/*
* 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
extern int ov_open(FILE *f,OggVorbis_File *vf,char *initial,long ibytes);
extern long ov_read(OggVorbis_File *vf,char *buffer,int length,int bigendianp,int word,int sgned,int *bitstream);
extern ogg_int64_t ov_pcm_total(OggVorbis_File *vf,int i);
extern long ov_read(OggVorbis_File *vf,char *buffer,int length,int bigendianp,int word,int sgned,int *bitstream);
extern vorbis_info *ov_info(OggVorbis_File *vf,int link);
extern vorbis_comment *ov_comment(OggVorbis_File *f, int num);
int load_WavPcm (const char *filename, ALenum *format, uint8_t** data, ALsizei *bitsize, ALsizei *freq) {
WAV_header_t WAVHeader;
FILE *wavfile;
int 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 found
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 = (uint8_t*) malloc (sizeof(uint8_t) * 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, uint8_t**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
vorbis_comment *vorbisComment; // other less useful data
int64_t pcm_length; // length of the decoded data
int size = 0;
int section, result, i;
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): %d\n", vorbisInfo->rate);
fprintf(stderr, "Bitrate Upper: %d\n", vorbisInfo->bitrate_upper);
fprintf(stderr, "Bitrate Nominal: %d\n", vorbisInfo->bitrate_nominal);
fprintf(stderr, "Bitrate Lower: %d\n", vorbisInfo->bitrate_lower);
fprintf(stderr, "Bitrate Windows: %d\n", vorbisInfo->bitrate_window);
fprintf(stderr, "Vendor: %s\n", vorbisComment->vendor);
fprintf(stderr, "PCM data size: %d\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 = (uint8_t*) malloc (sizeof(uint8_t) * 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, §ion);
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