/*
 *	Get Audio routines source file
 *
 *	Copyright (c) 1999 Albert L Faber
 *
 * 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: get_audio.c,v 1.79 2001/10/29 22:00:15 markt Exp $ */


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

#include <assert.h>

#ifdef HAVE_LIMITS_H
# include <limits.h>
#endif

#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

#define         MAX_U_32_NUM            0xFFFFFFFF


#include <math.h>
#include <sys/stat.h>

#include "lame.h"
#include "main.h"
#include "get_audio.h"
#include "portableio.h"
#include "timestatus.h"
#include "lametime.h"

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


/* global data for get_audio.c. */
int     count_samples_carefully;
int     pcmbitwidth;
unsigned int num_samples_read;
FILE   *musicin;


#ifdef AMIGA_MPEGA
int     lame_decode_initfile(const char *fullname,
                             mp3data_struct * const mp3data);
#else
int     lame_decode_initfile(FILE * const fd, mp3data_struct * const mp3data);
#endif

/* read mp3 file until mpglib returns one frame of PCM data */
int     lame_decode_fromfile(FILE * fd, short int pcm_l[], short int pcm_r[],
                             mp3data_struct * mp3data);

/* and for Vorbis: */
int     lame_decode_ogg_initfile( lame_global_flags*  gfp,
                                  FILE*               fd,
                                  mp3data_struct*     mp3data );
int     lame_decode_ogg_fromfile( lame_global_flags*  gfc,
                                  FILE*               fd,
                                  short int           pcm_l[],
                                  short int           pcm_r[],
                                  mp3data_struct*     mp3data );


static int read_samples_pcm(FILE * musicin, int sample_buffer[2304],
                            int frame_size, int samples_to_read);
static int read_samples_mp3(lame_global_flags * gfp, FILE * musicin,
                            short int mpg123pcm[2][1152], int num_chan);
static int read_samples_ogg(lame_global_flags * gfp, FILE * musicin,
                            short int mpg123pcm[2][1152], int num_chan);
void    CloseSndFile(sound_file_format input, FILE * musicin);
FILE   *OpenSndFile(lame_global_flags * gfp, char *);


/* Replacement for forward fseek(,,SEEK_CUR), because fseek() fails on pipes */


static int
fskip(FILE * fp, long offset, int whence)
{
#ifndef PIPE_BUF
    char    buffer[4096];
#else
    char    buffer[PIPE_BUF];
#endif
    int     read;

    if (0 == fseek(fp, offset, whence))
        return 0;

    if (whence != SEEK_CUR || offset < 0) {
        fprintf(stderr,
                "fskip problem: Mostly the return status of functions is not evaluate so it is more secure to polute <stderr>.\n");
        return -1;
    }

    while (offset > 0) {
        read = offset > sizeof(buffer) ? sizeof(buffer) : offset;
        if ((read = fread(buffer, 1, read, fp)) <= 0)
            return -1;
        offset -= read;
    }

    return 0;
}


FILE   *
init_outfile(char *outPath, int decode)
{
    FILE   *outf;
#ifdef __riscos__
    char   *p;
#endif

    /* open the output file */
    if (0 == strcmp(outPath, "-")) {
        lame_set_stream_binary_mode(outf = stdout);
    }
    else {
        if ((outf = fopen(outPath, "wb+")) == NULL)
            return NULL;
#ifdef __riscos__
        /* Assign correct file type */
        for (p = outPath; *p; p++) /* ugly, ugly to modify a string */
            switch (*p) {
            case '.':
                *p = '/';
                break;
            case '/':
                *p = '.';
                break;
            }
        SetFiletype(outPath, decode ? 0xFB1 /*WAV*/ : 0x1AD /*AMPEG*/);
#endif
    }
    return outf;
}






void
init_infile(lame_global_flags * gfp, char *inPath)
{
    /* open the input file */
    count_samples_carefully = 0;
    num_samples_read=0;
    pcmbitwidth=in_bitwidth;
    musicin = OpenSndFile(gfp, inPath);
}

void
close_infile(void)
{
    CloseSndFile(input_format, musicin);
}


void
SwapBytesInWords(short *ptr, int short_words)
{                       /* Some speedy code */
    unsigned long val;
    unsigned long *p = (unsigned long *) ptr;

#ifndef lint
# if defined(CHAR_BIT)
#  if CHAR_BIT != 8
#   error CHAR_BIT != 8
#  endif
# else
#  error can not determine number of bits in a char
# endif
#endif /* lint */

    assert(sizeof(short) == 2);


#if defined(SIZEOF_UNSIGNED_LONG) && SIZEOF_UNSIGNED_LONG == 4
    for (; short_words >= 2; short_words -= 2, p++) {
        val = *p;
        *p = ((val << 8) & 0xFF00FF00) | ((val >> 8) & 0x00FF00FF);
    }
    ptr = (short *) p;
    for (; short_words >= 1; short_words -= 1, ptr++) {
        val = *ptr;
        *ptr = ((val << 8) & 0xFF00) | ((val >> 8) & 0x00FF);
    }
#elif defined(SIZEOF_UNSIGNED_LONG) && SIZEOF_UNSIGNED_LONG == 8
    for (; short_words >= 4; short_words -= 4, p++) {
        val = *p;
        *p =
            ((val << 8) & 0xFF00FF00FF00FF00) | ((val >> 8) &
                                                 0x00FF00FF00FF00FF);
    }
    ptr = (short *) p;
    for (; short_words >= 1; short_words -= 1, ptr++) {
        val = *ptr;
        *ptr = ((val << 8) & 0xFF00) | ((val >> 8) & 0x00FF);
    }
#else
# ifdef SIZEOF_UNSIGNED_LONG
#  warning Using unoptimized SwapBytesInWords().
# endif
    for (; short_words >= 1; short_words -= 1, ptr++) {
        val = *ptr;
        *ptr = ((val << 8) & 0xFF00) | ((val >> 8) & 0x00FF);
    }
#endif

    assert(short_words == 0);
}



static int
get_audio_common( lame_global_flags * const gfp,
		  int buffer[2][1152], short buffer16[2][1152] );

/************************************************************************
*
* get_audio()
*
* PURPOSE:  reads a frame of audio data from a file to the buffer,
*   aligns the data for future processing, and separates the
*   left and right channels
*
************************************************************************/
int
get_audio( lame_global_flags * const gfp, int buffer[2][1152] )
{
    return( get_audio_common( gfp, buffer, NULL ) );
}

/*
  get_audio16 - behave as the original get_audio function, with a limited
                16 bit per sample output
*/
int
get_audio16( lame_global_flags * const gfp, short buffer[2][1152] )
{
    return( get_audio_common( gfp, NULL, buffer ) );
}

/************************************************************************
  get_audio_common - central functionality of get_audio*
    in: gfp
        buffer    output to the int buffer or 16-bit buffer
   out: buffer    int output    (if buffer != NULL)
        buffer16  16-bit output (if buffer == NULL) 
returns: samples read
note: either buffer or buffer16 must be allocated upon call
*/
static int
get_audio_common( lame_global_flags * const gfp,
		  int buffer[2][1152], short buffer16[2][1152] )
{
    int     num_channels = lame_get_num_channels( gfp );
    int     insamp[2 * 1152];
    short   buf_tmp16[2][1152];
    int     samples_read;
    int     framesize;
    int     samples_to_read;
    unsigned int remaining, tmp_num_samples;
    int     i;
    int     *p;

    /* 
     * NOTE: LAME can now handle arbritray size input data packets,
     * so there is no reason to read the input data in chuncks of
     * size "framesize".  EXCEPT:  the LAME graphical frame analyzer 
     * will get out of sync if we read more than framesize worth of data.
     */

    samples_to_read = framesize = lame_get_framesize(gfp);
    assert(framesize <= 1152);

    /* get num_samples */
    tmp_num_samples = lame_get_num_samples( gfp );

    /* if this flag has been set, then we are carefull to read
     * exactly num_samples and no more.  This is useful for .wav and .aiff
     * files which have id3 or other tags at the end.  Note that if you
     * are using LIBSNDFILE, this is not necessary 
     */
    if (count_samples_carefully) {
        remaining = tmp_num_samples - Min(tmp_num_samples, num_samples_read);
        if (remaining < framesize)
            samples_to_read = remaining;
    }

    switch (input_format) {
    case sf_mp1:
    case sf_mp2:
    case sf_mp3:
	if( buffer != NULL )
	    samples_read = read_samples_mp3( gfp, musicin,
					     buf_tmp16, num_channels );
	else
	    samples_read = read_samples_mp3( gfp, musicin,
					     buffer16, num_channels );
        break;
    case sf_ogg:
	if( buffer != NULL )
	    samples_read = read_samples_ogg( gfp, musicin,
					     buf_tmp16, num_channels );
	else
	    samples_read = read_samples_ogg( gfp, musicin,
					     buffer16, num_channels );
        break;
    default:
        samples_read =
            read_samples_pcm(musicin, insamp, num_channels * framesize,
                             num_channels * samples_to_read);
	p = insamp + samples_read;
        samples_read /= num_channels;
	if( buffer != NULL ) {	/* output to int buffer */
	    if( num_channels == 2 ) {
		for( i = samples_read; --i >= 0; ) {
		    buffer[1][i] = *--p;
 		    buffer[0][i] = *--p;
		}
	    } else if( num_channels == 1 ) {
		memset( buffer[1], 0, samples_read * sizeof(int) );
		for( i = samples_read; --i >= 0; ) {
 		    buffer[0][i] = *--p;
		}
	    } else
		assert(0);
	} else {		/* convert from int; output to 16-bit buffer */
	    if( num_channels == 2 ) {
		for( i = samples_read; --i >= 0; ) {
		    buffer16[1][i] = *--p >> (8 * sizeof(int) - 16);
 		    buffer16[0][i] = *--p >> (8 * sizeof(int) - 16);
		}
	    } else if( num_channels == 1 ) {
		memset( buffer16[1], 0, samples_read * sizeof(short) );
		for( i = samples_read; --i >= 0; ) {
 		    buffer16[0][i] = *--p >> (8 * sizeof(int) - 16);
		}
	    } else
		assert(0);
	}
    }

    /* LAME mp3 and ogg output 16bit -  convert to int, if necessary */
    if( input_format == sf_mp1 || input_format == sf_mp2 || 
        input_format == sf_mp3 || input_format == sf_ogg ) {
	if( buffer != NULL ) {
	    for( i = samples_read; --i >= 0; )
		buffer[0][i] = buf_tmp16[0][i] << (8 * sizeof(int) - 16);
	    if( num_channels == 2 ) {
		for( i = samples_read; --i >= 0; )
		    buffer[1][i] = buf_tmp16[1][i] << (8 * sizeof(int) - 16);
	    } else if( num_channels == 1 ) {
		memset( buffer[1], 0, samples_read * sizeof(int) );
	    } else
		assert(0);
	}
    }


    /* if num_samples = MAX_U_32_NUM, then it is considered infinitely long.
       Don't count the samples */
    if ( tmp_num_samples != MAX_U_32_NUM )
        num_samples_read += samples_read;

    return samples_read;
}



int
read_samples_ogg(lame_global_flags * const gfp,
                 FILE * const musicin,
                 short int oggpcm[2][1152], const int stereo)
{
    int     out = 0;

#ifdef HAVE_VORBIS
    static const char type_name[] = "Ogg Vorbis file";

    out =
        lame_decode_ogg_fromfile( gfp,
                                  musicin,
                                  oggpcm[0],
                                  oggpcm[1],
                                  &mp3input_data );
    /*
     * out < 0:  error, probably EOF
     * out = 0:  not possible with lame_decode_fromfile() ???
     * out > 0:  number of output samples
     */

    if (out < 0) {
        memset(oggpcm, 0, sizeof(**oggpcm) * 2 * 1152);
        return 0;
    }

    if (lame_get_num_channels( gfp ) != mp3input_data.stereo)
        fprintf(stderr,
                "Error: number of channels has changed in %s - not supported\n",
                type_name);
    if ( lame_get_in_samplerate( gfp ) != mp3input_data.samplerate )
        fprintf(stderr,
                "Error: sample frequency has changed in %s - not supported\n",
                type_name);

#else
    out = -1;           /* wanna read ogg without vorbis support? */
#endif

    return out;
}


int
read_samples_mp3(lame_global_flags * const gfp,
                 FILE * const musicin, short int mpg123pcm[2][1152], int stereo)
{
    int     out;
#if defined(AMIGA_MPEGA)  ||  defined(HAVE_MPGLIB)
    static const char type_name[] = "MP3 file";

    out =
        lame_decode_fromfile(musicin, mpg123pcm[0], mpg123pcm[1],
                             &mp3input_data);
    /*
     * out < 0:  error, probably EOF
     * out = 0:  not possible with lame_decode_fromfile() ???
     * out > 0:  number of output samples
     */
    if (out < 0) {
        memset(mpg123pcm, 0, sizeof(**mpg123pcm) * 2 * 1152);
        return 0;
    }

    if ( lame_get_num_channels( gfp ) != mp3input_data.stereo )
        fprintf(stderr,
                "Error: number of channels has changed in %s - not supported\n",
                type_name);
    if ( lame_get_in_samplerate( gfp ) != mp3input_data.samplerate )
        fprintf(stderr,
                "Error: sample frequency has changed in %s - not supported\n",
                type_name);

#else
    out = -1;
#endif
    return out;
}


int
WriteWaveHeader(FILE * const fp, const int pcmbytes,
                const int freq, const int channels, const int bits)
{
    int     bytes = (bits + 7) / 8;

    /* quick and dirty, but documented */
    fwrite("RIFF", 1, 4, fp); // label
    Write32BitsLowHigh(fp, pcmbytes + 44 - 8); // length in bytes without header
    fwrite("WAVEfmt ", 2, 4, fp); // 2 labels
    Write32BitsLowHigh(fp, 2 + 2 + 4 + 4 + 2 + 2); // length of PCM format declaration area
    Write16BitsLowHigh(fp, 1); // is PCM?
    Write16BitsLowHigh(fp, channels); // number of channels
    Write32BitsLowHigh(fp, freq); // sample frequency in [Hz]
    Write32BitsLowHigh(fp, freq * channels * bytes); // bytes per second
    Write16BitsLowHigh(fp, channels * bytes); // bytes per sample time
    Write16BitsLowHigh(fp, bits); // bits per sample
    fwrite("data", 1, 4, fp); // label
    Write32BitsLowHigh(fp, pcmbytes); // length in bytes of raw PCM data

    return ferror(fp) ? -1 : 0;
}




#if defined(LIBSNDFILE)

#if 0                   /* currently disabled */
# include "sndfile.h"   // prototype for sf_get_lib_version()
void
print_sndlib_version(FILE * fp)
{
    char    tmp[80];
    sf_get_lib_version(tmp, sizeof(tmp));
    fprintf(fp,
            "Input handled by %s  (http://www.zip.com.au/~erikd/libsndfile/)\n",
            tmp);
}
#endif

/*
** Copyright (C) 1999 Albert Faber
**
 * 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.
 */






void
CloseSndFile(sound_file_format input, FILE * musicin)
{
    SNDFILE *gs_pSndFileIn = (SNDFILE *) musicin;
    if (input == sf_mp1 || input == sf_mp2 || input == sf_mp3) {
#ifndef AMIGA_MPEGA
        if (fclose(musicin) != 0) {
            fprintf(stderr, "Could not close audio input file\n");
            exit(2);
        }
#endif
    }
    else {
        if (gs_pSndFileIn) {
            if (sf_close(gs_pSndFileIn) != 0) {
                fprintf(stderr, "Could not close sound file \n");
                exit(2);
            }
        }
    }
}



FILE   *
OpenSndFile(lame_global_flags * gfp, char *inPath)
{
    char   *lpszFileName = inPath;
    FILE   *musicin;
    SNDFILE *gs_pSndFileIn;
    SF_INFO gs_wfInfo;

    if (input_format == sf_mp1 ||
        input_format == sf_mp2 || input_format == sf_mp3) {
#ifdef AMIGA_MPEGA
        if (-1 == lame_decode_initfile(lpszFileName, &mp3input_data)) {
            fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
                    lpszFileName);
            exit(1);
        }
#endif
#ifdef HAVE_MPGLIB
        if ((musicin = fopen(lpszFileName, "rb")) == NULL) {
            fprintf(stderr, "Could not find \"%s\".\n", lpszFileName);
            exit(1);
        }
        if (-1 == lame_decode_initfile(musicin, &mp3input_data)) {
            fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
                    lpszFileName);
            exit(1);
        }
#endif

        if( -1 == lame_set_num_channels( gfp, mp3input_data.stereo ) ) {
            fprintf( stderr,
                     "Unsupported number of channels: %ud\n",
                     mp3input_data.stereo );
            exit( 1 );
        }
        (void) lame_set_in_samplerate( gfp, mp3input_data.samplerate );
        (void) lame_set_num_samples( gfp, mp3input_data.nsamp );
    }
    else if (input_format == sf_ogg) {
#ifdef HAVE_VORBIS
        if ((musicin = fopen(lpszFileName, "rb")) == NULL) {
            fprintf(stderr, "Could not find \"%s\".\n", lpszFileName);
            exit(1);
        }
        if ( -1 == lame_decode_ogg_initfile( gfp,
                                             musicin,
                                             &mp3input_data ) ) {
            fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
                    lpszFileName);
            exit(1);
        }
#else
        fprintf(stderr, "LAME not compiled with libvorbis support.\n");
        exit(1);
#endif


    }
    else {

        /* Try to open the sound file */
        /* set some defaults incase input is raw PCM */
        gs_wfInfo.seekable = (input_format != sf_raw); /* if user specified -r, set to not seekable */
        gs_wfInfo.samplerate = lame_get_in_samplerate( gfp );
        gs_wfInfo.pcmbitwidth = in_bitwidth;
        gs_wfInfo.channels = lame_get_num_channels( gfp );

	if (in_bitwidth == 8) {
	    if (in_signed)
		gs_wfInfo.format = SF_FORMAT_RAW_S8;
	    else
		gs_wfInfo.format = SF_FORMAT_RAW_U8;

	} else {
	    if (!in_signed) {
		fputs("Unsigned input only supported with bitwidth 8\n", stderr);
		exit(1);
	    }
	    if (in_endian != order_unknown) {
		if (in_endian == order_littleEndian)
		    gs_wfInfo.format = SF_FORMAT_RAW_LE;
		else
		    gs_wfInfo.format = SF_FORMAT_RAW_BE;
	    } else {
#ifndef WORDS_BIGENDIAN
		/* little endian */
		if (swapbytes)
		    gs_wfInfo.format = SF_FORMAT_RAW_BE;
		else
		    gs_wfInfo.format = SF_FORMAT_RAW_LE;
#else
		if (swapbytes)
		    gs_wfInfo.format = SF_FORMAT_RAW_LE;
		else
		    gs_wfInfo.format = SF_FORMAT_RAW_BE;
#endif
	    }
	}

        gs_pSndFileIn = sf_open_read(lpszFileName, &gs_wfInfo);
        musicin = (SNDFILE *) gs_pSndFileIn;

        /* Check result */
        if (gs_pSndFileIn == NULL) {
            sf_perror(gs_pSndFileIn);
            fprintf(stderr, "Could not open sound file \"%s\".\n",
                    lpszFileName);
            exit(1);
        }

        if ((gs_wfInfo.format == SF_FORMAT_RAW_LE) ||
            (gs_wfInfo.format == SF_FORMAT_RAW_BE) ||
	    (gs_wfInfo.format == SF_FORMAT_RAW_S8) ||
	    (gs_wfInfo.format == SF_FORMAT_RAW_U8))
	    input_format = sf_raw;

#ifdef _DEBUG_SND_FILE
        DEBUGF("\n\nSF_INFO structure\n");
        DEBUGF("samplerate        :%d\n", gs_wfInfo.samplerate);
        DEBUGF("samples           :%d\n", gs_wfInfo.samples);
        DEBUGF("channels          :%d\n", gs_wfInfo.channels);
        DEBUGF("pcmbitwidth       :%d\n", gs_wfInfo.pcmbitwidth);
        DEBUGF("format            :");

        /* new formats from sbellon@sbellon.de  1/2000 */

        switch (gs_wfInfo.format & SF_FORMAT_TYPEMASK) {
        case SF_FORMAT_WAV:
            DEBUGF("Microsoft WAV format (big endian). ");
            break;
        case SF_FORMAT_AIFF:
            DEBUGF("Apple/SGI AIFF format (little endian). ");
            break;
        case SF_FORMAT_AU:
            DEBUGF("Sun/NeXT AU format (big endian). ");
            break;
        case SF_FORMAT_AULE:
            DEBUGF("DEC AU format (little endian). ");
            break;
        case SF_FORMAT_RAW:
            DEBUGF("RAW PCM data. ");
            break;
        case SF_FORMAT_PAF:
            DEBUGF("Ensoniq PARIS file format. ");
            break;
        case SF_FORMAT_SVX:
            DEBUGF("Amiga IFF / SVX8 / SV16 format. ");
            break;
        case SF_FORMAT_NIST:
            DEBUGF("Sphere NIST format. ");
            break;
        default:
            assert(0);
            break;
        }

        switch (gs_wfInfo.format & SF_FORMAT_SUBMASK) {
        case SF_FORMAT_PCM:
            DEBUGF("PCM data in 8, 16, 24 or 32 bits.");
            break;
        case SF_FORMAT_FLOAT:
            DEBUGF("32 bit Intel x86 floats.");
            break;
        case SF_FORMAT_ULAW:
            DEBUGF("U-Law encoded.");
            break;
        case SF_FORMAT_ALAW:
            DEBUGF("A-Law encoded.");
            break;
        case SF_FORMAT_IMA_ADPCM:
            DEBUGF("IMA ADPCM.");
            break;
        case SF_FORMAT_MS_ADPCM:
            DEBUGF("Microsoft ADPCM.");
            break;
        case SF_FORMAT_PCM_BE:
            DEBUGF("Big endian PCM data.");
            break;
        case SF_FORMAT_PCM_LE:
            DEBUGF("Little endian PCM data.");
            break;
        case SF_FORMAT_PCM_S8:
            DEBUGF("Signed 8 bit PCM.");
            break;
        case SF_FORMAT_PCM_U8:
            DEBUGF("Unsigned 8 bit PCM.");
            break;
        case SF_FORMAT_SVX_FIB:
            DEBUGF("SVX Fibonacci Delta encoding.");
            break;
        case SF_FORMAT_SVX_EXP:
            DEBUGF("SVX Exponential Delta encoding.");
            break;
        default:
            assert(0);
            break;
        }

        DEBUGF("\n");
        DEBUGF("pcmbitwidth       :%d\n", gs_wfInfo.pcmbitwidth);
        DEBUGF("sections          :%d\n", gs_wfInfo.sections);
        DEBUGF("seekable          :\n", gs_wfInfo.seekable);
#endif

        (void) lame_set_num_samples( gfp, gs_wfInfo.samples );
        if( -1 == lame_set_num_channels( gfp, gs_wfInfo.channels ) ) {
            fprintf( stderr,
                     "Unsupported number of channels: %ud\n",
                     gs_wfInfo.channels );
            exit( 1 );
        }
        (void) lame_set_in_samplerate( gfp, gs_wfInfo.samplerate );
        pcmbitwidth = gs_wfInfo.pcmbitwidth;
    }

    if (lame_get_num_samples( gfp ) == MAX_U_32_NUM) {
        /* try to figure out num_samples */
        double  flen = lame_get_file_size( lpszFileName );

        if (flen >= 0) {
            /* try file size, assume 2 bytes per sample */
            if (input_format == sf_mp1 ||
                input_format == sf_mp2 || input_format == sf_mp3) {
		if (mp3input_data.bitrate>0) {
		    double  totalseconds =
			(flen * 8.0 / (1000.0 * mp3input_data.bitrate));
		    unsigned long tmp_num_samples =
			totalseconds * lame_get_in_samplerate( gfp );
		    
		    (void) lame_set_num_samples( gfp, tmp_num_samples );
		    mp3input_data.nsamp = tmp_num_samples;
		}
            }
            else {
                lame_set_num_samples( gfp,
                    flen / (2 * lame_get_num_channels( gfp )) );
            }
        }
    }


    return musicin;
}


/************************************************************************
*
* read_samples()
*
* PURPOSE:  reads the PCM samples from a file to the buffer
*
*  SEMANTICS:
* Reads #samples_read# number of shorts from #musicin# filepointer
* into #sample_buffer[]#.  Returns the number of samples read.
*
************************************************************************/

static int
read_samples_pcm(FILE * const musicin, int sample_buffer[2304],
                 int frame_size /* unused */ , int samples_to_read)
{
    int     i;
    int     samples_read;

    samples_read =
        sf_read_int((SNDFILE *) musicin, sample_buffer, samples_to_read);

    switch (pcmbitwidth) {
    case 8:
        for (i = 0; i < samples_read; i++)
            sample_buffer[i] <<= (8 * sizeof(int) - 8);
        break;
    case 16:
        for (i = 0; i < samples_read; i++)
            sample_buffer[i] <<= (8 * sizeof(int) - 16);
        break;
    case 24:
        for (i = 0; i < samples_read; i++)
            sample_buffer[i] <<= (8 * sizeof(int) - 24);
	break;
    case 32:
	break;
    default:
        fprintf(stderr, "Only 8, 16, 24 and 32 bit input files supported \n");
        exit(1);
    }

    return samples_read;
}


#else /* defined(LIBSNDFILE) */

/************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************
 *
 * OLD ISO/LAME routines follow.  Used if you dont have LIBSNDFILE
 * or for stdin/stdout support
 *
 ************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************
 ************************************************************************/



/************************************************************************
unpack_read_samples - read and unpack signed low-to-high byte or unsigned
                      single byte input. (used for read_samples function)
                      Output integers are stored in the native byte order
                      (little or big endian).  -jd
  in: samples_to_read
      bytes_per_sample
      swap_order    - set for high-to-low byte order input stream
 i/o: pcm_in
 out: sample_buffer  (must be allocated up to samples_to_read upon call)
returns: number of samples read
*/
static int
unpack_read_samples( const int samples_to_read, const int bytes_per_sample,
		     const int swap_order, int *sample_buffer, FILE *pcm_in )
{
    int samples_read;
    int i;
    int *op;			/* output pointer */
    unsigned char *ip = (unsigned char *) sample_buffer; /* input pointer */
    const int b = sizeof(int) * 8;

#define GA_URS_IFLOOP( ga_urs_bps ) \
    if( bytes_per_sample == ga_urs_bps ) \
	for( i = samples_read * bytes_per_sample; (i -= bytes_per_sample) >=0;)
    

    samples_read = fread( sample_buffer, bytes_per_sample, 
			  samples_to_read, pcm_in);
    op = sample_buffer + samples_read;

    GA_URS_IFLOOP( 1 )
	*--op = (ip[i] ^ 0x80)<<(b-8) | 0x7f<<(b-16);/* convert from unsigned*/
    if( swap_order == 0 ) {
	GA_URS_IFLOOP( 2 )
	    *--op = ip[i]<<(b-16) | ip[i+1]<<(b-8); 
	GA_URS_IFLOOP( 3 )
	    *--op = ip[i]<<(b-24) | ip[i+1]<<(b-16) | ip[i+2]<<(b-8);
	GA_URS_IFLOOP( 4 )
	    *--op = ip[i]<<(b-32) | ip[i+1]<<(b-24) | ip[i+2]<<(b-16) | ip[i+3] << (b-8);
    } else {
	GA_URS_IFLOOP( 2 )
	    *--op = ip[i]<<(b-8) | ip[i+1]<<(b-16); 
	GA_URS_IFLOOP( 3 )
	    *--op = ip[i]<<(b-8) | ip[i+1]<<(b-16) | ip[i+2]<<(b-24);
	GA_URS_IFLOOP( 4 )
	    *--op = ip[i]<<(b-8) | ip[i+1]<<(b-16) | ip[i+2]<<(b-24) | ip[i+3]<<(b-32);
    }
#undef GA_URS_IFLOOP
    return( samples_read );
}



/************************************************************************
*
* read_samples()
*
* PURPOSE:  reads the PCM samples from a file to the buffer
*
*  SEMANTICS:
* Reads #samples_read# number of shorts from #musicin# filepointer
* into #sample_buffer[]#.  Returns the number of samples read.
*
************************************************************************/

int
read_samples_pcm(FILE * musicin, int sample_buffer[2304], int frame_size,
                 int samples_to_read)
{
    int     samples_read;
    int     iswav = (input_format == sf_wave);
    int     hi_lo_order;	/* byte order of input stream */

    if( (32 == pcmbitwidth) || (24 == pcmbitwidth) || (16 == pcmbitwidth) ) {
				/* assume only recognized wav files are */
				/*  in little endian byte order */
	hi_lo_order = (!iswav == !swapbytes);
        samples_read = unpack_read_samples(samples_to_read, pcmbitwidth/8, 
                                           hi_lo_order,sample_buffer, musicin );
       
    } else if( 8 == pcmbitwidth ) {
	samples_read = unpack_read_samples( samples_to_read, 1, 0,
					    sample_buffer, musicin );
    } else {
        fprintf(stderr, "Only 8, 16, 24 and 32 bit input files supported \n");
        exit(1);
    }
    if (ferror(musicin)) {
        fprintf(stderr, "Error reading input file\n");
        exit(1);
    }

    return samples_read;
}



/* AIFF Definitions */

#define IFF_ID_FORM 0x464f524d /* "FORM" */
#define IFF_ID_AIFF 0x41494646 /* "AIFF" */
#define IFF_ID_AIFC 0x41494643 /* "AIFC" */
#define IFF_ID_COMM 0x434f4d4d /* "COMM" */
#define IFF_ID_SSND 0x53534e44 /* "SSND" */
#define IFF_ID_MPEG 0x4d504547 /* "MPEG" */

#define IFF_ID_NONE 0x4e4f4e45 /* "NONE" */ /* AIFF-C data format */
#define IFF_ID_2CBE 0x74776f73 /* "twos" */ /* AIFF-C data format */
#define IFF_ID_2CLE 0x736f7774 /* "sowt" */ /* AIFF-C data format */

#define WAV_ID_RIFF 0x52494646 /* "RIFF" */
#define WAV_ID_WAVE 0x57415645 /* "WAVE" */
#define WAV_ID_FMT  0x666d7420 /* "fmt " */
#define WAV_ID_DATA 0x64617461 /* "data" */




/*****************************************************************************
 *
 *	Read Microsoft Wave headers
 *
 *	By the time we get here the first 32-bits of the file have already been
 *	read, and we're pretty sure that we're looking at a WAV file.
 *
 *****************************************************************************/

static int
parse_wave_header(lame_global_flags * gfp, FILE * sf)
{
    int     format_tag = 0;
    int     channels = 0;
    int     block_align = 0;
    int     bits_per_sample = 0;
    int     samples_per_sec = 0;
    int     avg_bytes_per_sec = 0;


    int     is_wav = 0;
    long    data_length = 0, file_length, subSize = 0;
    int     loop_sanity = 0;

    file_length = Read32BitsHighLow(sf);

    if (Read32BitsHighLow(sf) != WAV_ID_WAVE)
        return 0;

    for (loop_sanity = 0; loop_sanity < 20; ++loop_sanity) {
        int     type = Read32BitsHighLow(sf);

        if (type == WAV_ID_FMT) {
            subSize = Read32BitsLowHigh(sf);
            if (subSize < 16) {
                /*DEBUGF(
                   "'fmt' chunk too short (only %ld bytes)!", subSize);  */
                return 0;
            }

            format_tag = Read16BitsLowHigh(sf);
            subSize -= 2;
            channels = Read16BitsLowHigh(sf);
            subSize -= 2;
            samples_per_sec = Read32BitsLowHigh(sf);
            subSize -= 4;
            avg_bytes_per_sec = Read32BitsLowHigh(sf);
            subSize -= 4;
            block_align = Read16BitsLowHigh(sf);
            subSize -= 2;
            bits_per_sample = Read16BitsLowHigh(sf);
            subSize -= 2;

            /* DEBUGF("   skipping %d bytes\n", subSize); */

            if (subSize > 0) {
                if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
                    return 0;
            };

        }
        else if (type == WAV_ID_DATA) {
            subSize = Read32BitsLowHigh(sf);
            data_length = subSize;
            is_wav = 1;
            /* We've found the audio data. Read no further! */
            break;

        }
        else {
            subSize = Read32BitsLowHigh(sf);
            if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
                return 0;
        }
    }

    if (format_tag != 1) {
	return 0; /* oh no! non-supported format  */
    }


    if (is_wav) {
        /* make sure the header is sane */
        if( -1 == lame_set_num_channels( gfp, channels ) ) {
            fprintf( stderr,
                     "Unsupported number of channels: %ud\n",
                     channels );
            exit( 1 );
        }
        (void) lame_set_in_samplerate( gfp, samples_per_sec );
        pcmbitwidth = bits_per_sample;
        (void) lame_set_num_samples( gfp,
            data_length / (channels * ((bits_per_sample+7) / 8)) );
    }
    return is_wav;
}



/************************************************************************
* aiff_check2
*
* PURPOSE:	Checks AIFF header information to make sure it is valid.
*	        returns 0 on success, 1 on errors
************************************************************************/

int
aiff_check2(const char *file_name, IFF_AIFF * const pcm_aiff_data)
{
    if (pcm_aiff_data->sampleType != IFF_ID_SSND) {
        fprintf(stderr, "Sound data is not PCM in '%s'\n", file_name);
        return 1;
    }
    if (pcm_aiff_data->sampleSize != sizeof(short) * CHAR_BIT) {
        fprintf(stderr, "Sound data is not %i bits in '%s'\n",
                sizeof(short) * CHAR_BIT, file_name);
        return 1;
    }
    if (pcm_aiff_data->numChannels != 1 && pcm_aiff_data->numChannels != 2) {
        fprintf(stderr, "Sound data is not mono or stereo in '%s'\n",
                file_name);
        return 1;
    }
    if (pcm_aiff_data->blkAlgn.blockSize != 0) {
        fprintf(stderr, "Block size is not 0 bytes in '%s'\n", file_name);
        return 1;
    }
    /* A bug, since we correctly skip the offset earlier in the code.
    if (pcm_aiff_data->blkAlgn.offset != 0) {
        fprintf(stderr, "Block offset is not 0 bytes in '%s'\n", file_name);
        return 1;
    } */

    return 0;
}

/*****************************************************************************
 *
 *	Read Audio Interchange File Format (AIFF) headers.
 *
 *	By the time we get here the first 32 bits of the file have already been
 *	read, and we're pretty sure that we're looking at an AIFF file.
 *
 *****************************************************************************/

static int
parse_aiff_header(lame_global_flags * gfp, FILE * sf)
{
    int     is_aiff = 0;
    long    chunkSize = 0, subSize = 0, typeID = 0, dataType = 0;
    IFF_AIFF aiff_info;

    memset(&aiff_info, 0, sizeof(aiff_info));
    chunkSize = Read32BitsHighLow(sf);

    typeID = Read32BitsHighLow(sf);
    if ((typeID != IFF_ID_AIFF)&&(typeID != IFF_ID_AIFC))
        return 0;

    while (chunkSize > 0) {
        int     type = Read32BitsHighLow(sf);
        chunkSize -= 4;

        /* DEBUGF(
           "found chunk type %08x '%4.4s'\n", type, (char*)&type); */

        /* don't use a switch here to make it easier to use 'break' for SSND */
        if (type == IFF_ID_COMM) {
            subSize = Read32BitsHighLow(sf);
            chunkSize -= subSize;

            aiff_info.numChannels = Read16BitsHighLow(sf);
            subSize -= 2;
            aiff_info.numSampleFrames = Read32BitsHighLow(sf);
            subSize -= 4;
            aiff_info.sampleSize = Read16BitsHighLow(sf);
            subSize -= 2;
            aiff_info.sampleRate = ReadIeeeExtendedHighLow(sf);
            subSize -= 10;

            if (typeID == IFF_ID_AIFC) {
                dataType = Read32BitsHighLow(sf);
                subSize -= 4;

                if ((dataType != IFF_ID_2CLE) && 
                    (dataType != IFF_ID_2CBE) &&
                    (dataType != IFF_ID_NONE))
                    return 0;
                    
                if (aiff_info.sampleSize == 16)
                    swapbytes = (!swapbytes == (dataType == IFF_ID_2CLE));
            }
            
            if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
                return 0;
        }
        else if (type == IFF_ID_SSND) {
            subSize = Read32BitsHighLow(sf);
            chunkSize -= subSize;

            aiff_info.blkAlgn.offset = Read32BitsHighLow(sf);
            subSize -= 4;
            aiff_info.blkAlgn.blockSize = Read32BitsHighLow(sf);
            subSize -= 4;

            if (fskip(sf, (long) aiff_info.blkAlgn.offset, SEEK_CUR) != 0)
                return 0;

            aiff_info.sampleType = IFF_ID_SSND;
            is_aiff = 1;

            /* We've found the audio data. Read no further! */
            break;

        }
        else {
            subSize = Read32BitsHighLow(sf);
            chunkSize -= subSize;

            if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
                return 0;
        }
    }

    /* DEBUGF("Parsed AIFF %d\n", is_aiff); */
    if (is_aiff) {
        /* make sure the header is sane */
        if (0 != aiff_check2("name" /*???????????? */ , &aiff_info))
            return 0;
        if( -1 == lame_set_num_channels( gfp, aiff_info.numChannels ) ) {
            fprintf( stderr,
                     "Unsupported number of channels: %ud\n",
                     aiff_info.numChannels );
            exit( 1 );
        }
        (void) lame_set_in_samplerate( gfp, aiff_info.sampleRate );
        pcmbitwidth = aiff_info.sampleSize;
        (void) lame_set_num_samples( gfp, aiff_info.numSampleFrames );
    }
    return is_aiff;
}



/************************************************************************
*
* parse_file_header
*
* PURPOSE: Read the header from a bytestream.  Try to determine whether
*		   it's a WAV file or AIFF without rewinding, since rewind
*		   doesn't work on pipes and there's a good chance we're reading
*		   from stdin (otherwise we'd probably be using libsndfile).
*
* When this function returns, the file offset will be positioned at the
* beginning of the sound data.
*
************************************************************************/

void
parse_file_header(lame_global_flags * gfp, FILE * sf)
{

    int     type = Read32BitsHighLow(sf);
    /*
       DEBUGF(
       "First word of input stream: %08x '%4.4s'\n", type, (char*) &type); 
     */
    count_samples_carefully = 0;
    input_format = sf_raw;

    if (type == WAV_ID_RIFF) {
        /* It's probably a WAV file */
        if (parse_wave_header(gfp, sf)) {
            input_format = sf_wave;
            count_samples_carefully = 1;
        } else {
	    fprintf( stderr, "Warning: corrupt or unsupported WAVE format\n"); 
        }
    }
    else if (type == IFF_ID_FORM) {
        /* It's probably an AIFF file */
        if (parse_aiff_header(gfp, sf)) {
            input_format = sf_aiff;
            count_samples_carefully = 1;
        }
    }
    if (input_format == sf_raw) {
        /*
           ** Assume it's raw PCM.  Since the audio data is assumed to begin
           ** at byte zero, this will unfortunately require seeking.
         */
        if (fseek(sf, 0L, SEEK_SET) != 0) {
            /* ignore errors */
        }
        input_format = sf_raw;
    }
}



void
CloseSndFile(sound_file_format input, FILE * musicin)
{
    if (fclose(musicin) != 0) {
        fprintf(stderr, "Could not close audio input file\n");
        exit(2);
    }
}





FILE   *
OpenSndFile(lame_global_flags * gfp, char *inPath)
{
    FILE   *musicin;

    /* set the defaults from info incase we cannot determine them from file */
    lame_set_num_samples( gfp, MAX_U_32_NUM );


    if (!strcmp(inPath, "-")) {
        lame_set_stream_binary_mode(musicin = stdin); /* Read from standard input. */
    }
    else {
        if ((musicin = fopen(inPath, "rb")) == NULL) {
            fprintf(stderr, "Could not find \"%s\".\n", inPath);
            exit(1);
        }
    }

    if (input_format == sf_mp1 ||
        input_format == sf_mp2 || input_format == sf_mp3) {
#ifdef AMIGA_MPEGA
        if (-1 == lame_decode_initfile(inPath, &mp3input_data)) {
            fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
                    inPath);
            exit(1);
        }
#endif
#ifdef HAVE_MPGLIB
        if (-1 == lame_decode_initfile(musicin, &mp3input_data)) {
            fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
                    inPath);
            exit(1);
        }
#endif
        if( -1 == lame_set_num_channels( gfp, mp3input_data.stereo ) ) {
            fprintf( stderr,
                     "Unsupported number of channels: %ud\n",
                     mp3input_data.stereo );
            exit( 1 );
        }
        (void) lame_set_in_samplerate( gfp, mp3input_data.samplerate );
        (void) lame_set_num_samples( gfp, mp3input_data.nsamp );
    }
    else if (input_format == sf_ogg) {
#ifdef HAVE_VORBIS
        if ( -1 == lame_decode_ogg_initfile( gfp,
                                             musicin,
                                             &mp3input_data ) ) {
            fprintf(stderr, "Error reading headers in ogg input file %s.\n",
                    inPath);
            exit(1);
        }
        if( -1 == lame_set_num_channels( gfp, mp3input_data.stereo ) ) {
            fprintf( stderr,
                     "Unsupported number of channels: %ud\n",
                     mp3input_data.stereo );
            exit( 1 );
        }
        (void) lame_set_in_samplerate( gfp, mp3input_data.samplerate );
        (void) lame_set_num_samples( gfp, mp3input_data.nsamp );
#else
        fprintf(stderr, "LAME not compiled with libvorbis support.\n");
        exit(1);
#endif
    }
    else {
        if (input_format != sf_raw) {
            parse_file_header(gfp, musicin);
        }

        if (input_format == sf_raw) {
            /* assume raw PCM */
            fprintf(stderr, "Assuming raw pcm input file");
            if (swapbytes)
                fprintf(stderr, " : Forcing byte-swapping\n");
            else
                fprintf(stderr, "\n");
        }
    }


    if (lame_get_num_samples( gfp ) == MAX_U_32_NUM && musicin != stdin) {

        double  flen = lame_get_file_size(inPath); /* try to figure out num_samples */

        if (flen >= 0) {
            /* try file size, assume 2 bytes per sample */
            if (input_format == sf_mp1 ||
                input_format == sf_mp2 || input_format == sf_mp3) {

                if (mp3input_data.bitrate > 0) {
                    double  totalseconds =
                        (flen * 8.0 / (1000.0 * mp3input_data.bitrate));
                    unsigned long tmp_num_samples =
                        totalseconds * lame_get_in_samplerate( gfp );

                    (void) lame_set_num_samples( gfp, tmp_num_samples );
                    mp3input_data.nsamp = tmp_num_samples;
                }
            }
            else {
                (void) lame_set_num_samples( gfp,
                    flen / (2 * lame_get_num_channels( gfp )) );
            }
        }
    }
    return musicin;
}
#endif /* defined(LIBSNDFILE) */





#if defined(HAVE_MPGLIB)
static int
check_aid(const unsigned char *header)
{
    return 0 == strncmp(header, "AiD\1", 4);
}

/*
 * Please check this and don't kill me if there's a bug
 * This is a (nearly?) complete header analysis for a MPEG-1/2/2.5 Layer I, II or III
 * data stream
 */

static int
is_syncword_mp123(const void *const headerptr)
{
    const unsigned char *const p = headerptr;
    static const char abl2[16] =
        { 0, 7, 7, 7, 0, 7, 0, 0, 0, 0, 0, 8, 8, 8, 8, 8 };

    if ((p[0] & 0xFF) != 0xFF)
        return 0;       // first 8 bits must be '1'
    if ((p[1] & 0xE0) != 0xE0)
        return 0;       // next 3 bits are also
    if ((p[1] & 0x18) == 0x08)
        return 0;       // no MPEG-1, -2 or -2.5
    if ((p[1] & 0x06) == 0x00)
        return 0;       // no Layer I, II and III
    if ((p[2] & 0xF0) == 0xF0)
        return 0;       // bad bitrate
    if ((p[2] & 0x0C) == 0x0C)
        return 0;       // no sample frequency with (32,44.1,48)/(1,2,4)    
    if ((p[1] & 0x06) == 0x04) // illegal Layer II bitrate/Channel Mode comb
        if (abl2[p[2] >> 4] & (1 << (p[3] >> 6)))
            return 0;
    return 1;
}

static int
is_syncword_mp3(const void *const headerptr)
{
    const unsigned char *const p = headerptr;

    if ((p[0] & 0xFF) != 0xFF)
        return 0;       // first 8 bits must be '1'
    if ((p[1] & 0xE0) != 0xE0)
        return 0;       // next 3 bits are also
    if ((p[1] & 0x18) == 0x08)
        return 0;       // no MPEG-1, -2 or -2.5
    if ((p[1] & 0x06) != 0x02)
        return 0;       // no Layer III (can be merged with 'next 3 bits are also' test, but don't do this, this decreases readability)
    if ((p[2] & 0xF0) == 0xF0)
        return 0;       // bad bitrate
    if ((p[2] & 0x0C) == 0x0C)
        return 0;       // no sample frequency with (32,44.1,48)/(1,2,4)    
    return 1;
}


int
lame_decode_initfile(FILE * fd, mp3data_struct * mp3data)
{
    //  VBRTAGDATA pTagData;
    // int xing_header,len2,num_frames;
    unsigned char buf[100];
    int     ret;
    int     len, aid_header;
    short int pcm_l[1152], pcm_r[1152];
    

    memset(mp3data, 0, sizeof(mp3data_struct));
    lame_decode_init();

    len = 4;
    if (fread(&buf, 1, len, fd) != len)
        return -1;      /* failed */
    aid_header = check_aid(buf);
    if (aid_header) {
        if (fread(&buf, 1, 2, fd) != 2)
            return -1;  /* failed */
        aid_header = (unsigned char) buf[0] + 256 * (unsigned char) buf[1];
        fprintf(stderr, "Album ID found.  length=%i \n", aid_header);
        /* skip rest of AID, except for 6 bytes we have already read */
        fskip(fd, aid_header - 6, SEEK_CUR);

        /* read 4 more bytes to set up buffer for MP3 header check */
        len = fread(&buf, 1, 4, fd);
    }


    /* look for valid 4 byte MPEG header  */
    if (len < 4)
        return -1;
    while (!is_syncword_mp123(buf)) {
        int     i;
        for (i = 0; i < len - 1; i++)
            buf[i] = buf[i + 1];
        if (fread(buf + len - 1, 1, 1, fd) != 1)
            return -1;  /* failed */
    }


    // now parse the current buffer looking for MP3 headers.   
    // (as of 11/00: mpglib modified so that for the first frame where 
    // headers are parsed, no data will be decoded.  
    // However, for freeformat, we need to decode an entire frame,
    // so mp3data->bitrate will be 0 until we have decoded the first
    // frame.  Cannot decode first frame here because we are not
    // yet prepared to handle the output.
    ret = lame_decode1_headersB(buf, len, pcm_l, pcm_r, mp3data,&enc_delay,&enc_padding);
    if (-1 == ret)
        return -1;

    /* repeat until we decode a valid mp3 header.  */
    while (!mp3data->header_parsed) {
        len = fread(buf, 1, sizeof(buf), fd);
        if (len != sizeof(buf))
            return -1;
        ret = lame_decode1_headersB(buf, len, pcm_l, pcm_r, mp3data,&enc_delay,&enc_padding);
        if (-1 == ret)
            return -1;
    }

    if (mp3data->bitrate==0) {
	fprintf(stderr,"Input file is freeformat.\n"); 
    }

    if (mp3data->totalframes > 0) {
        /* mpglib found a Xing VBR header and computed nsamp & totalframes */
    }
    else {
	/* set as unknown.  Later, we will take a guess based on file size
	 * ant bitrate */
        mp3data->nsamp = MAX_U_32_NUM;
    }


    /*
       fprintf(stderr,"ret = %i NEED_MORE=%i \n",ret,MP3_NEED_MORE);
       fprintf(stderr,"stereo = %i \n",mp.fr.stereo);
       fprintf(stderr,"samp = %i  \n",freqs[mp.fr.sampling_frequency]);
       fprintf(stderr,"framesize = %i  \n",framesize);
       fprintf(stderr,"bitrate = %i  \n",mp3data->bitrate);
       fprintf(stderr,"num frames = %ui  \n",num_frames);
       fprintf(stderr,"num samp = %ui  \n",mp3data->nsamp);
       fprintf(stderr,"mode     = %i  \n",mp.fr.mode);
     */

    return 0;
}

/*
For lame_decode_fromfile:  return code
  -1     error
   n     number of samples output.  either 576 or 1152 depending on MP3 file.


For lame_decode1_headers():  return code
  -1     error
   0     ok, but need more data before outputing any samples
   n     number of samples output.  either 576 or 1152 depending on MP3 file.
*/
int
lame_decode_fromfile(FILE * fd, short pcm_l[], short pcm_r[],
                     mp3data_struct * mp3data)
{
    int     ret = 0, len=0;
    unsigned char buf[1024];

    /* first see if we still have data buffered in the decoder: */
    ret = lame_decode1_headers(buf, len, pcm_l, pcm_r, mp3data);
    if (ret!=0) return ret;


    /* read until we get a valid output frame */
    while (1) {
        len = fread(buf, 1, 1024, fd);
        if (len == 0) {
	    /* we are done reading the file, but check for buffered data */
	    ret = lame_decode1_headers(buf, len, pcm_l, pcm_r, mp3data);
	    if (ret<=0) return -1;  // done with file
	    break;
	}

        ret = lame_decode1_headers(buf, len, pcm_l, pcm_r, mp3data);
        if (ret == -1) return -1;
	if (ret >0) break;
    }
    return ret;
}
#endif /* defined(HAVE_MPGLIB) */

/* end of get_audio.c */
