ref: 7012e484046d739bd07d0cb36f40fc17375abf59
dir: /frontend/main.c/
/*
* FAAC - Freeware Advanced Audio Coder
* Copyright (C) 2001 Menno Bakker
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library 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
* Lesser General Public License for more details.
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* $Id: main.c,v 1.33 2003/04/13 08:39:28 knik Exp $
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#ifdef WIN32
#include <windows.h>
#include <fcntl.h>
#endif
#ifdef __unix__
#include <sys/time.h>
#include <sys/resource.h>
#include <unistd.h>
#endif
#include <stdio.h>
#include <stdlib.h>
#include <ctype.h>
#include <math.h>
#ifdef HAVE_GETOPT_H
# include <getopt.h>
#else
# include "getopt.h"
# include "getopt.c"
#endif
#include <faac.h>
#include "input.h"
#ifndef min
#define min(a,b) ( (a) < (b) ? (a) : (b) )
#endif
/* globals */
char* progName;
int StringCompI(char const *str1, char const *str2, unsigned long len)
{
signed int c1 = 0, c2 = 0;
while (len--) {
c1 = tolower(*str1++);
c2 = tolower(*str2++);
if (c1 == 0 || c1 != c2)
break;
}
return c1 - c2;
}
int main(int argc, char *argv[])
{
int frames, currentFrame;
faacEncHandle hEncoder;
pcmfile_t *infile = NULL;
unsigned int sr, chan;
unsigned long samplesInput, maxBytesOutput;
faacEncConfigurationPtr myFormat;
unsigned int mpegVersion = MPEG2;
unsigned int objectType = LOW;
unsigned int useMidSide = 1;
static unsigned int useTns = 1;
unsigned int useAdts = 1;
int cutOff = -1;
unsigned long quantqual = 0;
char *audioFileName;
char *aacFileName;
short *pcmbuf;
unsigned char *bitbuf;
int bytesInput = 0;
int rawInput = 0;
int dieUsage = 0;
FILE *outfile;
fprintf(stderr, "FAAC - Freeware Advanced Audio Coder\n");
// get faac version
hEncoder = faacEncOpen(44100, 2, &samplesInput, &maxBytesOutput);
myFormat = faacEncGetCurrentConfiguration(hEncoder);
if (myFormat->version == FAAC_CFG_VERSION)
{
fprintf(stderr, "libfaac version %s\n", myFormat->name);
faacEncClose(hEncoder);
}
else
{
fprintf(stderr, __FILE__ "(%d): wrong libfaac version\n", __LINE__);
faacEncClose(hEncoder);
return 1;
}
/* begin process command line */
progName = argv[0];
while (1) {
int c = -1;
int option_index = 0;
static struct option long_options[] = {
{ "mpeg", 0, 0, 'm' },
{ "objecttype", 0, 0, 'o' },
{ "raw", 0, 0, 'r' },
{ "nomidside", 0, 0, 'n' },
{ "notns", 0, &useTns, 0 },
{ "cutoff", 1, 0, 'c' },
{ "quality", 1, 0, 'q' },
{ "pcmraw", 0, 0, 'P'},
{ "pcmsamplerate", 1, 0, 'R'},
{ "pcmsamplebits", 1, 0, 'B'},
{ "pcmchannels", 1, 0, 'C'},
{ 0, 0, 0, 0}
};
c = getopt_long(argc, argv, "m:o:rnc:q:PR:B:C:",
long_options, &option_index);
if (c == -1)
break;
if (!c)
continue;
switch (c) {
case 'm': {
unsigned int i;
if (optarg) {
if (sscanf(optarg, "%u", &i) < 1) {
mpegVersion = MPEG2;
} else {
if (i == 2)
mpegVersion = MPEG2;
else
mpegVersion = MPEG4;
}
} else {
mpegVersion = MPEG2;
}
break;
}
case 'o': {
unsigned char i[10];
if (optarg) {
if (sscanf(optarg, "%s", i) < 1) {
objectType = LOW;
} else {
if (StringCompI(i, "MAIN", 4) == 0)
objectType = MAIN;
else if (StringCompI(i, "LTP", 3) == 0)
objectType = LTP;
else
objectType = LOW;
}
} else {
objectType = LOW;
}
break;
}
case 'r': {
useAdts = 0;
break;
}
case 'n': {
useMidSide = 0;
break;
}
case 'c': {
unsigned int i;
if (sscanf(optarg, "%u", &i) > 0) {
cutOff = i;
}
break;
}
case 'q':
{
unsigned int i;
if (sscanf(optarg, "%u", &i) > 0)
{
if (i > 0 && i < 1000)
quantqual = i;
}
break;
}
case 'P':
{
rawInput = 1;
infile = malloc(sizeof(*infile));
if (infile == NULL) {
fprintf(stderr, "%s: unable to allocate memory\n", progName);
return 1;
}
infile->f = NULL;
infile->channels = 2;
infile->samplerate = 44100;
infile->samplebits = 16;
infile->samples = 0;
break;
}
case 'R':
{
unsigned int i;
if (rawInput != 1) {
fprintf(stderr, "%s: for raw pcm input -P needs to be specified first\n",
progName);
dieUsage = 1;
break;
}
if (sscanf(optarg, "%u", &i) > 0)
infile->samplerate = i;
break;
}
case 'B':
{
unsigned int i;
if (rawInput != 1) {
fprintf(stderr, "%s: for raw pcm input -P needs to be specified first\n",
progName);
dieUsage = 1;
break;
}
if (sscanf(optarg, "%u", &i) > 0)
infile->samplebits = i;
break;
}
case 'C':
{
unsigned int i;
if (rawInput != 1) {
fprintf(stderr, "%s: for raw pcm input -P needs to be specified first\n",
progName);
dieUsage = 1;
break;
}
if (sscanf(optarg, "%u", &i) > 0)
infile->channels = i;
break;
}
case '?':
break;
default:
fprintf(stderr, "%s: unknown option specified, ignoring: %c\n",
progName, c);
}
}
/* check that we have at least two non-option arguments */
if ((argc - optind) < 2 || dieUsage == 1)
{
fprintf(stderr, "\nUsage: %s -options infile outfile\n", progName);
fprintf(stderr, "Options:\n");
fprintf(stderr, " -m X AAC MPEG version, X can be 2 or 4.\n");
fprintf(stderr, " -o X AAC object type, X can be LC, MAIN or LTP.\n");
fprintf(stderr, " -n Don\'t use mid/side coding.\n");
fprintf(stderr, " -r RAW AAC output file.\n");
fprintf(stderr, " --notns\tDisable TNS coding.\n");
fprintf(stderr,
" -c <bandwidth>\tSet the bandwidth in Hz. (default=automatic)\n");
fprintf(stderr, " -q <quality>\tSet quantizer quality.\n");
fprintf(stderr, " -P Raw PCM input mode (default 44100Hz 16bit stereo).\n");
fprintf(stderr, " -R Raw PCM input rate.\n");
fprintf(stderr, " -B Raw PCM input sample size (16 default or 8bits).\n");
fprintf(stderr, " -C Raw PCM input channels.\n");
fprintf(stderr, "\n Note: output bitrate depends on -c and -q.\n");
return 1;
}
/* point to the specified file names */
audioFileName = argv[optind++];
aacFileName = argv[optind++];
/* open the audio input file */
if (rawInput == 1) {
if (!strcmp(audioFileName, "-")) {
#ifdef WIN32
setmode(fileno(stdin), O_BINARY);
#endif
infile->f = stdin;
infile->samples = 0;
}
else {
if (!(infile->f = fopen(audioFileName, "rb")))
{
fprintf(stderr, "Couldn't open input file %s\n", audioFileName);
perror("Reason");
return 1;
}
fseek(infile->f, 0 , SEEK_END);
infile->samples = ftell(infile->f) /
(((infile->samplebits > 8) ? 2 : 1) * infile->channels);
rewind(infile->f);
}
} else {
infile = wav_open_read(audioFileName);
}
if (infile == NULL)
{
fprintf(stderr, "Couldn't open input file %s\n", audioFileName);
return 1;
}
/* open the aac output file */
outfile = fopen(aacFileName, "wb");
if (!outfile)
{
fprintf(stderr, "Couldn't create output file %s\n", aacFileName);
return 1;
}
/* determine input file parameters */
sr = infile->samplerate;
chan = infile->channels;
/* open the encoder library */
hEncoder = faacEncOpen(sr, chan, &samplesInput, &maxBytesOutput);
pcmbuf = (short*)malloc(samplesInput*sizeof(short));
bitbuf = (unsigned char*)malloc(maxBytesOutput*sizeof(unsigned char));
if (cutOff <= 0)
{
if (cutOff < 0) // default
cutOff = 0;
else // disabled
cutOff = sr / 2;
}
if (cutOff > (sr / 2))
cutOff = sr / 2;
/* put the options in the configuration struct */
myFormat = faacEncGetCurrentConfiguration(hEncoder);
myFormat->aacObjectType = objectType;
myFormat->mpegVersion = mpegVersion;
myFormat->useTns = useTns;
myFormat->allowMidside = useMidSide;
//myFormat->bitRate = bitRate;
myFormat->bandWidth = cutOff;
if (quantqual > 0)
myFormat->quantqual = quantqual;
myFormat->outputFormat = useAdts;
if (!faacEncSetConfiguration(hEncoder, myFormat)) {
fprintf(stderr, "Unsupported output format!\n");
return 1;
}
cutOff = myFormat->bandWidth;
quantqual = myFormat->quantqual;
fprintf(stderr, "Quantization quality: %ld\n", quantqual);
fprintf(stderr, "Bandwidth: %d Hz\n", cutOff);
if (myFormat->useTns | myFormat->allowMidside)
fprintf(stderr, "Using:%s%s\n",
myFormat->useTns ? " TNS" : "",
myFormat->allowMidside ? " M/S" : "");
if (outfile)
{
int showcnt = 0;
#ifdef _WIN32
long begin = GetTickCount();
#endif
if (infile->samples)
frames = (int)infile->samples / 1024 + 2;
else
frames = 0;
currentFrame = 0;
fprintf(stderr, "Encoding %s\n", audioFileName);
if (frames != 0)
fprintf(stderr,
" frame | elapsed/estim | play/CPU | ETA\n");
else
fprintf(stderr,
" frame | elapsed | play/CPU\n");
/* encoding loop */
for ( ;; )
{
int bytesWritten;
bytesInput = wav_read_short(infile, pcmbuf, samplesInput) * sizeof(short);
/* call the actual encoding routine */
bytesWritten = faacEncEncode(hEncoder,
pcmbuf,
bytesInput/2,
bitbuf,
maxBytesOutput);
if (bytesWritten)
{
currentFrame++;
showcnt--;
}
if ((showcnt <= 0) || !bytesWritten)
{
double timeused;
#ifdef __unix__
struct rusage usage;
#endif
#ifdef _WIN32
char percent[50];
timeused = (GetTickCount() - begin) * 1e-3;
#else
#ifdef __unix__
if (getrusage(RUSAGE_SELF, &usage) == 0) {
timeused = (double)usage.ru_utime.tv_sec +
(double)usage.ru_utime.tv_usec * 1e-6;
}
else
timeused = 0;
#else
timeused = (double)clock() * (1.0 / CLOCKS_PER_SEC);
#endif
#endif
if (currentFrame && (timeused > 0.1))
{
showcnt += 50;
if (frames != 0)
fprintf(stderr,
"\r%5d/%-5d (%3d%%)| %6.1f/%-6.1f | %8.3f | %.1f ",
currentFrame, frames, currentFrame*100/frames,
timeused,
timeused * frames / currentFrame,
(1024.0 * currentFrame / sr) / timeused,
timeused * (frames - currentFrame) / currentFrame);
else
fprintf(stderr,
"\r %5d | %6.1f | %8.3f ",
currentFrame,
timeused,
(1024.0 * currentFrame / sr) / timeused);
fflush(stderr);
#ifdef _WIN32
if (frames != 0)
{
sprintf(percent, "%.2f%% encoding %s",
100.0 * currentFrame / frames, audioFileName);
SetConsoleTitle(percent);
}
#endif
}
}
/* all done, bail out */
if (!bytesInput && !bytesWritten)
break ;
if (bytesWritten < 0)
{
fprintf(stderr, "faacEncEncode() failed\n");
break ;
}
/* write bitstream to aac file */
fwrite(bitbuf, 1, bytesWritten, outfile);
}
fprintf(stderr, "\n\n");
/* clean up */
fclose(outfile);
}
faacEncClose(hEncoder);
wav_close(infile);
if (pcmbuf) free(pcmbuf);
if (bitbuf) free(bitbuf);
return 0;
}
/*
$Log: main.c,v $
Revision 1.33 2003/04/13 08:39:28 knik
removed psymodel setting
Revision 1.32 2003/03/27 17:11:06 knik
updated library interface
-b bitrate option replaced with -q quality option
TNS enabled by default
Revision 1.31 2002/12/23 19:02:43 knik
added some headers
Revision 1.30 2002/12/15 15:16:55 menno
Some portability changes
Revision 1.29 2002/11/23 17:34:59 knik
replaced libsndfile with input.c
improved bandwidth/bitrate calculation formula
Revision 1.28 2002/08/30 16:20:45 knik
misplaced #endif
Revision 1.27 2002/08/19 16:33:54 knik
automatic bitrate setting
more advanced status line
*/