ref: 707ae6e5bee8fb4dcd24b39ff71c072c0e29e02d
dir: /aacDECdrop/audio.c/
/* ** FAAD - Freeware Advanced Audio Decoder ** Copyright (C) 2002 M. Bakker ** ** 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; either version 2 of the License, or ** (at your option) any later version. ** ** 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. ** ** $Id: audio.c,v 1.6 2002/08/14 10:55:28 menno Exp $ **/ #ifdef _WIN32 #include <io.h> #endif #include <stdlib.h> #include <stdio.h> #include <fcntl.h> #include <math.h> #include <faad.h> #include "audio.h" audio_file *open_audio_file(char *infile, int samplerate, int channels, int outputFormat, int fileType) { audio_file *aufile = malloc(sizeof(audio_file)); aufile->outputFormat = outputFormat; aufile->samplerate = samplerate; aufile->channels = channels; aufile->samples = 0; aufile->fileType = fileType; switch (outputFormat) { case FAAD_FMT_16BIT: case FAAD_FMT_16BIT_DITHER: aufile->bits_per_sample = 16; break; case FAAD_FMT_24BIT: aufile->bits_per_sample = 24; break; case FAAD_FMT_32BIT: case FAAD_FMT_FLOAT: aufile->bits_per_sample = 32; break; default: if (aufile) free(aufile); return NULL; } #ifdef _WIN32 if(infile[0] == '-') { setmode(fileno(stdout), O_BINARY); } #endif aufile->sndfile = fopen(infile, "wb"); if (aufile->sndfile == NULL) { if (aufile) free(aufile); return NULL; } if (aufile->fileType == OUTPUT_WAV) { write_wav_header(aufile); } return aufile; } int write_audio_file(audio_file *aufile, void *sample_buffer, int samples) { switch (aufile->outputFormat) { case FAAD_FMT_16BIT: case FAAD_FMT_16BIT_DITHER: return write_audio_16bit(aufile, sample_buffer, samples); case FAAD_FMT_24BIT: return write_audio_24bit(aufile, sample_buffer, samples); case FAAD_FMT_32BIT: return write_audio_32bit(aufile, sample_buffer, samples); case FAAD_FMT_FLOAT: return write_audio_float(aufile, sample_buffer, samples); default: return 0; } return 0; } void close_audio_file(audio_file *aufile) { if (aufile->fileType == OUTPUT_WAV) { fseek(aufile->sndfile, 0, SEEK_SET); write_wav_header(aufile); } fclose(aufile->sndfile); if (aufile) free(aufile); } static int write_wav_header(audio_file *aufile) { unsigned char header[44]; unsigned char* p = header; unsigned int bytes = (aufile->bits_per_sample + 7) / 8; float data_size = (float)bytes * aufile->samples; unsigned long word32; int ret; *p++ = 'R'; *p++ = 'I'; *p++ = 'F'; *p++ = 'F'; word32 = data_size + (44 - 8) < (float)MAXWAVESIZE ? (unsigned long)data_size + (44 - 8) : (unsigned long)MAXWAVESIZE; *p++ = (unsigned char)(word32 >> 0); *p++ = (unsigned char)(word32 >> 8); *p++ = (unsigned char)(word32 >> 16); *p++ = (unsigned char)(word32 >> 24); *p++ = 'W'; *p++ = 'A'; *p++ = 'V'; *p++ = 'E'; *p++ = 'f'; *p++ = 'm'; *p++ = 't'; *p++ = ' '; *p++ = 0x10; *p++ = 0x00; *p++ = 0x00; *p++ = 0x00; if (aufile->outputFormat == FAAD_FMT_FLOAT) { *p++ = 0x03; *p++ = 0x00; } else { *p++ = 0x01; *p++ = 0x00; } *p++ = (unsigned char)(aufile->channels >> 0); *p++ = (unsigned char)(aufile->channels >> 8); *p++ = (unsigned char)(aufile->samplerate >> 0); *p++ = (unsigned char)(aufile->samplerate >> 8); *p++ = (unsigned char)(aufile->samplerate >> 16); *p++ = (unsigned char)(aufile->samplerate >> 24); word32 *= bytes * aufile->channels; *p++ = (unsigned char)(word32 >> 0); *p++ = (unsigned char)(word32 >> 8); *p++ = (unsigned char)(word32 >> 16); *p++ = (unsigned char)(word32 >> 24); word32 = bytes * aufile->channels; *p++ = (unsigned char)(word32 >> 0); *p++ = (unsigned char)(word32 >> 8); *p++ = (unsigned char)(aufile->bits_per_sample >> 0); *p++ = (unsigned char)(aufile->bits_per_sample >> 8); *p++ = 'd'; *p++ = 'a'; *p++ = 't'; *p++ = 'a'; word32 = data_size < MAXWAVESIZE ? (unsigned long)data_size : (unsigned long)MAXWAVESIZE; *p++ = (unsigned char)(word32 >> 0); *p++ = (unsigned char)(word32 >> 8); *p++ = (unsigned char)(word32 >> 16); *p++ = (unsigned char)(word32 >> 24); ret = fwrite(header, sizeof(header), 1, aufile->sndfile); return ret; } static int write_audio_16bit(audio_file *aufile, void *sample_buffer, unsigned int samples) { int ret; unsigned int i; short *sample_buffer16 = (short*)sample_buffer; char *data = malloc(samples*aufile->bits_per_sample*sizeof(char)/8); aufile->samples += samples; for (i = 0; i < samples; i++) { data[i*2] = sample_buffer16[i] & 0xFF; data[i*2+1] = (sample_buffer16[i] >> 8) & 0xFF; } ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile); if (data) free(data); return ret; } static int write_audio_24bit(audio_file *aufile, void *sample_buffer, unsigned int samples) { int ret; unsigned int i; long *sample_buffer24 = (long*)sample_buffer; char *data = malloc(samples*aufile->bits_per_sample*sizeof(char)/8); aufile->samples += samples; for (i = 0; i < samples; i++) { data[i*3] = sample_buffer24[i] & 0xFF; data[i*3+1] = (sample_buffer24[i] >> 8) & 0xFF; data[i*3+2] = (sample_buffer24[i] >> 16) & 0xFF; } ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile); if (data) free(data); return ret; } static int write_audio_32bit(audio_file *aufile, void *sample_buffer, unsigned int samples) { int ret; unsigned int i; long *sample_buffer32 = (long*)sample_buffer; char *data = malloc(samples*aufile->bits_per_sample*sizeof(char)/8); aufile->samples += samples; for (i = 0; i < samples; i++) { data[i*4] = sample_buffer32[i] & 0xFF; data[i*4+1] = (sample_buffer32[i] >> 8) & 0xFF; data[i*4+2] = (sample_buffer32[i] >> 16) & 0xFF; data[i*4+3] = (sample_buffer32[i] >> 24) & 0xFF; } ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile); if (data) free(data); return ret; } static int write_audio_float(audio_file *aufile, void *sample_buffer, unsigned int samples) { int ret; unsigned int i; float *sample_buffer_f = (float*)sample_buffer; unsigned char *data = malloc(samples*aufile->bits_per_sample*sizeof(char)/8); aufile->samples += samples; for (i = 0; i < samples; i++) { int exponent, mantissa, negative = 0 ; float in = sample_buffer_f[i]; data[i*4] = 0; data[i*4+1] = 0; data[i*4+2] = 0; data[i*4+3] = 0; if (in == 0.0) continue; if (in < 0.0) { in *= -1.0; negative = 1; } in = frexp(in, &exponent); exponent += 126; in *= (float)0x1000000; mantissa = (((int)in) & 0x7FFFFF); if (negative) data[i*4+3] |= 0x80; if (exponent & 0x01) data[i*4+2] |= 0x80; data[i*4] = mantissa & 0xFF; data[i*4+1] = (mantissa >> 8) & 0xFF; data[i*4+2] |= (mantissa >> 16) & 0x7F; data[i*4+3] |= (exponent >> 1) & 0x7F; } ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile); if (data) free(data); return ret; }