--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/misc/openalbridge/loaders.c Sat Apr 17 08:30:34 2010 +0000
@@ -0,0 +1,248 @@
+/*
+ * 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"
+#include "wrappers.h"
+#include "vorbis/vorbisfile.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