author | smxx |
Wed, 03 Mar 2010 16:19:37 +0000 | |
changeset 2925 | d088b5f813c9 |
parent 2529 | 51e5df1c8462 |
permissions | -rw-r--r-- |
2191 | 1 |
/* |
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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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7eb31efcfb9b
updates licence and fix a memory leak (which was consuming iphone memory)
koda
parents:
2220
diff
changeset
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* it under the terms of the GNU Lesser 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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2257
7eb31efcfb9b
updates licence and fix a memory leak (which was consuming iphone memory)
koda
parents:
2220
diff
changeset
|
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* GNU Lesser General Public License for more details. |
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* |
2257
7eb31efcfb9b
updates licence and fix a memory leak (which was consuming iphone memory)
koda
parents:
2220
diff
changeset
|
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* You should have received a copy of the GNU Lesser 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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2443
fececcbc2189
Smaxx patch for fixing all sound related issues on any system
koda
parents:
2421
diff
changeset
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int load_wavpcm (const char *filename, ALenum *format, char ** data, ALsizei *bitsize, ALsizei *freq) { |
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WAV_header_t WAVHeader; |
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FILE *wavfile; |
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int32_t t; |
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uint32_t n = 0; |
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uint8_t sub0, sub1, sub2, sub3; |
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wavfile = Fopen(filename, "rb"); |
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fread(&WAVHeader.ChunkID, sizeof(uint32_t), 1, wavfile); /*RIFF*/ |
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fread(&WAVHeader.ChunkSize, sizeof(uint32_t), 1, wavfile); |
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fread(&WAVHeader.Format, sizeof(uint32_t), 1, wavfile); /*WAVE*/ |
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#ifdef DEBUG |
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fprintf(stderr, "ChunkID: %X\n", ENDIAN_BIG_32(WAVHeader.ChunkID)); |
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fprintf(stderr, "ChunkSize: %d\n", ENDIAN_LITTLE_32(WAVHeader.ChunkSize)); |
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fprintf(stderr, "Format: %X\n", ENDIAN_BIG_32(WAVHeader.Format)); |
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#endif |
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fread(&WAVHeader.Subchunk1ID, sizeof(uint32_t), 1, wavfile); /*fmt */ |
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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", ENDIAN_BIG_32(WAVHeader.Subchunk1ID)); |
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fprintf(stderr, "Subchunk1Size: %d\n", ENDIAN_LITTLE_32(WAVHeader.Subchunk1Size)); |
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fprintf(stderr, "AudioFormat: %d\n", ENDIAN_LITTLE_16(WAVHeader.AudioFormat)); |
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fprintf(stderr, "NumChannels: %d\n", ENDIAN_LITTLE_16(WAVHeader.NumChannels)); |
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fprintf(stderr, "SampleRate: %d\n", ENDIAN_LITTLE_32(WAVHeader.SampleRate)); |
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fprintf(stderr, "ByteRate: %d\n", ENDIAN_LITTLE_32(WAVHeader.ByteRate)); |
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fprintf(stderr, "BlockAlign: %d\n", ENDIAN_LITTLE_16(WAVHeader.BlockAlign)); |
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fprintf(stderr, "BitsPerSample: %d\n", ENDIAN_LITTLE_16(WAVHeader.BitsPerSample)); |
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#endif |
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/*remove useless header chunks by looking for the WAV_HEADER_SUBCHUNK2ID integer */ |
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do { |
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t = fread(&sub0, sizeof(uint8_t), 1, wavfile); |
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if(sub0 == 0x64) { |
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t = fread(&sub1, sizeof(uint8_t), 1, wavfile); |
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if(sub1 == 0x61) { |
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t = fread(&sub2, sizeof(uint8_t), 1, wavfile); |
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if(sub2 == 0x74) { |
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t = fread(&sub3, sizeof(uint8_t), 1, wavfile); |
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if(sub3 == 0x61) { |
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WAVHeader.Subchunk2ID = WAV_HEADER_SUBCHUNK2ID; |
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break; |
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} |
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} |
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} |
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} |
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if (t <= 0) { |
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/*eof*/ |
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errno = EILSEQ; |
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err_ret("(%s) ERROR - wrong WAV header", prog); |
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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", ENDIAN_LITTLE_32(WAVHeader.Subchunk2ID)); |
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fprintf(stderr, "Subchunk2Size: %d\n", ENDIAN_LITTLE_32(WAVHeader.Subchunk2Size)); |
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#endif |
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*data = (char*) Malloc (sizeof(char) * ENDIAN_LITTLE_32(WAVHeader.Subchunk2Size)); |
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/*read the actual sound data*/ |
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do { |
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n += fread(&((*data)[n]), sizeof(uint8_t), 4, wavfile); |
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} while (n < ENDIAN_LITTLE_32(WAVHeader.Subchunk2Size)); |
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fclose(wavfile); |
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#ifdef DEBUG |
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err_msg("(%s) INFO - WAV data loaded", prog); |
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#endif |
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/*set parameters for OpenAL*/ |
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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 (ENDIAN_LITTLE_16(WAVHeader.NumChannels) == 1) { |
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if (ENDIAN_LITTLE_16(WAVHeader.BitsPerSample) == 8) |
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*format = AL_FORMAT_MONO8; |
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else { |
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if (ENDIAN_LITTLE_16(WAVHeader.BitsPerSample) == 16) |
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*format = AL_FORMAT_MONO16; |
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else { |
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errno = EILSEQ; |
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err_ret("(%s) ERROR - wrong WAV header [bitsample value]", prog); |
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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 (ENDIAN_LITTLE_16(WAVHeader.NumChannels) == 2) { |
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if (ENDIAN_LITTLE_16(WAVHeader.BitsPerSample) == 8) |
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*format = AL_FORMAT_STEREO8; |
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else { |
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if (ENDIAN_LITTLE_16(WAVHeader.BitsPerSample) == 16) |
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*format = AL_FORMAT_STEREO16; |
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else { |
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errno = EILSEQ; |
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err_ret("(%s) ERROR - wrong WAV header [bitsample value]", prog); |
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return AL_FALSE; |
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} |
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} |
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} else { |
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errno = EILSEQ; |
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err_ret("(%s) ERROR - wrong WAV header [format value]", prog); |
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return AL_FALSE; |
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} |
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} |
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*bitsize = ENDIAN_LITTLE_32(WAVHeader.Subchunk2Size); |
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*freq = ENDIAN_LITTLE_32(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, char **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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/*stream handle*/ |
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OggVorbis_File oggStream; |
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/*some formatting data*/ |
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vorbis_info *vorbisInfo; |
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/*length of the decoded data*/ |
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int64_t pcm_length; |
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/*other vars*/ |
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int section, result, size, endianness; |
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2443
fececcbc2189
Smaxx patch for fixing all sound related issues on any system
koda
parents:
2421
diff
changeset
|
161 |
#ifdef DEBUG |
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int i; |
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/*other less useful data*/ |
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vorbis_comment *vorbisComment; |
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2443
fececcbc2189
Smaxx patch for fixing all sound related issues on any system
koda
parents:
2421
diff
changeset
|
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#endif |
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result = ov_fopen((char*) filename, &oggStream); |
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if (result < 0) { |
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errno = EINVAL; |
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err_ret("(%s) ERROR - ov_fopen() failed with %X", prog, result); |
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ov_clear(&oggStream); |
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return AL_FALSE; |
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} |
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/*load OGG header and determine the decoded data size*/ |
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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): %ld\n", vorbisInfo->rate); |
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fprintf(stderr, "Bitrate Upper: %ld\n", vorbisInfo->bitrate_upper); |
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fprintf(stderr, "Bitrate Nominal: %ld\n", vorbisInfo->bitrate_nominal); |
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fprintf(stderr, "Bitrate Lower: %ld\n", vorbisInfo->bitrate_lower); |
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fprintf(stderr, "Bitrate Windows: %ld\n", vorbisInfo->bitrate_window); |
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fprintf(stderr, "Vendor: %s\n", vorbisComment->vendor); |
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fprintf(stderr, "PCM data size: %lld\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 = (char*) Malloc (sizeof(char) * 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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errno = EILSEQ; |
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err_ret("(%s) ERROR - wrong OGG header [channel %d]", prog, vorbisInfo->channels); |
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ov_clear(&oggStream); |
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return AL_FALSE; |
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} |
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} |
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size = 0; |
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#ifdef __LITTLE_ENDIAN__ |
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endianness = 0; |
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#elif __BIG_ENDIAN__ |
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endianness = 1; |
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#endif |
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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, endianness, 2, 1, §ion); |
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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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errno = EILSEQ; |
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err_ret("(%s) ERROR - End of file from OGG stream", prog); |
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ov_clear(&oggStream); |
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return AL_FALSE; |
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} |
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} |
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} |
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/*set the last fields*/ |
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*bitsize = size; |
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*freq = vorbisInfo->rate; |
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/*cleaning time*/ |
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ov_clear(&oggStream); |
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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 |