From cc2160de5cc05dc3aa77f3a34358e66f6193c8c9 Mon Sep 17 00:00:00 2001 From: Michał Cichoń Date: Tue, 26 Jun 2012 20:35:30 +0200 Subject: Add support for AAC. --- faad2/src/frontend/audio.c | 500 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 500 insertions(+) create mode 100644 faad2/src/frontend/audio.c (limited to 'faad2/src/frontend/audio.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 +#endif +#include +#include +#include +#include +#include +#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; +} -- cgit v1.2.3