread.php?t=1153 */
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		FILE			*oggFile;		/*ogg handle*/
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		OggVorbis_File  oggStream;		/*stream handle*/
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		vorbis_info		*vorbisInfo;	/*some formatting data*/
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		int64_t			pcm_length;		/*length of the decoded data*/
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		int size = 0;
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		int section, result;
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#ifdef DEBUG
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		int i;
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		vorbis_comment	*vorbisComment;	/*other less useful data*/
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#endif
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		oggFile = Fopen(filename, "rb");
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		result = ov_open(oggFile, &oggStream, NULL, 0);
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		/*TODO: check returning value of result*/
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		vorbisInfo = ov_info(&oggStream, -1);
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		pcm_length = ov_pcm_total(&oggStream,-1) << vorbisInfo->channels;	
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#ifdef DEBUG
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		vorbisComment = ov_comment(&oggStream, -1);
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		fprintf(stderr, "Version: %d\n", vorbisInfo->version);
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		fprintf(stderr, "Channels: %d\n", vorbisInfo->channels);
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		fprintf(stderr, "Rate (Hz): %d\n", vorbisInfo->rate);
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		fprintf(stderr, "Bitrate Upper: %d\n", vorbisInfo->bitrate_upper);
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		fprintf(stderr, "Bitrate Nominal: %d\n", vorbisInfo->bitrate_nominal);
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		fprintf(stderr, "Bitrate Lower: %d\n", vorbisInfo->bitrate_lower);
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		fprintf(stderr, "Bitrate Windows: %d\n", vorbisInfo->bitrate_window);
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		fprintf(stderr, "Vendor: %s\n", vorbisComment->vendor);
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		fprintf(stderr, "PCM data size: %d\n", pcm_length);
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		fprintf(stderr, "# comment: %d\n", vorbisComment->comments);
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		for (i = 0; i < vorbisComment->comments; i++)
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			fprintf(stderr, "\tComment %d: %s\n", i, vorbisComment->user_comments[i]);
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#endif
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		/*allocates enough room for the decoded data*/
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		*data = (uint8_t*) malloc (sizeof(uint8_t) * pcm_length);
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		/*there *should* not be ogg at 8 bits*/
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		if (vorbisInfo->channels == 1)
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			*format = AL_FORMAT_MONO16;
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		else {
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			if (vorbisInfo->channels == 2)
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				*format = AL_FORMAT_STEREO16;
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			else {
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				fprintf(stderr, "ERROR: wrong OGG header - channel value (%d)\n", vorbisInfo->channels);
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				return AL_FALSE;
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			}
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		}
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		while(size < pcm_length)	{
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			/*ov_read decodes the ogg stream and storse the pcm in data*/
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			result = ov_read (&oggStream, *data + size, pcm_length - size, 0, 2, 1, &section);
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			if(result > 0) {
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				size += result;
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			} else {
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				if (result == 0)
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					break;
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				else { 
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					fprintf(stderr, "ERROR: end of file from OGG stream\n");
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					return AL_FALSE;
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				}
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			}
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		}
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		/*records the last fields*/
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		*bitsize = size;
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		*freq = vorbisInfo->rate;
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		return AL_TRUE;
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	}
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#ifdef __CPLUSPLUS
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}
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#endif