tweaked hellish impact sound a bit. and yeah, we really need to limit how frequently impact sounds can be triggered on bounce >.<
/*
* 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;
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);
#ifdef DEBUG
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));
#endif
/*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);
#ifdef DEBUG
fprintf(stderr, "Subchunk2ID: %X\n", ENDIAN_LITTLE_32(WAVHeader.Subchunk2ID));
fprintf(stderr, "Subchunk2Size: %d\n", ENDIAN_LITTLE_32(WAVHeader.Subchunk2Size));
#endif
*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 {
errno = EILSEQ;
err_ret("(%s) ERROR - wrong WAV header [bitsample value]", prog);
return AL_FALSE;
}
}
} else {
if (ENDIAN_LITTLE_16(WAVHeader.NumChannels) == 2) {
if (ENDIAN_LITTLE_16(WAVHeader.BitsPerSample) == 8)
*format = AL_FORMAT_STEREO8;
else {
if (ENDIAN_LITTLE_16(WAVHeader.BitsPerSample) == 16)
*format = AL_FORMAT_STEREO16;
else {
errno = EILSEQ;
err_ret("(%s) ERROR - wrong WAV header [bitsample value]", prog);
return AL_FALSE;
}
}
} else {
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;
#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 {
errno = EILSEQ;
err_ret("(%s) ERROR - wrong OGG header [channel %d]", prog, vorbisInfo->channels);
ov_clear(&oggStream);
return AL_FALSE;
}
}
size = 0;
#ifdef __LITTLE_ENDIAN__
endianness = 0;
#elif __BIG_ENDIAN__
endianness = 1;
#endif
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, §ion);
if (result > 0) {
size += result;
} else {
if (result == 0)
break;
else {
errno = EILSEQ;
err_ret("(%s) ERROR - End of file from OGG stream", prog);
ov_clear(&oggStream);
return AL_FALSE;
}
}
}
/*set the last fields*/
*bitsize = size;
*freq = vorbisInfo->rate;
/*cleaning time*/
ov_clear(&oggStream);
return AL_TRUE;
}
#ifdef __CPLUSPLUS
}
#endif