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, 0 insertions, 2234 deletions
diff --git a/faad2/src/frontend/Makefile.am b/faad2/src/frontend/Makefile.am deleted file mode 100644 index 8bda787..0000000 --- a/faad2/src/frontend/Makefile.am +++ /dev/null @@ -1,12 +0,0 @@ -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 deleted file mode 100644 index 7691ce9..0000000 --- a/faad2/src/frontend/audio.c +++ /dev/null @@ -1,500 +0,0 @@ -/*
-** 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 deleted file mode 100644 index b4d3a67..0000000 --- a/faad2/src/frontend/audio.h +++ /dev/null @@ -1,75 +0,0 @@ -/*
-** 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 deleted file mode 100644 index 83727c7..0000000 --- a/faad2/src/frontend/faad.man +++ /dev/null @@ -1,85 +0,0 @@ -.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 deleted file mode 100644 index f8f1596..0000000 --- a/faad2/src/frontend/faad.sln +++ /dev/null @@ -1,36 +0,0 @@ -
-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 deleted file mode 100644 index b33f6f4..0000000 --- a/faad2/src/frontend/faad.vcproj +++ /dev/null @@ -1,256 +0,0 @@ -<?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 deleted file mode 100644 index 2f97c07..0000000 --- a/faad2/src/frontend/main.c +++ /dev/null @@ -1,1270 +0,0 @@ -/*
-** 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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