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-rw-r--r--faad2/src/frontend/audio.c500
1 files changed, 500 insertions, 0 deletions
diff --git a/faad2/src/frontend/audio.c b/faad2/src/frontend/audio.c
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+++ b/faad2/src/frontend/audio.c
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+/*
+** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
+** Copyright (C) 2003-2005 M. Bakker, Nero AG, http://www.nero.com
+**
+** 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.
+**
+** Any non-GPL usage of this software or parts of this software is strictly
+** forbidden.
+**
+** The "appropriate copyright message" mentioned in section 2c of the GPLv2
+** must read: "Code from FAAD2 is copyright (c) Nero AG, www.nero.com"
+**
+** Commercial non-GPL licensing of this software is possible.
+** For more info contact Nero AG through Mpeg4AAClicense@nero.com.
+**
+** $Id: audio.c,v 1.29 2008/09/19 22:50:17 menno Exp $
+**/
+
+#ifdef _WIN32
+#include <io.h>
+#endif
+#include <stdlib.h>
+#include <stdio.h>
+#include <fcntl.h>
+#include <math.h>
+#include <neaacdec.h>
+#include "audio.h"
+
+
+audio_file *open_audio_file(char *infile, int samplerate, int channels,
+ int outputFormat, int fileType, long channelMask)
+{
+ audio_file *aufile = malloc(sizeof(audio_file));
+
+ aufile->outputFormat = outputFormat;
+
+ aufile->samplerate = samplerate;
+ aufile->channels = channels;
+ aufile->total_samples = 0;
+ aufile->fileType = fileType;
+ aufile->channelMask = channelMask;
+
+ switch (outputFormat)
+ {
+ case FAAD_FMT_16BIT:
+ 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;
+ }
+
+ if(infile[0] == '-')
+ {
+#ifdef _WIN32
+ setmode(fileno(stdout), O_BINARY);
+#endif
+ aufile->sndfile = stdout;
+ aufile->toStdio = 1;
+ } else {
+ aufile->toStdio = 0;
+ aufile->sndfile = fopen(infile, "wb");
+ }
+
+ if (aufile->sndfile == NULL)
+ {
+ if (aufile) free(aufile);
+ return NULL;
+ }
+
+ if (aufile->fileType == OUTPUT_WAV)
+ {
+ if (aufile->channelMask)
+ write_wav_extensible_header(aufile, aufile->channelMask);
+ else
+ write_wav_header(aufile);
+ }
+
+ return aufile;
+}
+
+int write_audio_file(audio_file *aufile, void *sample_buffer, int samples, int offset)
+{
+ char *buf = (char *)sample_buffer;
+ switch (aufile->outputFormat)
+ {
+ case FAAD_FMT_16BIT:
+ return write_audio_16bit(aufile, buf + offset*2, samples);
+ case FAAD_FMT_24BIT:
+ return write_audio_24bit(aufile, buf + offset*4, samples);
+ case FAAD_FMT_32BIT:
+ return write_audio_32bit(aufile, buf + offset*4, samples);
+ case FAAD_FMT_FLOAT:
+ return write_audio_float(aufile, buf + offset*4, samples);
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+void close_audio_file(audio_file *aufile)
+{
+ if ((aufile->fileType == OUTPUT_WAV) && (aufile->toStdio == 0))
+ {
+ fseek(aufile->sndfile, 0, SEEK_SET);
+
+ if (aufile->channelMask)
+ write_wav_extensible_header(aufile, aufile->channelMask);
+ else
+ write_wav_header(aufile);
+ }
+
+ if (aufile->toStdio == 0)
+ 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->total_samples;
+ unsigned long word32;
+
+ *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);
+
+ word32 = (unsigned long)(aufile->samplerate + 0.5);
+ *p++ = (unsigned char)(word32 >> 0);
+ *p++ = (unsigned char)(word32 >> 8);
+ *p++ = (unsigned char)(word32 >> 16);
+ *p++ = (unsigned char)(word32 >> 24);
+
+ word32 = aufile->samplerate * 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);
+
+ return fwrite(header, sizeof(header), 1, aufile->sndfile);
+}
+
+static int write_wav_extensible_header(audio_file *aufile, long channelMask)
+{
+ unsigned char header[68];
+ unsigned char* p = header;
+ unsigned int bytes = (aufile->bits_per_sample + 7) / 8;
+ float data_size = (float)bytes * aufile->total_samples;
+ unsigned long word32;
+
+ *p++ = 'R'; *p++ = 'I'; *p++ = 'F'; *p++ = 'F';
+
+ word32 = (data_size + (68 - 8) < (float)MAXWAVESIZE) ?
+ (unsigned long)data_size + (68 - 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*/0x28; *p++ = 0x00; *p++ = 0x00; *p++ = 0x00;
+
+ /* WAVE_FORMAT_EXTENSIBLE */
+ *p++ = 0xFE; *p++ = 0xFF;
+
+ *p++ = (unsigned char)(aufile->channels >> 0);
+ *p++ = (unsigned char)(aufile->channels >> 8);
+
+ word32 = (unsigned long)(aufile->samplerate + 0.5);
+ *p++ = (unsigned char)(word32 >> 0);
+ *p++ = (unsigned char)(word32 >> 8);
+ *p++ = (unsigned char)(word32 >> 16);
+ *p++ = (unsigned char)(word32 >> 24);
+
+ word32 = aufile->samplerate * 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);
+
+ /* cbSize */
+ *p++ = (unsigned char)(22);
+ *p++ = (unsigned char)(0);
+
+ /* WAVEFORMATEXTENSIBLE */
+
+ /* wValidBitsPerSample */
+ *p++ = (unsigned char)(aufile->bits_per_sample >> 0);
+ *p++ = (unsigned char)(aufile->bits_per_sample >> 8);
+
+ /* dwChannelMask */
+ word32 = channelMask;
+ *p++ = (unsigned char)(word32 >> 0);
+ *p++ = (unsigned char)(word32 >> 8);
+ *p++ = (unsigned char)(word32 >> 16);
+ *p++ = (unsigned char)(word32 >> 24);
+
+ /* SubFormat */
+ if (aufile->outputFormat == FAAD_FMT_FLOAT)
+ {
+ /* KSDATAFORMAT_SUBTYPE_IEEE_FLOAT: 00000003-0000-0010-8000-00aa00389b71 */
+ *p++ = 0x03;
+ *p++ = 0x00;
+ *p++ = 0x00;
+ *p++ = 0x00;
+ *p++ = 0x00; *p++ = 0x00; *p++ = 0x10; *p++ = 0x00; *p++ = 0x80; *p++ = 0x00;
+ *p++ = 0x00; *p++ = 0xaa; *p++ = 0x00; *p++ = 0x38; *p++ = 0x9b; *p++ = 0x71;
+ } else {
+ /* KSDATAFORMAT_SUBTYPE_PCM: 00000001-0000-0010-8000-00aa00389b71 */
+ *p++ = 0x01;
+ *p++ = 0x00;
+ *p++ = 0x00;
+ *p++ = 0x00;
+ *p++ = 0x00; *p++ = 0x00; *p++ = 0x10; *p++ = 0x00; *p++ = 0x80; *p++ = 0x00;
+ *p++ = 0x00; *p++ = 0xaa; *p++ = 0x00; *p++ = 0x38; *p++ = 0x9b; *p++ = 0x71;
+ }
+
+ /* end WAVEFORMATEXTENSIBLE */
+
+ *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);
+
+ return fwrite(header, sizeof(header), 1, aufile->sndfile);
+}
+
+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->total_samples += samples;
+
+ if (aufile->channels == 6 && aufile->channelMask)
+ {
+ for (i = 0; i < samples; i += aufile->channels)
+ {
+ short r1, r2, r3, r4, r5, r6;
+ r1 = sample_buffer16[i];
+ r2 = sample_buffer16[i+1];
+ r3 = sample_buffer16[i+2];
+ r4 = sample_buffer16[i+3];
+ r5 = sample_buffer16[i+4];
+ r6 = sample_buffer16[i+5];
+ sample_buffer16[i] = r2;
+ sample_buffer16[i+1] = r3;
+ sample_buffer16[i+2] = r1;
+ sample_buffer16[i+3] = r6;
+ sample_buffer16[i+4] = r4;
+ sample_buffer16[i+5] = r5;
+ }
+ }
+
+ for (i = 0; i < samples; i++)
+ {
+ data[i*2] = (char)(sample_buffer16[i] & 0xFF);
+ data[i*2+1] = (char)((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->total_samples += samples;
+
+ if (aufile->channels == 6 && aufile->channelMask)
+ {
+ for (i = 0; i < samples; i += aufile->channels)
+ {
+ long r1, r2, r3, r4, r5, r6;
+ r1 = sample_buffer24[i];
+ r2 = sample_buffer24[i+1];
+ r3 = sample_buffer24[i+2];
+ r4 = sample_buffer24[i+3];
+ r5 = sample_buffer24[i+4];
+ r6 = sample_buffer24[i+5];
+ sample_buffer24[i] = r2;
+ sample_buffer24[i+1] = r3;
+ sample_buffer24[i+2] = r1;
+ sample_buffer24[i+3] = r6;
+ sample_buffer24[i+4] = r4;
+ sample_buffer24[i+5] = r5;
+ }
+ }
+
+ for (i = 0; i < samples; i++)
+ {
+ data[i*3] = (char)(sample_buffer24[i] & 0xFF);
+ data[i*3+1] = (char)((sample_buffer24[i] >> 8) & 0xFF);
+ data[i*3+2] = (char)((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->total_samples += samples;
+
+ if (aufile->channels == 6 && aufile->channelMask)
+ {
+ for (i = 0; i < samples; i += aufile->channels)
+ {
+ long r1, r2, r3, r4, r5, r6;
+ r1 = sample_buffer32[i];
+ r2 = sample_buffer32[i+1];
+ r3 = sample_buffer32[i+2];
+ r4 = sample_buffer32[i+3];
+ r5 = sample_buffer32[i+4];
+ r6 = sample_buffer32[i+5];
+ sample_buffer32[i] = r2;
+ sample_buffer32[i+1] = r3;
+ sample_buffer32[i+2] = r1;
+ sample_buffer32[i+3] = r6;
+ sample_buffer32[i+4] = r4;
+ sample_buffer32[i+5] = r5;
+ }
+ }
+
+ for (i = 0; i < samples; i++)
+ {
+ data[i*4] = (char)(sample_buffer32[i] & 0xFF);
+ data[i*4+1] = (char)((sample_buffer32[i] >> 8) & 0xFF);
+ data[i*4+2] = (char)((sample_buffer32[i] >> 16) & 0xFF);
+ data[i*4+3] = (char)((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->total_samples += samples;
+
+ if (aufile->channels == 6 && aufile->channelMask)
+ {
+ for (i = 0; i < samples; i += aufile->channels)
+ {
+ float r1, r2, r3, r4, r5, r6;
+ r1 = sample_buffer_f[i];
+ r2 = sample_buffer_f[i+1];
+ r3 = sample_buffer_f[i+2];
+ r4 = sample_buffer_f[i+3];
+ r5 = sample_buffer_f[i+4];
+ r6 = sample_buffer_f[i+5];
+ sample_buffer_f[i] = r2;
+ sample_buffer_f[i+1] = r3;
+ sample_buffer_f[i+2] = r1;
+ sample_buffer_f[i+3] = r6;
+ sample_buffer_f[i+4] = r4;
+ sample_buffer_f[i+5] = r5;
+ }
+ }
+
+ 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 = (float)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;
+}