main.c

音频编码

/*
 *      Command line frontend program
 *
 *      Copyright (c) 1999 Mark Taylor
 *                    2000 Takehiro TOMIANGA
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Library General Public
 * License as published by the Free Software Foundation; either
 * version 2 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
 * Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library 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 1.92 2005/03/13 14:42:21 robert Exp $ */

#ifdef HAVE_CONFIG_H
# include <config.h>
#endif

#include <assert.h>
#include <stdio.h>

#ifdef STDC_HEADERS
# include <stdlib.h>
# include <string.h>
#else
# ifndef HAVE_STRCHR
#  define strchr index
#  define strrchr rindex
# endif
char   *strchr(), *strrchr();
# ifndef HAVE_MEMCPY
#  define memcpy(d, s, n) bcopy ((s), (d), (n))
#  define memmove(d, s, n) bcopy ((s), (d), (n))
# endif
#endif

#ifdef HAVE_FCNTL_H
# include <fcntl.h>
#endif

#ifdef __sun__
/* woraround for SunOS 4.x, it has SEEK_* defined here */
#include <unistd.h>
#endif

#if defined(_WIN32)
# include <windows.h>
#endif


/*
 main.c is example code for how to use libmp3lame.a.  To use this library,
 you only need the library and lame.h.  All other .h files are private
 to the library.
*/
#include "lame.h"

#include "brhist.h"
#include "parse.h"
#include "main.h"
#include "get_audio.h"
#include "portableio.h"
#include "timestatus.h"

/* PLL 14/04/2000 */
#if macintosh
#include <console.h>
#endif

#ifdef WITH_DMALLOC
#include <dmalloc.h>
#endif




/************************************************************************
*
* main
*
* PURPOSE:  MPEG-1,2 Layer III encoder with GPSYCHO
* psychoacoustic model.
*
************************************************************************/

int
parse_args_from_string(lame_global_flags * const gfp, const char *p,
                       char *inPath, char *outPath)
{                       /* Quick & very Dirty */
    char   *q;
    char   *f;
    char   *r[128];
    int     c = 0;
    int     ret;

    if (p == NULL || *p == '\0')
        return 0;

    f = q = malloc(strlen(p) + 1);
    strcpy(q, p);

    r[c++] = "lhama";
    while (1) {
        r[c++] = q;
        while (*q != ' ' && *q != '\0')
            q++;
        if (*q == '\0')
            break;
        *q++ = '\0';
    }
    r[c] = NULL;

    ret = parse_args(gfp, c, r, inPath, outPath, NULL, NULL);
    free(f);
    return ret;
}





FILE   *
init_files(lame_global_flags * gf, char *inPath, char *outPath)
{
    FILE   *outf;
    /* Mostly it is not useful to use the same input and output name.
       This test is very easy and buggy and don't recognize different names
       assigning the same file
     */
    if (0 != strcmp("-", outPath) && 0 == strcmp(inPath, outPath)) {
        fprintf(stderr, "Input file and Output file are the same. Abort.\n");
        return NULL;
    }

    /* open the wav/aiff/raw pcm or mp3 input file.  This call will
     * open the file, try to parse the headers and
     * set gf.samplerate, gf.num_channels, gf.num_samples.
     * if you want to do your own file input, skip this call and set
     * samplerate, num_channels and num_samples yourself.
     */
    init_infile(gf, inPath);
    if ((outf = init_outfile(outPath, lame_get_decode_only(gf))) == NULL) {
        fprintf(stderr, "Can't init outfile '%s'\n", outPath);
        return NULL;
    }

    return outf;
}






/* the simple lame decoder */
/* After calling lame_init(), lame_init_params() and
 * init_infile(), call this routine to read the input MP3 file
 * and output .wav data to the specified file pointer*/
/* lame_decoder will ignore the first 528 samples, since these samples
 * represent the mpglib delay (and are all 0).  skip = number of additional
 * samples to skip, to (for example) compensate for the encoder delay */

int
lame_decoder(lame_global_flags * gfp, FILE * outf, int skip, char *inPath,
             char *outPath)
{
    short int Buffer[2][1152];
    int     iread;
    double  wavsize;
    int     i;
    void    (*WriteFunction) (FILE * fp, char *p, int n);
    int tmp_num_channels = lame_get_num_channels( gfp );



    if (silent < 10) fprintf(stderr, "\rinput:  %s%s(%g kHz, %i channel%s, ",
            strcmp(inPath, "-") ? inPath : "<stdin>",
            strlen(inPath) > 26 ? "\n\t" : "  ",
            lame_get_in_samplerate( gfp ) / 1.e3,
            tmp_num_channels, tmp_num_channels != 1 ? "s" : "");

    switch (input_format) {
    case sf_mp3:
        if (skip==0) {
            if (enc_delay>-1) skip = enc_delay + 528+1;
            else skip=lame_get_encoder_delay(gfp)+528+1;
        }else{
            /* user specified a value of skip. just add for decoder */
            skip += 528+1; /* mp3 decoder has a 528 sample delay, plus user supplied "skip" */
        }

        if (silent < 10) fprintf(stderr, "MPEG-%u%s Layer %s", 2 - lame_get_version(gfp),
                lame_get_out_samplerate( gfp ) < 16000 ? ".5" : "", "III");
        break;
    case sf_mp2:
        skip += 240 + 1;
        if (silent < 10) fprintf(stderr, "MPEG-%u%s Layer %s", 2 - lame_get_version(gfp),
                lame_get_out_samplerate( gfp ) < 16000 ? ".5" : "", "II");
        break;
    case sf_mp1:
        skip += 240 + 1;
        if (silent < 10) fprintf(stderr, "MPEG-%u%s Layer %s", 2 - lame_get_version(gfp),
                lame_get_out_samplerate( gfp ) < 16000 ? ".5" : "", "I");
        break;
    case sf_raw:
        if (silent < 10) fprintf(stderr, "raw PCM data");
        mp3input_data.nsamp = lame_get_num_samples( gfp );
        mp3input_data.framesize = 1152;
        skip = 0;       /* other formats have no delay *//* is += 0 not better ??? */
        break;
    case sf_wave:
        if (silent < 10) fprintf(stderr, "Microsoft WAVE");
        mp3input_data.nsamp = lame_get_num_samples( gfp );
        mp3input_data.framesize = 1152;
        skip = 0;       /* other formats have no delay *//* is += 0 not better ??? */
        break;
    case sf_aiff:
        if (silent < 10) fprintf(stderr, "SGI/Apple AIFF");
        mp3input_data.nsamp = lame_get_num_samples( gfp );
        mp3input_data.framesize = 1152;
        skip = 0;       /* other formats have no delay *//* is += 0 not better ??? */
        break;
    default:
        if (silent < 10) fprintf(stderr, "unknown");
        mp3input_data.nsamp = lame_get_num_samples( gfp );
        mp3input_data.framesize = 1152;
        skip = 0;       /* other formats have no delay *//* is += 0 not better ??? */
        assert(0);
        break;
    }

    if (silent < 10) {
	fprintf(stderr, ")\noutput: %s%s(16 bit, Microsoft WAVE)\n",
		strcmp(outPath, "-") ? outPath : "<stdout>",
		strlen(outPath) > 45 ? "\n\t" : "  ");

	if (skip > 0)
	    fprintf(stderr,
		    "skipping initial %i samples (encoder+decoder delay)\n",
		    skip);
    }

    if ( 0 == disable_wav_header )
        WriteWaveHeader(outf, 0x7FFFFFFF, lame_get_in_samplerate( gfp ),
                        tmp_num_channels,
                        16);
    /* unknown size, so write maximum 32 bit signed value */

    wavsize = -skip;
    WriteFunction = swapbytes ? WriteBytesSwapped : WriteBytes;
    mp3input_data.totalframes = mp3input_data.nsamp / mp3input_data.framesize;

    assert(tmp_num_channels >= 1 && tmp_num_channels <= 2);

    do {
        iread = get_audio16(gfp, Buffer); /* read in 'iread' samples */
        mp3input_data.framenum += iread / mp3input_data.framesize;
        wavsize += iread;

        if (silent <= 0)
            decoder_progress(gfp, &mp3input_data);

        skip -= (i = skip < iread ? skip : iread); /* 'i' samples are to skip in this frame */

        for (; i < iread; i++) {
            if ( disable_wav_header ) {
                WriteFunction(outf, (char *) &Buffer[0][i], sizeof(short));
                if (tmp_num_channels == 2)
                    WriteFunction(outf, (char *) &Buffer[1][i], sizeof(short));
            }
            else {
                Write16BitsLowHigh(outf, Buffer[0][i]);
                if (tmp_num_channels == 2)
                    Write16BitsLowHigh(outf, Buffer[1][i]);
            }
        }
    } while (iread);

    i = (16 / 8) * tmp_num_channels;
    assert(i > 0);
    if (wavsize <= 0) {
        if (silent < 10) fprintf(stderr, "WAVE file contains 0 PCM samples\n");
        wavsize = 0;
    }
    else if (wavsize > 0xFFFFFFD0 / i) {
        if (silent < 10) fprintf(stderr,
                "Very huge WAVE file, can't set filesize accordingly\n");
        wavsize = 0xFFFFFFD0;
    }
    else {
        wavsize *= i;
    }

    if ( 0 == disable_wav_header )
        if (!fseek(outf, 0l, SEEK_SET)) /* if outf is seekable, rewind and adjust length */
            WriteWaveHeader(outf, (int)wavsize, lame_get_in_samplerate( gfp ),
                            tmp_num_channels, 16);
    fclose(outf);

    if (silent <= 0)
	decoder_progress_finish(gfp);
    return 0;
}
















int
lame_encoder(lame_global_flags * gf, FILE * outf, int nogap, char *inPath,
             char *outPath)
{
    unsigned char mp3buffer[LAME_MAXMP3BUFFER];
    int     Buffer[2][1152];
    int     iread, imp3;
    static const char *mode_names[2][4] = {
        {"stereo", "j-stereo", "dual-ch", "single-ch"},
        {"stereo", "force-ms", "dual-ch", "single-ch"}
    };
    int     frames;

    if (silent < 10) {
        lame_print_config(gf); /* print useful information about options being used */

        fprintf(stderr, "Encoding %s%s to %s\n",
                strcmp(inPath, "-") ? inPath : "<stdin>",
                strlen(inPath) + strlen(outPath) < 66 ? "" : "\n     ",
                strcmp(outPath, "-") ? outPath : "<stdout>");

        fprintf(stderr,
                "Encoding as %g kHz ", 1.e-3 * lame_get_out_samplerate(gf));

	{
            const char *appendix = "";

            switch (lame_get_VBR(gf)) {
            case vbr_mt:
            case vbr_rh:
            case vbr_mtrh:
                appendix = "ca. ";
                fprintf(stderr, "VBR(q=%i)", lame_get_VBR_q(gf));
                break;
            case vbr_abr:
                fprintf(stderr, "average %d kbps",
                        lame_get_VBR_mean_bitrate_kbps(gf));
                break;
            default:
                fprintf(stderr, "%3d kbps", lame_get_brate(gf));
                break;
            }
            fprintf(stderr, " %s MPEG-%u%s Layer III (%s%gx) qval=%i\n",
                    mode_names[lame_get_force_ms(gf)][lame_get_mode(gf)],
                    2 - lame_get_version(gf),
                    lame_get_out_samplerate(gf) < 16000 ? ".5" : "",
                    appendix,
                    0.1 * (int) (10. * lame_get_compression_ratio(gf) + 0.5),
                    lame_get_quality(gf));
        }

        if (silent <= -10)
            lame_print_internals(gf);

        fflush(stderr);
    }


    /* encode until we hit eof */
    do {
        /* read in 'iread' samples */
        iread = get_audio(gf, Buffer);
        frames = lame_get_frameNum(gf);


 /********************** status display  *****************************/
        if (silent <= 0) {
            if (update_interval > 0) {
                timestatus_klemm(gf);
            }
            else {
                if (0 == frames % 50) {
#ifdef BRHIST
                    brhist_jump_back();
#endif
                    timestatus(lame_get_out_samplerate(gf),
                               frames,
                               lame_get_totalframes(gf),
                               lame_get_framesize(gf));