openalbridge/loaders.c
author nemo
Wed, 26 Aug 2009 20:32:27 +0000
changeset 2332 351abbbb12f3
parent 2266 289dc8e51210
child 2415 35d09cbf819a
permissions -rw-r--r--
Lower bound on speech text width for proper rendering
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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 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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 * GNU Lesser General Public License for more details.
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 *
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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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    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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        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 = (char*) Malloc (sizeof(char) * 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, 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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        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             section, result, size = 0;
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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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	result = ov_fopen((char*) filename, &oggStream);
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	if (result < 0) {
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		fprintf (stderr, "ERROR: ov_open_callbacks failed with %X", result);
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                ov_clear(&oggStream);
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		return -1;
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	}
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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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                fprintf(stderr, "ERROR: wrong OGG header - channel value (%d)\n", 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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        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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                    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