diff options
Diffstat (limited to 'faad2/src/frontend')
| -rw-r--r-- | faad2/src/frontend/Makefile.am | 12 | ||||
| -rw-r--r-- | faad2/src/frontend/audio.c | 500 | ||||
| -rw-r--r-- | faad2/src/frontend/audio.h | 75 | ||||
| -rw-r--r-- | faad2/src/frontend/faad.man | 85 | ||||
| -rw-r--r-- | faad2/src/frontend/faad.sln | 36 | ||||
| -rw-r--r-- | faad2/src/frontend/faad.vcproj | 256 | ||||
| -rw-r--r-- | faad2/src/frontend/main.c | 1270 |
7 files changed, 2234 insertions, 0 deletions
diff --git a/faad2/src/frontend/Makefile.am b/faad2/src/frontend/Makefile.am new file mode 100644 index 0000000..8bda787 --- /dev/null +++ b/faad2/src/frontend/Makefile.am @@ -0,0 +1,12 @@ +bin_PROGRAMS = faad
+man_MANS = faad.man
+
+INCLUDES = -I$(top_srcdir)/include -I$(top_srcdir)/common/faad \
+ -I$(top_srcdir)/common/mp4ff
+
+faad_LDADD = $(top_builddir)/libfaad/libfaad.la \
+ $(top_builddir)/common/mp4ff/libmp4ff.a
+
+faad_SOURCES = main.c \
+ audio.c audio.h \
+ $(top_srcdir)/common/faad/getopt.c
diff --git a/faad2/src/frontend/audio.c b/faad2/src/frontend/audio.c new file mode 100644 index 0000000..7691ce9 --- /dev/null +++ b/faad2/src/frontend/audio.c @@ -0,0 +1,500 @@ +/*
+** 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;
+}
diff --git a/faad2/src/frontend/audio.h b/faad2/src/frontend/audio.h new file mode 100644 index 0000000..b4d3a67 --- /dev/null +++ b/faad2/src/frontend/audio.h @@ -0,0 +1,75 @@ +/*
+** 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.h,v 1.19 2007/11/01 12:33:29 menno Exp $
+**/
+
+#ifndef AUDIO_H_INCLUDED
+#define AUDIO_H_INCLUDED
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define MAXWAVESIZE 4294967040LU
+
+#define OUTPUT_WAV 1
+#define OUTPUT_RAW 2
+
+typedef struct
+{
+ int toStdio;
+ int outputFormat;
+ FILE *sndfile;
+ unsigned int fileType;
+ unsigned long samplerate;
+ unsigned int bits_per_sample;
+ unsigned int channels;
+ unsigned long total_samples;
+ long channelMask;
+} audio_file;
+
+audio_file *open_audio_file(char *infile, int samplerate, int channels,
+ int outputFormat, int fileType, long channelMask);
+int write_audio_file(audio_file *aufile, void *sample_buffer, int samples, int offset);
+void close_audio_file(audio_file *aufile);
+static int write_wav_header(audio_file *aufile);
+static int write_wav_extensible_header(audio_file *aufile, long channelMask);
+static int write_audio_16bit(audio_file *aufile, void *sample_buffer,
+ unsigned int samples);
+static int write_audio_24bit(audio_file *aufile, void *sample_buffer,
+ unsigned int samples);
+static int write_audio_32bit(audio_file *aufile, void *sample_buffer,
+ unsigned int samples);
+static int write_audio_float(audio_file *aufile, void *sample_buffer,
+ unsigned int samples);
+
+
+#ifdef __cplusplus
+}
+#endif
+#endif
diff --git a/faad2/src/frontend/faad.man b/faad2/src/frontend/faad.man new file mode 100644 index 0000000..83727c7 --- /dev/null +++ b/faad2/src/frontend/faad.man @@ -0,0 +1,85 @@ +.TH FAAD "1" "October 2006" "faad 2.5" ""
+.SH NAME
+faad \(em Process an Advanced Audio Codec stream
+
+.SH "SYNOPSIS"
+.B faad
+[options] [\-w | \-o <output_filename> | \-a <output_filename>] input_filename
+
+.SH "DESCRIPTION"
+This utility provides a command line interface to libfaad2. This program reads in MPEG\(hy4 AAC files, processes, and outputs them in either Microsoft WAV, MPEG\(hy4 AAC ADTS, or standard PCM formats.
+
+.SH "OPTIONS"
+.TP
+.BI \-a " <filename>" ", \-\^\-adtsout" " <filename>"
+Sets the processing to output to the specified file in MPEG\(hy4 AAC ADTS format
+.TP
+.BI \-b " <number>" ", \-\^\-bits" " <number>"
+Set the output (individual) sample format. The number takes one of the following values:
+.RS
+.RS
+1: 16\(hybit PCM data (default).
+.br
+2: 24\(hybit PCM data.
+.br
+3: 32\(hybit PCM data.
+.br
+4: 32\(hybit floating\hy(point data.
+.br
+5: 64\(hybit floating\hy(point data.
+.RE
+.RE
+.TP
+.B \-d ", \-\^\-downmix"
+Set the processing to downsample from 5.1 (surround sound and bass) channels to 2 channels (stereo).
+.TP
+.BI \-f " <number>" ", \-\^\-format" " <number>"
+Set the output file format. The number takes one of the following values:
+.RS
+.RS
+1: Microsoft WAV format (default).
+.br
+2: Raw PCM data.
+.RE
+.RE
+.TP
+.BI \-g
+Set the processing to not perform gapless decoding.
+.TP
+.B \-h ", \-\^\-help"
+Shows a usage summary.
+.TP
+.B \-i ", \-\^\-info"
+Shows information about the about the input file.
+.TP
+.BI \-l " <number>" ", \-\^\-objecttype" " <number>"
+Sets the MPEG\hy(4 profile and object type for the processing to use. The number takes one of the following values:
+.RS
+.RS
+1: Main object type.
+.br
+2: Low Complexity (LC) object type (default).
+.br
+4: Long Term Prediction (LTP) object type.
+.br
+23: Low Delay (LD) object type.
+.RE
+.RE
+.TP
+.BI \-o " <filename>" ", \-\^\-outfile" " <number>"
+Sets the filename for processing output.
+.TP
+.B \-q ", \-\^\-quiet"
+Quiet \- Suppresses status messages during processing.
+.TP
+.B \-t ", \-\^\-oldformat"
+Sets the processing to use the old MPEG\(hy4 AAC ADTS format when outputting in said format.
+.TP
+.B \-w ", \-\^\-stdio"
+Sets the processing output to be sent to the standard out.
+
+.SH "AUTHOR"
+Matthew W. S. Bell <matthew (at) bells23.org.uk>
+
+.SH "SEE ALSO"
+\fBfaac\fP(1)
\ No newline at end of file diff --git a/faad2/src/frontend/faad.sln b/faad2/src/frontend/faad.sln new file mode 100644 index 0000000..f8f1596 --- /dev/null +++ b/faad2/src/frontend/faad.sln @@ -0,0 +1,36 @@ +
+Microsoft Visual Studio Solution File, Format Version 9.00
+# Visual Studio 2005
+Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "faad", "faad.vcproj", "{2BD8CBB3-DFC9-4A6A-9B7A-07ED749BED58}"
+ ProjectSection(ProjectDependencies) = postProject
+ {F470BB4A-7675-4D6A-B310-41F33AC6F987} = {F470BB4A-7675-4D6A-B310-41F33AC6F987}
+ {BC3EFE27-9015-4C9C-AD3C-72B3B7ED2114} = {BC3EFE27-9015-4C9C-AD3C-72B3B7ED2114}
+ EndProjectSection
+EndProject
+Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "libfaad", "..\libfaad\libfaad.vcproj", "{BC3EFE27-9015-4C9C-AD3C-72B3B7ED2114}"
+EndProject
+Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "mp4ff", "..\common\mp4ff\mp4ff.vcproj", "{F470BB4A-7675-4D6A-B310-41F33AC6F987}"
+EndProject
+Global
+ GlobalSection(SolutionConfigurationPlatforms) = preSolution
+ Debug|Win32 = Debug|Win32
+ Release|Win32 = Release|Win32
+ EndGlobalSection
+ GlobalSection(ProjectConfigurationPlatforms) = postSolution
+ {2BD8CBB3-DFC9-4A6A-9B7A-07ED749BED58}.Debug|Win32.ActiveCfg = Debug|Win32
+ {2BD8CBB3-DFC9-4A6A-9B7A-07ED749BED58}.Debug|Win32.Build.0 = Debug|Win32
+ {2BD8CBB3-DFC9-4A6A-9B7A-07ED749BED58}.Release|Win32.ActiveCfg = Release|Win32
+ {2BD8CBB3-DFC9-4A6A-9B7A-07ED749BED58}.Release|Win32.Build.0 = Release|Win32
+ {BC3EFE27-9015-4C9C-AD3C-72B3B7ED2114}.Debug|Win32.ActiveCfg = Debug|Win32
+ {BC3EFE27-9015-4C9C-AD3C-72B3B7ED2114}.Debug|Win32.Build.0 = Debug|Win32
+ {BC3EFE27-9015-4C9C-AD3C-72B3B7ED2114}.Release|Win32.ActiveCfg = Release|Win32
+ {BC3EFE27-9015-4C9C-AD3C-72B3B7ED2114}.Release|Win32.Build.0 = Release|Win32
+ {F470BB4A-7675-4D6A-B310-41F33AC6F987}.Debug|Win32.ActiveCfg = Debug|Win32
+ {F470BB4A-7675-4D6A-B310-41F33AC6F987}.Debug|Win32.Build.0 = Debug|Win32
+ {F470BB4A-7675-4D6A-B310-41F33AC6F987}.Release|Win32.ActiveCfg = Release|Win32
+ {F470BB4A-7675-4D6A-B310-41F33AC6F987}.Release|Win32.Build.0 = Release|Win32
+ EndGlobalSection
+ GlobalSection(SolutionProperties) = preSolution
+ HideSolutionNode = FALSE
+ EndGlobalSection
+EndGlobal
diff --git a/faad2/src/frontend/faad.vcproj b/faad2/src/frontend/faad.vcproj new file mode 100644 index 0000000..b33f6f4 --- /dev/null +++ b/faad2/src/frontend/faad.vcproj @@ -0,0 +1,256 @@ +<?xml version="1.0" encoding="Windows-1252"?>
+<VisualStudioProject
+ ProjectType="Visual C++"
+ Version="8.00"
+ Name="faad"
+ ProjectGUID="{2BD8CBB3-DFC9-4A6A-9B7A-07ED749BED58}"
+ >
+ <Platforms>
+ <Platform
+ Name="Win32"
+ />
+ </Platforms>
+ <ToolFiles>
+ </ToolFiles>
+ <Configurations>
+ <Configuration
+ Name="Debug|Win32"
+ OutputDirectory=".\Debug"
+ IntermediateDirectory=".\Debug"
+ ConfigurationType="1"
+ InheritedPropertySheets="$(VCInstallDir)VCProjectDefaults\UpgradeFromVC71.vsprops"
+ UseOfMFC="0"
+ ATLMinimizesCRunTimeLibraryUsage="false"
+ CharacterSet="2"
+ >
+ <Tool
+ Name="VCPreBuildEventTool"
+ />
+ <Tool
+ Name="VCCustomBuildTool"
+ />
+ <Tool
+ Name="VCXMLDataGeneratorTool"
+ />
+ <Tool
+ Name="VCWebServiceProxyGeneratorTool"
+ />
+ <Tool
+ Name="VCMIDLTool"
+ TypeLibraryName=".\Debug/faad.tlb"
+ />
+ <Tool
+ Name="VCCLCompilerTool"
+ AdditionalOptions=""
+ Optimization="0"
+ AdditionalIncludeDirectories="../include,../common/mp4ff,../common/faad"
+ PreprocessorDefinitions="WIN32,_DEBUG,_CONSOLE"
+ BasicRuntimeChecks="3"
+ RuntimeLibrary="3"
+ UsePrecompiledHeader="0"
+ PrecompiledHeaderFile=".\Debug/faad.pch"
+ AssemblerListingLocation=".\Debug/"
+ ObjectFile=".\Debug/"
+ ProgramDataBaseFileName=".\Debug/"
+ WarningLevel="3"
+ SuppressStartupBanner="true"
+ DebugInformationFormat="4"
+ CompileAs="0"
+ />
+ <Tool
+ Name="VCManagedResourceCompilerTool"
+ />
+ <Tool
+ Name="VCResourceCompilerTool"
+ PreprocessorDefinitions="_DEBUG"
+ Culture="1043"
+ />
+ <Tool
+ Name="VCPreLinkEventTool"
+ />
+ <Tool
+ Name="VCLinkerTool"
+ AdditionalOptions="/MACHINE:I386"
+ AdditionalDependencies="ws2_32.lib odbc32.lib odbccp32.lib Advapi32.lib"
+ OutputFile=".\Debug/faad.exe"
+ LinkIncremental="2"
+ SuppressStartupBanner="true"
+ GenerateDebugInformation="true"
+ ProgramDatabaseFile=".\Debug/faad.pdb"
+ SubSystem="1"
+ />
+ <Tool
+ Name="VCALinkTool"
+ />
+ <Tool
+ Name="VCManifestTool"
+ />
+ <Tool
+ Name="VCXDCMakeTool"
+ />
+ <Tool
+ Name="VCBscMakeTool"
+ />
+ <Tool
+ Name="VCFxCopTool"
+ />
+ <Tool
+ Name="VCAppVerifierTool"
+ />
+ <Tool
+ Name="VCWebDeploymentTool"
+ />
+ <Tool
+ Name="VCPostBuildEventTool"
+ />
+ </Configuration>
+ <Configuration
+ Name="Release|Win32"
+ OutputDirectory=".\Release"
+ IntermediateDirectory=".\Release"
+ ConfigurationType="1"
+ InheritedPropertySheets="$(VCInstallDir)VCProjectDefaults\UpgradeFromVC71.vsprops"
+ UseOfMFC="0"
+ ATLMinimizesCRunTimeLibraryUsage="false"
+ CharacterSet="2"
+ >
+ <Tool
+ Name="VCPreBuildEventTool"
+ />
+ <Tool
+ Name="VCCustomBuildTool"
+ />
+ <Tool
+ Name="VCXMLDataGeneratorTool"
+ />
+ <Tool
+ Name="VCWebServiceProxyGeneratorTool"
+ />
+ <Tool
+ Name="VCMIDLTool"
+ TypeLibraryName=".\Release/faad.tlb"
+ />
+ <Tool
+ Name="VCCLCompilerTool"
+ AdditionalOptions=""
+ Optimization="1"
+ InlineFunctionExpansion="1"
+ EnableIntrinsicFunctions="true"
+ FavorSizeOrSpeed="1"
+ AdditionalIncludeDirectories="../include,../common/mp4ff,../common/faad"
+ PreprocessorDefinitions="WIN32,NDEBUG,_CONSOLE"
+ StringPooling="true"
+ RuntimeLibrary="2"
+ EnableFunctionLevelLinking="true"
+ UsePrecompiledHeader="0"
+ PrecompiledHeaderFile=".\Release/faad.pch"
+ AssemblerListingLocation=".\Release/"
+ ObjectFile=".\Release/"
+ ProgramDataBaseFileName=".\Release/"
+ WarningLevel="3"
+ SuppressStartupBanner="true"
+ CompileAs="0"
+ />
+ <Tool
+ Name="VCManagedResourceCompilerTool"
+ />
+ <Tool
+ Name="VCResourceCompilerTool"
+ PreprocessorDefinitions="NDEBUG"
+ Culture="1043"
+ />
+ <Tool
+ Name="VCPreLinkEventTool"
+ />
+ <Tool
+ Name="VCLinkerTool"
+ AdditionalOptions="/MACHINE:I386"
+ AdditionalDependencies="ws2_32.lib Advapi32.lib"
+ OutputFile=".\Release/faad.exe"
+ LinkIncremental="1"
+ SuppressStartupBanner="true"
+ SubSystem="1"
+ />
+ <Tool
+ Name="VCALinkTool"
+ />
+ <Tool
+ Name="VCManifestTool"
+ />
+ <Tool
+ Name="VCXDCMakeTool"
+ />
+ <Tool
+ Name="VCBscMakeTool"
+ />
+ <Tool
+ Name="VCFxCopTool"
+ />
+ <Tool
+ Name="VCAppVerifierTool"
+ />
+ <Tool
+ Name="VCWebDeploymentTool"
+ />
+ <Tool
+ Name="VCPostBuildEventTool"
+ />
+ </Configuration>
+ </Configurations>
+ <References>
+ </References>
+ <Files>
+ <Filter
+ Name="Source Files"
+ Filter="cpp;c;cxx;rc;def;r;odl;idl;hpj;bat"
+ >
+ <File
+ RelativePath=".\audio.c"
+ >
+ </File>
+ <File
+ RelativePath="..\common\faad\getopt.c"
+ >
+ </File>
+ <File
+ RelativePath=".\main.c"
+ >
+ </File>
+ </Filter>
+ <Filter
+ Name="Header Files"
+ Filter="h;hpp;hxx;hm;inl"
+ >
+ <File
+ RelativePath=".\audio.h"
+ >
+ </File>
+ <File
+ RelativePath="..\common\faad\getopt.h"
+ >
+ </File>
+ <File
+ RelativePath="..\common\mp4v2\mp4.h"
+ >
+ </File>
+ <File
+ RelativePath="..\common\mp4v2\mpeg4ip.h"
+ >
+ </File>
+ <File
+ RelativePath="..\include\neaacdec.h"
+ >
+ </File>
+ <File
+ RelativePath="..\common\mp4v2\systems.h"
+ >
+ </File>
+ <File
+ RelativePath="..\common\mp4v2\win32_ver.h"
+ >
+ </File>
+ </Filter>
+ </Files>
+ <Globals>
+ </Globals>
+</VisualStudioProject>
diff --git a/faad2/src/frontend/main.c b/faad2/src/frontend/main.c new file mode 100644 index 0000000..2f97c07 --- /dev/null +++ b/faad2/src/frontend/main.c @@ -0,0 +1,1270 @@ +/*
+** 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: main.c,v 1.85 2008/09/22 17:55:09 menno Exp $
+**/
+
+#ifdef _WIN32
+#define WIN32_LEAN_AND_MEAN
+#include <windows.h>
+#define off_t __int64
+#else
+#include <time.h>
+#endif
+
+#include <fcntl.h>
+#include <stdio.h>
+#include <stdarg.h>
+#include <stdlib.h>
+#include <string.h>
+#include <getopt.h>
+
+#include <neaacdec.h>
+#include <mp4ff.h>
+
+#include "audio.h"
+
+#ifndef min
+#define min(a,b) ( (a) < (b) ? (a) : (b) )
+#endif
+
+#define MAX_CHANNELS 6 /* make this higher to support files with
+ more channels */
+
+
+static int quiet = 0;
+
+static void faad_fprintf(FILE *stream, const char *fmt, ...)
+{
+ va_list ap;
+
+ if (!quiet)
+ {
+ va_start(ap, fmt);
+
+ vfprintf(stream, fmt, ap);
+
+ va_end(ap);
+ }
+}
+
+/* FAAD file buffering routines */
+typedef struct {
+ long bytes_into_buffer;
+ long bytes_consumed;
+ long file_offset;
+ unsigned char *buffer;
+ int at_eof;
+ FILE *infile;
+} aac_buffer;
+
+
+static int fill_buffer(aac_buffer *b)
+{
+ int bread;
+
+ if (b->bytes_consumed > 0)
+ {
+ if (b->bytes_into_buffer)
+ {
+ memmove((void*)b->buffer, (void*)(b->buffer + b->bytes_consumed),
+ b->bytes_into_buffer*sizeof(unsigned char));
+ }
+
+ if (!b->at_eof)
+ {
+ bread = fread((void*)(b->buffer + b->bytes_into_buffer), 1,
+ b->bytes_consumed, b->infile);
+
+ if (bread != b->bytes_consumed)
+ b->at_eof = 1;
+
+ b->bytes_into_buffer += bread;
+ }
+
+ b->bytes_consumed = 0;
+
+ if (b->bytes_into_buffer > 3)
+ {
+ if (memcmp(b->buffer, "TAG", 3) == 0)
+ b->bytes_into_buffer = 0;
+ }
+ if (b->bytes_into_buffer > 11)
+ {
+ if (memcmp(b->buffer, "LYRICSBEGIN", 11) == 0)
+ b->bytes_into_buffer = 0;
+ }
+ if (b->bytes_into_buffer > 8)
+ {
+ if (memcmp(b->buffer, "APETAGEX", 8) == 0)
+ b->bytes_into_buffer = 0;
+ }
+ }
+
+ return 1;
+}
+
+static void advance_buffer(aac_buffer *b, int bytes)
+{
+ b->file_offset += bytes;
+ b->bytes_consumed = bytes;
+ b->bytes_into_buffer -= bytes;
+ if (b->bytes_into_buffer < 0)
+ b->bytes_into_buffer = 0;
+}
+
+static int adts_sample_rates[] = {96000,88200,64000,48000,44100,32000,24000,22050,16000,12000,11025,8000,7350,0,0,0};
+
+static int adts_parse(aac_buffer *b, int *bitrate, float *length)
+{
+ int frames, frame_length;
+ int t_framelength = 0;
+ int samplerate;
+ float frames_per_sec, bytes_per_frame;
+
+ /* Read all frames to ensure correct time and bitrate */
+ for (frames = 0; /* */; frames++)
+ {
+ fill_buffer(b);
+
+ if (b->bytes_into_buffer > 7)
+ {
+ /* check syncword */
+ if (!((b->buffer[0] == 0xFF)&&((b->buffer[1] & 0xF6) == 0xF0)))
+ break;
+
+ if (frames == 0)
+ samplerate = adts_sample_rates[(b->buffer[2]&0x3c)>>2];
+
+ frame_length = ((((unsigned int)b->buffer[3] & 0x3)) << 11)
+ | (((unsigned int)b->buffer[4]) << 3) | (b->buffer[5] >> 5);
+
+ t_framelength += frame_length;
+
+ if (frame_length > b->bytes_into_buffer)
+ break;
+
+ advance_buffer(b, frame_length);
+ } else {
+ break;
+ }
+ }
+
+ frames_per_sec = (float)samplerate/1024.0f;
+ if (frames != 0)
+ bytes_per_frame = (float)t_framelength/(float)(frames*1000);
+ else
+ bytes_per_frame = 0;
+ *bitrate = (int)(8. * bytes_per_frame * frames_per_sec + 0.5);
+ if (frames_per_sec != 0)
+ *length = (float)frames/frames_per_sec;
+ else
+ *length = 1;
+
+ return 1;
+}
+
+
+
+uint32_t read_callback(void *user_data, void *buffer, uint32_t length)
+{
+ return fread(buffer, 1, length, (FILE*)user_data);
+}
+
+uint32_t seek_callback(void *user_data, uint64_t position)
+{
+ return fseek((FILE*)user_data, position, SEEK_SET);
+}
+
+/* MicroSoft channel definitions */
+#define SPEAKER_FRONT_LEFT 0x1
+#define SPEAKER_FRONT_RIGHT 0x2
+#define SPEAKER_FRONT_CENTER 0x4
+#define SPEAKER_LOW_FREQUENCY 0x8
+#define SPEAKER_BACK_LEFT 0x10
+#define SPEAKER_BACK_RIGHT 0x20
+#define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
+#define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
+#define SPEAKER_BACK_CENTER 0x100
+#define SPEAKER_SIDE_LEFT 0x200
+#define SPEAKER_SIDE_RIGHT 0x400
+#define SPEAKER_TOP_CENTER 0x800
+#define SPEAKER_TOP_FRONT_LEFT 0x1000
+#define SPEAKER_TOP_FRONT_CENTER 0x2000
+#define SPEAKER_TOP_FRONT_RIGHT 0x4000
+#define SPEAKER_TOP_BACK_LEFT 0x8000
+#define SPEAKER_TOP_BACK_CENTER 0x10000
+#define SPEAKER_TOP_BACK_RIGHT 0x20000
+#define SPEAKER_RESERVED 0x80000000
+
+static long aacChannelConfig2wavexChannelMask(NeAACDecFrameInfo *hInfo)
+{
+ if (hInfo->channels == 6 && hInfo->num_lfe_channels)
+ {
+ return SPEAKER_FRONT_LEFT + SPEAKER_FRONT_RIGHT +
+ SPEAKER_FRONT_CENTER + SPEAKER_LOW_FREQUENCY +
+ SPEAKER_BACK_LEFT + SPEAKER_BACK_RIGHT;
+ } else {
+ return 0;
+ }
+}
+
+static char *position2string(int position)
+{
+ switch (position)
+ {
+ case FRONT_CHANNEL_CENTER: return "Center front";
+ case FRONT_CHANNEL_LEFT: return "Left front";
+ case FRONT_CHANNEL_RIGHT: return "Right front";
+ case SIDE_CHANNEL_LEFT: return "Left side";
+ case SIDE_CHANNEL_RIGHT: return "Right side";
+ case BACK_CHANNEL_LEFT: return "Left back";
+ case BACK_CHANNEL_RIGHT: return "Right back";
+ case BACK_CHANNEL_CENTER: return "Center back";
+ case LFE_CHANNEL: return "LFE";
+ case UNKNOWN_CHANNEL: return "Unknown";
+ default: return "";
+ }
+
+ return "";
+}
+
+static void print_channel_info(NeAACDecFrameInfo *frameInfo)
+{
+ /* print some channel info */
+ int i;
+ long channelMask = aacChannelConfig2wavexChannelMask(frameInfo);
+
+ faad_fprintf(stderr, " ---------------------\n");
+ if (frameInfo->num_lfe_channels > 0)
+ {
+ faad_fprintf(stderr, " | Config: %2d.%d Ch |", frameInfo->channels-frameInfo->num_lfe_channels, frameInfo->num_lfe_channels);
+ } else {
+ faad_fprintf(stderr, " | Config: %2d Ch |", frameInfo->channels);
+ }
+ if (channelMask)
+ faad_fprintf(stderr, " WARNING: channels are reordered according to\n");
+ else
+ faad_fprintf(stderr, "\n");
+ faad_fprintf(stderr, " ---------------------");
+ if (channelMask)
+ faad_fprintf(stderr, " MS defaults defined in WAVE_FORMAT_EXTENSIBLE\n");
+ else
+ faad_fprintf(stderr, "\n");
+ faad_fprintf(stderr, " | Ch | Position |\n");
+ faad_fprintf(stderr, " ---------------------\n");
+ for (i = 0; i < frameInfo->channels; i++)
+ {
+ faad_fprintf(stderr, " | %.2d | %-14s |\n", i, position2string((int)frameInfo->channel_position[i]));
+ }
+ faad_fprintf(stderr, " ---------------------\n");
+ faad_fprintf(stderr, "\n");
+}
+
+static int FindAdtsSRIndex(int sr)
+{
+ int i;
+
+ for (i = 0; i < 16; i++)
+ {
+ if (sr == adts_sample_rates[i])
+ return i;
+ }
+ return 16 - 1;
+}
+
+static unsigned char *MakeAdtsHeader(int *dataSize, NeAACDecFrameInfo *hInfo, int old_format)
+{
+ unsigned char *data;
+ int profile = (hInfo->object_type - 1) & 0x3;
+ int sr_index = ((hInfo->sbr == SBR_UPSAMPLED) || (hInfo->sbr == NO_SBR_UPSAMPLED)) ?
+ FindAdtsSRIndex(hInfo->samplerate / 2) : FindAdtsSRIndex(hInfo->samplerate);
+ int skip = (old_format) ? 8 : 7;
+ int framesize = skip + hInfo->bytesconsumed;
+
+ if (hInfo->header_type == ADTS)
+ framesize -= skip;
+
+ *dataSize = 7;
+
+ data = malloc(*dataSize * sizeof(unsigned char));
+ memset(data, 0, *dataSize * sizeof(unsigned char));
+
+ data[0] += 0xFF; /* 8b: syncword */
+
+ data[1] += 0xF0; /* 4b: syncword */
+ /* 1b: mpeg id = 0 */
+ /* 2b: layer = 0 */
+ data[1] += 1; /* 1b: protection absent */
+
+ data[2] += ((profile << 6) & 0xC0); /* 2b: profile */
+ data[2] += ((sr_index << 2) & 0x3C); /* 4b: sampling_frequency_index */
+ /* 1b: private = 0 */
+ data[2] += ((hInfo->channels >> 2) & 0x1); /* 1b: channel_configuration */
+
+ data[3] += ((hInfo->channels << 6) & 0xC0); /* 2b: channel_configuration */
+ /* 1b: original */
+ /* 1b: home */
+ /* 1b: copyright_id */
+ /* 1b: copyright_id_start */
+ data[3] += ((framesize >> 11) & 0x3); /* 2b: aac_frame_length */
+
+ data[4] += ((framesize >> 3) & 0xFF); /* 8b: aac_frame_length */
+
+ data[5] += ((framesize << 5) & 0xE0); /* 3b: aac_frame_length */
+ data[5] += ((0x7FF >> 6) & 0x1F); /* 5b: adts_buffer_fullness */
+
+ data[6] += ((0x7FF << 2) & 0x3F); /* 6b: adts_buffer_fullness */
+ /* 2b: num_raw_data_blocks */
+
+ return data;
+}
+
+/* globals */
+char *progName;
+
+static const char *file_ext[] =
+{
+ NULL,
+ ".wav",
+ ".aif",
+ ".au",
+ ".au",
+ ".pcm",
+ NULL
+};
+
+static void usage(void)
+{
+ faad_fprintf(stdout, "\nUsage:\n");
+ faad_fprintf(stdout, "%s [options] infile.aac\n", progName);
+ faad_fprintf(stdout, "Options:\n");
+ faad_fprintf(stdout, " -h Shows this help screen.\n");
+ faad_fprintf(stdout, " -i Shows info about the input file.\n");
+ faad_fprintf(stdout, " -a X Write MPEG-4 AAC ADTS output file.\n");
+ faad_fprintf(stdout, " -t Assume old ADTS format.\n");
+ faad_fprintf(stdout, " -o X Set output filename.\n");
+ faad_fprintf(stdout, " -f X Set output format. Valid values for X are:\n");
+ faad_fprintf(stdout, " 1: Microsoft WAV format (default).\n");
+ faad_fprintf(stdout, " 2: RAW PCM data.\n");
+ faad_fprintf(stdout, " -b X Set output sample format. Valid values for X are:\n");
+ faad_fprintf(stdout, " 1: 16 bit PCM data (default).\n");
+ faad_fprintf(stdout, " 2: 24 bit PCM data.\n");
+ faad_fprintf(stdout, " 3: 32 bit PCM data.\n");
+ faad_fprintf(stdout, " 4: 32 bit floating point data.\n");
+ faad_fprintf(stdout, " 5: 64 bit floating point data.\n");
+ faad_fprintf(stdout, " -s X Force the samplerate to X (for RAW files).\n");
+ faad_fprintf(stdout, " -l X Set object type. Supported object types:\n");
+ faad_fprintf(stdout, " 1: Main object type.\n");
+ faad_fprintf(stdout, " 2: LC (Low Complexity) object type.\n");
+ faad_fprintf(stdout, " 4: LTP (Long Term Prediction) object type.\n");
+ faad_fprintf(stdout, " 23: LD (Low Delay) object type.\n");
+ faad_fprintf(stdout, " -d Down matrix 5.1 to 2 channels\n");
+ faad_fprintf(stdout, " -w Write output to stdio instead of a file.\n");
+ faad_fprintf(stdout, " -g Disable gapless decoding.\n");
+ faad_fprintf(stdout, " -q Quiet - suppresses status messages.\n");
+ faad_fprintf(stdout, "Example:\n");
+ faad_fprintf(stdout, " %s infile.aac\n", progName);
+ faad_fprintf(stdout, " %s infile.mp4\n", progName);
+ faad_fprintf(stdout, " %s -o outfile.wav infile.aac\n", progName);
+ faad_fprintf(stdout, " %s -w infile.aac > outfile.wav\n", progName);
+ faad_fprintf(stdout, " %s -a outfile.aac infile.aac\n", progName);
+ return;
+}
+
+static int decodeAACfile(char *aacfile, char *sndfile, char *adts_fn, int to_stdout,
+ int def_srate, int object_type, int outputFormat, int fileType,
+ int downMatrix, int infoOnly, int adts_out, int old_format,
+ float *song_length)
+{
+ int tagsize;
+ unsigned long samplerate;
+ unsigned char channels;
+ void *sample_buffer;
+
+ audio_file *aufile;
+
+ FILE *adtsFile;
+ unsigned char *adtsData;
+ int adtsDataSize;
+
+ NeAACDecHandle hDecoder;
+ NeAACDecFrameInfo frameInfo;
+ NeAACDecConfigurationPtr config;
+
+ char percents[200];
+ int percent, old_percent = -1;
+ int bread, fileread;
+ int header_type = 0;
+ int bitrate = 0;
+ float length = 0;
+
+ int first_time = 1;
+
+ aac_buffer b;
+
+ memset(&b, 0, sizeof(aac_buffer));
+
+ if (adts_out)
+ {
+ adtsFile = fopen(adts_fn, "wb");
+ if (adtsFile == NULL)
+ {
+ faad_fprintf(stderr, "Error opening file: %s\n", adts_fn);
+ return 1;
+ }
+ }
+
+ b.infile = fopen(aacfile, "rb");
+ if (b.infile == NULL)
+ {
+ /* unable to open file */
+ faad_fprintf(stderr, "Error opening file: %s\n", aacfile);
+ return 1;
+ }
+
+ fseek(b.infile, 0, SEEK_END);
+ fileread = ftell(b.infile);
+ fseek(b.infile, 0, SEEK_SET);
+
+ if (!(b.buffer = (unsigned char*)malloc(FAAD_MIN_STREAMSIZE*MAX_CHANNELS)))
+ {
+ faad_fprintf(stderr, "Memory allocation error\n");
+ return 0;
+ }
+ memset(b.buffer, 0, FAAD_MIN_STREAMSIZE*MAX_CHANNELS);
+
+ bread = fread(b.buffer, 1, FAAD_MIN_STREAMSIZE*MAX_CHANNELS, b.infile);
+ b.bytes_into_buffer = bread;
+ b.bytes_consumed = 0;
+ b.file_offset = 0;
+
+ if (bread != FAAD_MIN_STREAMSIZE*MAX_CHANNELS)
+ b.at_eof = 1;
+
+ tagsize = 0;
+ if (!memcmp(b.buffer, "ID3", 3))
+ {
+ /* high bit is not used */
+ tagsize = (b.buffer[6] << 21) | (b.buffer[7] << 14) |
+ (b.buffer[8] << 7) | (b.buffer[9] << 0);
+
+ tagsize += 10;
+ advance_buffer(&b, tagsize);
+ fill_buffer(&b);
+ }
+
+ hDecoder = NeAACDecOpen();
+
+ /* Set the default object type and samplerate */
+ /* This is useful for RAW AAC files */
+ config = NeAACDecGetCurrentConfiguration(hDecoder);
+ if (def_srate)
+ config->defSampleRate = def_srate;
+ config->defObjectType = object_type;
+ config->outputFormat = outputFormat;
+ config->downMatrix = downMatrix;
+ config->useOldADTSFormat = old_format;
+ //config->dontUpSampleImplicitSBR = 1;
+ NeAACDecSetConfiguration(hDecoder, config);
+
+ /* get AAC infos for printing */
+ header_type = 0;
+ if ((b.buffer[0] == 0xFF) && ((b.buffer[1] & 0xF6) == 0xF0))
+ {
+ adts_parse(&b, &bitrate, &length);
+ fseek(b.infile, tagsize, SEEK_SET);
+
+ bread = fread(b.buffer, 1, FAAD_MIN_STREAMSIZE*MAX_CHANNELS, b.infile);
+ if (bread != FAAD_MIN_STREAMSIZE*MAX_CHANNELS)
+ b.at_eof = 1;
+ else
+ b.at_eof = 0;
+ b.bytes_into_buffer = bread;
+ b.bytes_consumed = 0;
+ b.file_offset = tagsize;
+
+ header_type = 1;
+ } else if (memcmp(b.buffer, "ADIF", 4) == 0) {
+ int skip_size = (b.buffer[4] & 0x80) ? 9 : 0;
+ bitrate = ((unsigned int)(b.buffer[4 + skip_size] & 0x0F)<<19) |
+ ((unsigned int)b.buffer[5 + skip_size]<<11) |
+ ((unsigned int)b.buffer[6 + skip_size]<<3) |
+ ((unsigned int)b.buffer[7 + skip_size] & 0xE0);
+
+ length = (float)fileread;
+ if (length != 0)
+ {
+ length = ((float)length*8.f)/((float)bitrate) + 0.5f;
+ }
+
+ bitrate = (int)((float)bitrate/1000.0f + 0.5f);
+
+ header_type = 2;
+ }
+
+ *song_length = length;
+
+ fill_buffer(&b);
+ if ((bread = NeAACDecInit(hDecoder, b.buffer,
+ b.bytes_into_buffer, &samplerate, &channels)) < 0)
+ {
+ /* If some error initializing occured, skip the file */
+ faad_fprintf(stderr, "Error initializing decoder library.\n");
+ if (b.buffer)
+ free(b.buffer);
+ NeAACDecClose(hDecoder);
+ fclose(b.infile);
+ return 1;
+ }
+ advance_buffer(&b, bread);
+ fill_buffer(&b);
+
+ /* print AAC file info */
+ faad_fprintf(stderr, "%s file info:\n", aacfile);
+ switch (header_type)
+ {
+ case 0:
+ faad_fprintf(stderr, "RAW\n\n");
+ break;
+ case 1:
+ faad_fprintf(stderr, "ADTS, %.3f sec, %d kbps, %d Hz\n\n",
+ length, bitrate, samplerate);
+ break;
+ case 2:
+ faad_fprintf(stderr, "ADIF, %.3f sec, %d kbps, %d Hz\n\n",
+ length, bitrate, samplerate);
+ break;
+ }
+
+ if (infoOnly)
+ {
+ NeAACDecClose(hDecoder);
+ fclose(b.infile);
+ if (b.buffer)
+ free(b.buffer);
+ return 0;
+ }
+
+ do
+ {
+ sample_buffer = NeAACDecDecode(hDecoder, &frameInfo,
+ b.buffer, b.bytes_into_buffer);
+
+ if (adts_out == 1)
+ {
+ int skip = (old_format) ? 8 : 7;
+ adtsData = MakeAdtsHeader(&adtsDataSize, &frameInfo, old_format);
+
+ /* write the adts header */
+ fwrite(adtsData, 1, adtsDataSize, adtsFile);
+
+ /* write the frame data */
+ if (frameInfo.header_type == ADTS)
+ fwrite(b.buffer + skip, 1, frameInfo.bytesconsumed - skip, adtsFile);
+ else
+ fwrite(b.buffer, 1, frameInfo.bytesconsumed, adtsFile);
+ }
+
+ /* update buffer indices */
+ advance_buffer(&b, frameInfo.bytesconsumed);
+
+ if (frameInfo.error > 0)
+ {
+ faad_fprintf(stderr, "Error: %s\n",
+ NeAACDecGetErrorMessage(frameInfo.error));
+ }
+
+ /* open the sound file now that the number of channels are known */
+ if (first_time && !frameInfo.error)
+ {
+ /* print some channel info */
+ print_channel_info(&frameInfo);
+
+ if (!adts_out)
+ {
+ /* open output file */
+ if (!to_stdout)
+ {
+ aufile = open_audio_file(sndfile, frameInfo.samplerate, frameInfo.channels,
+ outputFormat, fileType, aacChannelConfig2wavexChannelMask(&frameInfo));
+ } else {
+ aufile = open_audio_file("-", frameInfo.samplerate, frameInfo.channels,
+ outputFormat, fileType, aacChannelConfig2wavexChannelMask(&frameInfo));
+ }
+ if (aufile == NULL)
+ {
+ if (b.buffer)
+ free(b.buffer);
+ NeAACDecClose(hDecoder);
+ fclose(b.infile);
+ return 0;
+ }
+ } else {
+ faad_fprintf(stderr, "Writing output MPEG-4 AAC ADTS file.\n\n");
+ }
+ first_time = 0;
+ }
+
+ percent = min((int)(b.file_offset*100)/fileread, 100);
+ if (percent > old_percent)
+ {
+ old_percent = percent;
+ sprintf(percents, "%d%% decoding %s.", percent, aacfile);
+ faad_fprintf(stderr, "%s\r", percents);
+#ifdef _WIN32
+ SetConsoleTitle(percents);
+#endif
+ }
+
+ if ((frameInfo.error == 0) && (frameInfo.samples > 0) && (!adts_out))
+ {
+ if (write_audio_file(aufile, sample_buffer, frameInfo.samples, 0) == 0)
+ break;
+ }
+
+ /* fill buffer */
+ fill_buffer(&b);
+
+ if (b.bytes_into_buffer == 0)
+ sample_buffer = NULL; /* to make sure it stops now */
+
+ } while (sample_buffer != NULL);
+
+ NeAACDecClose(hDecoder);
+
+ if (adts_out == 1)
+ {
+ fclose(adtsFile);
+ }
+
+ fclose(b.infile);
+
+ if (!first_time && !adts_out)
+ close_audio_file(aufile);
+
+ if (b.buffer)
+ free(b.buffer);
+
+ return frameInfo.error;
+}
+
+static int GetAACTrack(mp4ff_t *infile)
+{
+ /* find AAC track */
+ int i, rc;
+ int numTracks = mp4ff_total_tracks(infile);
+
+ for (i = 0; i < numTracks; i++)
+ {
+ unsigned char *buff = NULL;
+ int buff_size = 0;
+ mp4AudioSpecificConfig mp4ASC;
+
+ mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
+
+ if (buff)
+ {
+ rc = NeAACDecAudioSpecificConfig(buff, buff_size, &mp4ASC);
+ free(buff);
+
+ if (rc < 0)
+ continue;
+ return i;
+ }
+ }
+
+ /* can't decode this */
+ return -1;
+}
+
+static const unsigned long srates[] =
+{
+ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000,
+ 12000, 11025, 8000
+};
+
+static int decodeMP4file(char *mp4file, char *sndfile, char *adts_fn, int to_stdout,
+ int outputFormat, int fileType, int downMatrix, int noGapless,
+ int infoOnly, int adts_out, float *song_length)
+{
+ int track;
+ unsigned long samplerate;
+ unsigned char channels;
+ void *sample_buffer;
+
+ mp4ff_t *infile;
+ long sampleId, numSamples;
+
+ audio_file *aufile;
+
+ FILE *mp4File;
+ FILE *adtsFile;
+ unsigned char *adtsData;
+ int adtsDataSize;
+
+ NeAACDecHandle hDecoder;
+ NeAACDecConfigurationPtr config;
+ NeAACDecFrameInfo frameInfo;
+ mp4AudioSpecificConfig mp4ASC;
+
+ unsigned char *buffer;
+ int buffer_size;
+
+ char percents[200];
+ int percent, old_percent = -1;
+
+ int first_time = 1;
+
+ /* for gapless decoding */
+ unsigned int useAacLength = 1;
+ unsigned int initial = 1;
+ unsigned int framesize;
+ unsigned long timescale;
+
+
+ /* initialise the callback structure */
+ mp4ff_callback_t *mp4cb = malloc(sizeof(mp4ff_callback_t));
+
+ mp4File = fopen(mp4file, "rb");
+ mp4cb->read = read_callback;
+ mp4cb->seek = seek_callback;
+ mp4cb->user_data = mp4File;
+
+
+ hDecoder = NeAACDecOpen();
+
+ /* Set configuration */
+ config = NeAACDecGetCurrentConfiguration(hDecoder);
+ config->outputFormat = outputFormat;
+ config->downMatrix = downMatrix;
+ //config->dontUpSampleImplicitSBR = 1;
+ NeAACDecSetConfiguration(hDecoder, config);
+
+ if (adts_out)
+ {
+ adtsFile = fopen(adts_fn, "wb");
+ if (adtsFile == NULL)
+ {
+ faad_fprintf(stderr, "Error opening file: %s\n", adts_fn);
+ return 1;
+ }
+ }
+
+ infile = mp4ff_open_read(mp4cb);
+ if (!infile)
+ {
+ /* unable to open file */
+ faad_fprintf(stderr, "Error opening file: %s\n", mp4file);
+ return 1;
+ }
+
+ if ((track = GetAACTrack(infile)) < 0)
+ {
+ faad_fprintf(stderr, "Unable to find correct AAC sound track in the MP4 file.\n");
+ NeAACDecClose(hDecoder);
+ mp4ff_close(infile);
+ free(mp4cb);
+ fclose(mp4File);
+ return 1;
+ }
+
+ buffer = NULL;
+ buffer_size = 0;
+ mp4ff_get_decoder_config(infile, track, &buffer, &buffer_size);
+
+ if(NeAACDecInit2(hDecoder, buffer, buffer_size,
+ &samplerate, &channels) < 0)
+ {
+ /* If some error initializing occured, skip the file */
+ faad_fprintf(stderr, "Error initializing decoder library.\n");
+ NeAACDecClose(hDecoder);
+ mp4ff_close(infile);
+ free(mp4cb);
+ fclose(mp4File);
+ return 1;
+ }
+
+ timescale = mp4ff_time_scale(infile, track);
+ framesize = 1024;
+ useAacLength = 0;
+
+ if (buffer)
+ {
+ if (NeAACDecAudioSpecificConfig(buffer, buffer_size, &mp4ASC) >= 0)
+ {
+ if (mp4ASC.frameLengthFlag == 1) framesize = 960;
+ if (mp4ASC.sbr_present_flag == 1) framesize *= 2;
+ }
+ free(buffer);
+ }
+
+ /* print some mp4 file info */
+ faad_fprintf(stderr, "%s file info:\n\n", mp4file);
+ {
+ char *tag = NULL, *item = NULL;
+ int k, j;
+ char *ot[6] = { "NULL", "MAIN AAC", "LC AAC", "SSR AAC", "LTP AAC", "HE AAC" };
+ long samples = mp4ff_num_samples(infile, track);
+ float f = 1024.0;
+ float seconds;
+ if (mp4ASC.sbr_present_flag == 1)
+ {
+ f = f * 2.0;
+ }
+ seconds = (float)samples*(float)(f-1.0)/(float)mp4ASC.samplingFrequency;
+
+ *song_length = seconds;
+
+ faad_fprintf(stderr, "%s\t%.3f secs, %d ch, %d Hz\n\n", ot[(mp4ASC.objectTypeIndex > 5)?0:mp4ASC.objectTypeIndex],
+ seconds, mp4ASC.channelsConfiguration, mp4ASC.samplingFrequency);
+
+#define PRINT_MP4_METADATA
+#ifdef PRINT_MP4_METADATA
+ j = mp4ff_meta_get_num_items(infile);
+ for (k = 0; k < j; k++)
+ {
+ if (mp4ff_meta_get_by_index(infile, k, &item, &tag))
+ {
+ if (item != NULL && tag != NULL)
+ {
+ faad_fprintf(stderr, "%s: %s\n", item, tag);
+ free(item); item = NULL;
+ free(tag); tag = NULL;
+ }
+ }
+ }
+ if (j > 0) faad_fprintf(stderr, "\n");
+#endif
+ }
+
+ if (infoOnly)
+ {
+ NeAACDecClose(hDecoder);
+ mp4ff_close(infile);
+ free(mp4cb);
+ fclose(mp4File);
+ return 0;
+ }
+
+ numSamples = mp4ff_num_samples(infile, track);
+
+ for (sampleId = 0; sampleId < numSamples; sampleId++)
+ {
+ int rc;
+ long dur;
+ unsigned int sample_count;
+ unsigned int delay = 0;
+
+ /* get acces unit from MP4 file */
+ buffer = NULL;
+ buffer_size = 0;
+
+ dur = mp4ff_get_sample_duration(infile, track, sampleId);
+ rc = mp4ff_read_sample(infile, track, sampleId, &buffer, &buffer_size);
+ if (rc == 0)
+ {
+ faad_fprintf(stderr, "Reading from MP4 file failed.\n");
+ NeAACDecClose(hDecoder);
+ mp4ff_close(infile);
+ free(mp4cb);
+ fclose(mp4File);
+ return 1;
+ }
+
+ sample_buffer = NeAACDecDecode(hDecoder, &frameInfo, buffer, buffer_size);
+
+ if (adts_out == 1)
+ {
+ adtsData = MakeAdtsHeader(&adtsDataSize, &frameInfo, 0);
+
+ /* write the adts header */
+ fwrite(adtsData, 1, adtsDataSize, adtsFile);
+
+ fwrite(buffer, 1, frameInfo.bytesconsumed, adtsFile);
+ }
+
+ if (buffer) free(buffer);
+
+ if (!noGapless)
+ {
+ if (sampleId == 0) dur = 0;
+
+ if (useAacLength || (timescale != samplerate)) {
+ sample_count = frameInfo.samples;
+ } else {
+ sample_count = (unsigned int)(dur * frameInfo.channels);
+ if (sample_count > frameInfo.samples)
+ sample_count = frameInfo.samples;
+
+ if (!useAacLength && !initial && (sampleId < numSamples/2) && (sample_count != frameInfo.samples))
+ {
+ faad_fprintf(stderr, "MP4 seems to have incorrect frame duration, using values from AAC data.\n");
+ useAacLength = 1;
+ sample_count = frameInfo.samples;
+ }
+ }
+
+ if (initial && (sample_count < framesize*frameInfo.channels) && (frameInfo.samples > sample_count))
+ delay = frameInfo.samples - sample_count;
+ } else {
+ sample_count = frameInfo.samples;
+ }
+
+ /* open the sound file now that the number of channels are known */
+ if (first_time && !frameInfo.error)
+ {
+ /* print some channel info */
+ print_channel_info(&frameInfo);
+
+ if (!adts_out)
+ {
+ /* open output file */
+ if(!to_stdout)
+ {
+ aufile = open_audio_file(sndfile, frameInfo.samplerate, frameInfo.channels,
+ outputFormat, fileType, aacChannelConfig2wavexChannelMask(&frameInfo));
+ } else {
+#ifdef _WIN32
+ setmode(fileno(stdout), O_BINARY);
+#endif
+ aufile = open_audio_file("-", frameInfo.samplerate, frameInfo.channels,
+ outputFormat, fileType, aacChannelConfig2wavexChannelMask(&frameInfo));
+ }
+ if (aufile == NULL)
+ {
+ NeAACDecClose(hDecoder);
+ mp4ff_close(infile);
+ free(mp4cb);
+ fclose(mp4File);
+ return 0;
+ }
+ }
+ first_time = 0;
+ }
+
+ if (sample_count > 0) initial = 0;
+
+ percent = min((int)(sampleId*100)/numSamples, 100);
+ if (percent > old_percent)
+ {
+ old_percent = percent;
+ sprintf(percents, "%d%% decoding %s.", percent, mp4file);
+ faad_fprintf(stderr, "%s\r", percents);
+#ifdef _WIN32
+ SetConsoleTitle(percents);
+#endif
+ }
+
+ if ((frameInfo.error == 0) && (sample_count > 0) && (!adts_out))
+ {
+ if (write_audio_file(aufile, sample_buffer, sample_count, delay) == 0)
+ break;
+ }
+
+ if (frameInfo.error > 0)
+ {
+ faad_fprintf(stderr, "Warning: %s\n",
+ NeAACDecGetErrorMessage(frameInfo.error));
+ }
+ }
+
+ NeAACDecClose(hDecoder);
+
+ if (adts_out == 1)
+ {
+ fclose(adtsFile);
+ }
+
+ mp4ff_close(infile);
+
+ if (!first_time && !adts_out)
+ close_audio_file(aufile);
+
+ free(mp4cb);
+ fclose(mp4File);
+
+ return frameInfo.error;
+}
+
+int main(int argc, char *argv[])
+{
+ int result;
+ int infoOnly = 0;
+ int writeToStdio = 0;
+ int object_type = LC;
+ int def_srate = 0;
+ int downMatrix = 0;
+ int format = 1;
+ int outputFormat = FAAD_FMT_16BIT;
+ int outfile_set = 0;
+ int adts_out = 0;
+ int old_format = 0;
+ int showHelp = 0;
+ int mp4file = 0;
+ int noGapless = 0;
+ char *fnp;
+ char aacFileName[255];
+ char audioFileName[255];
+ char adtsFileName[255];
+ unsigned char header[8];
+ float length = 0;
+ FILE *hMP4File;
+
+/* System dependant types */
+#ifdef _WIN32
+ long begin;
+#else
+ clock_t begin;
+#endif
+
+ unsigned long cap = NeAACDecGetCapabilities();
+
+
+ /* begin process command line */
+ progName = argv[0];
+ while (1) {
+ int c = -1;
+ int option_index = 0;
+ static struct option long_options[] = {
+ { "quiet", 0, 0, 'q' },
+ { "outfile", 0, 0, 'o' },
+ { "adtsout", 0, 0, 'a' },
+ { "oldformat", 0, 0, 't' },
+ { "format", 0, 0, 'f' },
+ { "bits", 0, 0, 'b' },
+ { "samplerate", 0, 0, 's' },
+ { "objecttype", 0, 0, 'l' },
+ { "downmix", 0, 0, 'd' },
+ { "info", 0, 0, 'i' },
+ { "stdio", 0, 0, 'w' },
+ { "stdio", 0, 0, 'g' },
+ { "help", 0, 0, 'h' },
+ { 0, 0, 0, 0 }
+ };
+
+ c = getopt_long(argc, argv, "o:a:s:f:b:l:wgdhitq",
+ long_options, &option_index);
+
+ if (c == -1)
+ break;
+
+ switch (c) {
+ case 'o':
+ if (optarg)
+ {
+ outfile_set = 1;
+ strcpy(audioFileName, optarg);
+ }
+ break;
+ case 'a':
+ if (optarg)
+ {
+ adts_out = 1;
+ strcpy(adtsFileName, optarg);
+ }
+ break;
+ case 's':
+ if (optarg)
+ {
+ char dr[10];
+ if (sscanf(optarg, "%s", dr) < 1) {
+ def_srate = 0;
+ } else {
+ def_srate = atoi(dr);
+ }
+ }
+ break;
+ case 'f':
+ if (optarg)
+ {
+ char dr[10];
+ if (sscanf(optarg, "%s", dr) < 1)
+ {
+ format = 1;
+ } else {
+ format = atoi(dr);
+ if ((format < 1) || (format > 2))
+ showHelp = 1;
+ }
+ }
+ break;
+ case 'b':
+ if (optarg)
+ {
+ char dr[10];
+ if (sscanf(optarg, "%s", dr) < 1)
+ {
+ outputFormat = FAAD_FMT_16BIT; /* just use default */
+ } else {
+ outputFormat = atoi(dr);
+ if ((outputFormat < 1) || (outputFormat > 5))
+ showHelp = 1;
+ }
+ }
+ break;
+ case 'l':
+ if (optarg)
+ {
+ char dr[10];
+ if (sscanf(optarg, "%s", dr) < 1)
+ {
+ object_type = LC; /* default */
+ } else {
+ object_type = atoi(dr);
+ if ((object_type != LC) &&
+ (object_type != MAIN) &&
+ (object_type != LTP) &&
+ (object_type != LD))
+ {
+ showHelp = 1;
+ }
+ }
+ }
+ break;
+ case 't':
+ old_format = 1;
+ break;
+ case 'd':
+ downMatrix = 1;
+ break;
+ case 'w':
+ writeToStdio = 1;
+ break;
+ case 'g':
+ noGapless = 1;
+ break;
+ case 'i':
+ infoOnly = 1;
+ break;
+ case 'h':
+ showHelp = 1;
+ break;
+ case 'q':
+ quiet = 1;
+ break;
+ default:
+ break;
+ }
+ }
+
+
+ faad_fprintf(stderr, " *********** Ahead Software MPEG-4 AAC Decoder V%s ******************\n\n", FAAD2_VERSION);
+ faad_fprintf(stderr, " Build: %s\n", __DATE__);
+ faad_fprintf(stderr, " Copyright 2002-2004: Ahead Software AG\n");
+ faad_fprintf(stderr, " http://www.audiocoding.com\n");
+ if (cap & FIXED_POINT_CAP)
+ faad_fprintf(stderr, " Fixed point version\n");
+ else
+ faad_fprintf(stderr, " Floating point version\n");
+ faad_fprintf(stderr, "\n");
+ faad_fprintf(stderr, " This program is free software; you can redistribute it and/or modify\n");
+ faad_fprintf(stderr, " it under the terms of the GNU General Public License.\n");
+ faad_fprintf(stderr, "\n");
+ faad_fprintf(stderr, " **************************************************************************\n\n");
+
+
+ /* check that we have at least two non-option arguments */
+ /* Print help if requested */
+ if (((argc - optind) < 1) || showHelp)
+ {
+ usage();
+ return 1;
+ }
+
+#if 0
+ /* only allow raw data on stdio */
+ if (writeToStdio == 1)
+ {
+ format = 2;
+ }
+#endif
+
+ /* point to the specified file name */
+ strcpy(aacFileName, argv[optind]);
+
+#ifdef _WIN32
+ begin = GetTickCount();
+#else
+ begin = clock();
+#endif
+
+ /* Only calculate the path and open the file for writing if
+ we are not writing to stdout.
+ */
+ if(!writeToStdio && !outfile_set)
+ {
+ strcpy(audioFileName, aacFileName);
+
+ fnp = (char *)strrchr(audioFileName,'.');
+
+ if (fnp)
+ fnp[0] = '\0';
+
+ strcat(audioFileName, file_ext[format]);
+ }
+
+ /* check for mp4 file */
+ mp4file = 0;
+ hMP4File = fopen(aacFileName, "rb");
+ if (!hMP4File)
+ {
+ faad_fprintf(stderr, "Error opening file: %s\n", aacFileName);
+ return 1;
+ }
+ fread(header, 1, 8, hMP4File);
+ fclose(hMP4File);
+ if (header[4] == 'f' && header[5] == 't' && header[6] == 'y' && header[7] == 'p')
+ mp4file = 1;
+
+ if (mp4file)
+ {
+ result = decodeMP4file(aacFileName, audioFileName, adtsFileName, writeToStdio,
+ outputFormat, format, downMatrix, noGapless, infoOnly, adts_out, &length);
+ } else {
+ result = decodeAACfile(aacFileName, audioFileName, adtsFileName, writeToStdio,
+ def_srate, object_type, outputFormat, format, downMatrix, infoOnly, adts_out,
+ old_format, &length);
+ }
+
+ if (!result && !infoOnly)
+ {
+#ifdef _WIN32
+ float dec_length = (float)(GetTickCount()-begin)/1000.0;
+ SetConsoleTitle("FAAD");
+#else
+ /* clock() grabs time since the start of the app but when we decode
+ multiple files, each file has its own starttime (begin).
+ */
+ float dec_length = (float)(clock() - begin)/(float)CLOCKS_PER_SEC;
+#endif
+ faad_fprintf(stderr, "Decoding %s took: %5.2f sec. %5.2fx real-time.\n", aacFileName,
+ dec_length, length/dec_length);
+ }
+
+ return 0;
+}
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