From a68df043bfbc7f8f38332143577877846631eca4 Mon Sep 17 00:00:00 2001 From: Michał Cichoń Date: Tue, 25 Aug 2015 19:58:37 +0200 Subject: Update build environment - remove faad2 - remove mad - remove polarssl - remove pthreads - add libcurl - add vtparse with UTF8 support - update project to use Visual Studio 2015 --- faad2/src/aacDECdrop/audio.c | 494 ------------------------------------------- 1 file changed, 494 deletions(-) delete mode 100644 faad2/src/aacDECdrop/audio.c (limited to 'faad2/src/aacDECdrop/audio.c') diff --git a/faad2/src/aacDECdrop/audio.c b/faad2/src/aacDECdrop/audio.c deleted file mode 100644 index 843d650..0000000 --- a/faad2/src/aacDECdrop/audio.c +++ /dev/null @@ -1,494 +0,0 @@ -/* -** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding -** Copyright (C) 2003-2004 M. Bakker, Ahead Software 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. -** -** Commercial non-GPL licensing of this software is possible. -** For more info contact Ahead Software through Mpeg4AAClicense@nero.com. -** -** $Id: audio.c,v 1.11 2004/02/06 10:23:27 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; - } else { - 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) - { - fseek(aufile->sndfile, 0, SEEK_SET); - - if (aufile->channelMask) - write_wav_extensible_header(aufile, aufile->channelMask); - else - write_wav_header(aufile); - } - - fclose(aufile->sndfile); - - if (aufile) free(aufile); -} - -static int write_wav_header(audio_file *aufile) -{ - unsigned char header[44]; - unsigned char* p = header; - unsigned int bytes = (aufile->bits_per_sample + 7) / 8; - float data_size = (float)bytes * aufile->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