#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "common.h"
#include "encoder.h"
#include "musicin.h"
#include "options.h"
#include "audio_read.h"
#include "bitstream.h"
#include "mem.h"
#include "crc.h"
#include "psycho_n1.h"
#include "psycho_0.h"
#include "psycho_1.h"
#include "psycho_2.h"
#include "psycho_3.h"
#include "psycho_4.h"
#include "encode.h"
#include "availbits.h"
#include "subband.h"
#include "encode_new.h"
#include "toolame.h"
#include "bwf_wavefile.h"

#include <assert.h>

FILE *musicin;
Bit_stream_struc bs;
char *programName;
char toolameversion[10] = "0.2l";

void global_init (void)
{
  glopts.usepsy = TRUE;
  glopts.usepadbit = TRUE;
  glopts.quickmode = FALSE;
  glopts.quickcount = 10;
  glopts.downmix = FALSE;
  glopts.byteswap = FALSE;
  glopts.channelswap = FALSE;
  glopts.vbr = FALSE;
  glopts.vbrlevel = 0;
  glopts.athlevel = 0;
  glopts.verbosity = 2;
  glopts.outputformat = 0;
}

/************************************************************************
*
* main
*
* PURPOSE:  MPEG II Encoder with
* psychoacoustic models 1 (MUSICAM) and 2 (AT&T)
*
* SEMANTICS:  One overlapping frame of audio of up to 2 channels are
* processed at a time in the following order:
* (associated routines are in parentheses)
*
* 1.  Filter sliding window of data to get 32 subband
* samples per channel.
* (window_subband,filter_subband)
*
* 2.  If joint stereo mode, combine left and right channels
* for subbands above #jsbound#.
* (combine_LR)
*
* 3.  Calculate scalefactors for the frame, and 
* also calculate scalefactor select information.
* (*_scale_factor_calc)
*
* 4.  Calculate psychoacoustic masking levels using selected
* psychoacoustic model.
* (psycho_i, psycho_ii)
*
* 5.  Perform iterative bit allocation for subbands with low
* mask_to_noise ratios using masking levels from step 4.
* (*_main_bit_allocation)
*
* 6.  If error protection flag is active, add redundancy for
* error protection.
* (*_CRC_calc)
*
* 7.  Pack bit allocation, scalefactors, and scalefactor select
*headerrmation onto bitstream.
* (*_encode_bit_alloc,*_encode_scale,transmission_pattern)
*
* 8.  Quantize subbands and pack them into bitstream
* (*_subband_quantization, *_sample_encoding)
*
************************************************************************/

int frameNum = 0;

int main (int argc, char **argv)
{
  typedef double SBS[2][3][SCALE_BLOCK][SBLIMIT];
  SBS *sb_sample;
  typedef double JSBS[3][SCALE_BLOCK][SBLIMIT];
  JSBS *j_sample;
  typedef double IN[2][HAN_SIZE];
  IN *win_que;
  typedef unsigned int SUB[2][3][SCALE_BLOCK][SBLIMIT];
  SUB *subband;

  frame_info frame;
  frame_header header;
  char original_file_name[MAX_NAME_SIZE];
  char encoded_file_name[MAX_NAME_SIZE];
  short **win_buf;
  static short buffer[2][1152];
  static unsigned int bit_alloc[2][SBLIMIT], scfsi[2][SBLIMIT];
  static unsigned int scalar[2][3][SBLIMIT], j_scale[3][SBLIMIT];
  static double smr[2][SBLIMIT], lgmin[2][SBLIMIT], max_sc[2][SBLIMIT];
  // FLOAT snr32[32];
  short sam[2][1344];		/* was [1056]; */
  int model, nch, error_protection;
  static unsigned int crc;
  int sb, ch, adb;
  unsigned long frameBits, sentBits = 0;
  unsigned long num_samples;
  int lg_frame;
  int i;

  /* Used to keep the SNR values for the fast/quick psy models */
  static FLOAT smrdef[2][32];

  static int psycount = 0;
  extern int minimum;

  sb_sample = (SBS *) mem_alloc (sizeof (SBS), "sb_sample");
  j_sample = (JSBS *) mem_alloc (sizeof (JSBS), "j_sample");
  win_que = (IN *) mem_alloc (sizeof (IN), "Win_que");
  subband = (SUB *) mem_alloc (sizeof (SUB), "subband");
  win_buf = (short **) mem_alloc (sizeof (short *) * 2, "win_buf");

  /* clear buffers */
  memset ((char *) buffer, 0, sizeof (buffer));
  memset ((char *) bit_alloc, 0, sizeof (bit_alloc));
  memset ((char *) scalar, 0, sizeof (scalar));
  memset ((char *) j_scale, 0, sizeof (j_scale));
  memset ((char *) scfsi, 0, sizeof (scfsi));
  memset ((char *) smr, 0, sizeof (smr));
  memset ((char *) lgmin, 0, sizeof (lgmin));
  memset ((char *) max_sc, 0, sizeof (max_sc));
  //memset ((char *) snr32, 0, sizeof (snr32));
  memset ((char *) sam, 0, sizeof (sam));

  global_init ();
  
  header.extension = 0;
  frame.header = &header;
  frame.tab_num = -1;		/* no table loaded */
  frame.alloc = NULL;
  header.version = MPEG_AUDIO_ID;	/* Default: MPEG-1 */

  programName = argv[0];
  if (argc == 1)		/* no command-line args */
    short_usage ();
  else
    parse_args (argc, argv, &frame, &model, &num_samples, original_file_name,
		encoded_file_name);
  print_config (&frame, &model, original_file_name, encoded_file_name);

  /* this will load the alloc tables and do some other stuff */
  hdr_to_frps (&frame);
  nch = frame.nch;
  error_protection = header.error_protection;

  while (get_audio (musicin, buffer, num_samples, nch, &header) > 0) {
    if (glopts.verbosity > 1)
      if (++frameNum % 10 == 0)
	fprintf (stderr, "[%4u]\r", frameNum);
    fflush (stderr);
    win_buf[0] = &buffer[0][0];
    win_buf[1] = &buffer[1][0];

    adb = available_bits (&header, &glopts);
    lg_frame = adb / 8;
    if (header.dab_extension) {
      /* in 24 kHz we always have 4 bytes */
      if (header.sampling_frequency == 1)
	header.dab_extension = 4;
/* You must have one frame in memory if you are in DAB mode                 */
/* in conformity of the norme ETS 300 401 http://www.etsi.org               */
      /* see bitstream.c            */
      if (frameNum == 1)
	minimum = lg_frame + MINIMUM;
      adb -= header.dab_extension * 8 + header.dab_length * 8 + 16;
    }

    {
      int gr, bl, ch;
      /* New polyphase filter
	 Combines windowing and filtering. Ricardo Feb'03 */
      for( gr = 0; gr < 3; gr++ )
	for ( bl = 0; bl < 12; bl++ )
	  for ( ch = 0; ch < nch; ch++ )
	    WindowFilterSubband( &buffer[ch][gr * 12 * 32 + 32 * bl], ch,
				 &(*sb_sample)[ch][gr][bl][0] );
    }

#ifdef REFERENCECODE
    {
      /* Old code. left here for reference */
      int gr, bl, ch;
      for (gr = 0; gr < 3; gr++)
	for (bl = 0; bl < SCALE_BLOCK; bl++)
	  for (ch = 0; ch < nch; ch++) {
	    window_subband (&win_buf[ch], &(*win_que)[ch][0], ch);
	    filter_subband (&(*win_que)[ch][0], &(*sb_sample)[ch][gr][bl][0]);
	  }
    }
#endif


#ifdef NEWENCODE
    scalefactor_calc_new(*sb_sample, scalar, nch, frame.sblimit);
    find_sf_max (scalar, &frame, max_sc);
    if (frame.actual_mode == MPG_MD_JOINT_STEREO) {
      /* this way we calculate more mono than we need */
      /* but it is cheap */
      combine_LR_new (*sb_sample, *j_sample, frame.sblimit);
      scalefactor_calc_new (j_sample, &j_scale, 1, frame.sblimit);
    }
#else
    scale_factor_calc (*sb_sample, scalar, nch, frame.sblimit);
    pick_scale (scalar, &frame, max_sc);
    if (frame.actual_mode == MPG_MD_JOINT_STEREO) {
      /* this way we calculate more mono than we need */
      /* but it is cheap */
      combine_LR (*sb_sample, *j_sample, frame.sblimit);
      scale_factor_calc (j_sample, &j_scale, 1, frame.sblimit);
    }
#endif



    if ((glopts.quickmode == TRUE) && (++psycount % glopts.quickcount != 0)) {
      /* We're using quick mode, so we're only calculating the model every
         'quickcount' frames. Otherwise, just copy the old ones across */
      for (ch = 0; ch < nch; ch++) {
	for (sb = 0; sb < SBLIMIT; sb++)
	  smr[ch][sb] = smrdef[ch][sb];
      }
    } else {
      /* calculate the psymodel */
      switch (model) {
      case -1:
	psycho_n1 (smr, nch);
	break;
      case 0:	/* Psy Model A */
	psycho_0 (smr, nch, scalar, (FLOAT) s_freq[header.version][header.sampling_frequency] * 1000);	
	break;
      case 1:
	psycho_1 (buffer, max_sc, smr, &frame);
	break;
      case 2:
	for (ch = 0; ch < nch; ch++) {
	  psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
		     (FLOAT) s_freq[header.version][header.sampling_frequency] *
		     1000, &glopts);
	}
	break;
      case 3:
	/* Modified psy model 1 */
	psycho_3 (buffer, max_sc, smr, &frame, &glopts);
	break;
      case 4:
	/* Modified Psycho Model 2 */
	for (ch = 0; ch < nch; ch++) {
	  psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
		     (FLOAT) s_freq[header.version][header.sampling_frequency] *
		     1000, &glopts);
	}
	break;	
      case 5:
	/* Model 5 comparse model 1 and 3 */
	psycho_1 (buffer, max_sc, smr, &frame);
	fprintf(stdout,"1 ");
	smr_dump(smr,nch);
	psycho_3 (buffer, max_sc, smr, &frame, &glopts);
	fprintf(stdout,"3 ");
	smr_dump(smr,nch);
	break;
      case 6:
	/* Model 6 compares model 2 and 4 */
	for (ch = 0; ch < nch; ch++) 
	  psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
		    (FLOAT) s_freq[header.version][header.sampling_frequency] *
		    1000, &glopts);
	fprintf(stdout,"2 ");
	smr_dump(smr,nch);
	for (ch = 0; ch < nch; ch++) 
	  psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
		     (FLOAT) s_freq[header.version][header.sampling_frequency] *
		     1000, &glopts);
	fprintf(stdout,"4 ");
	smr_dump(smr,nch);
	break;
      case 7:
	fprintf(stdout,"Frame: %i\n",frameNum);
	/* Dump the SMRs for all models */	
	psycho_1 (buffer, max_sc, smr, &frame);
	fprintf(stdout,"1");
	smr_dump(smr, nch);
	psycho_3 (buffer, max_sc, smr, &frame, &glopts);
	fprintf(stdout,"3");
	smr_dump(smr,nch);
	for (ch = 0; ch < nch; ch++) 
	  psycho_2 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], //snr32,
		    (FLOAT) s_freq[header.version][header.sampling_frequency] *
		    1000, &glopts);
	fprintf(stdout,"2");
	smr_dump(smr,nch);
	for (ch = 0; ch < nch; ch++) 
	  psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
		     (FLOAT) s_freq[header.version][header.sampling_frequency] *
		     1000, &glopts);
	fprintf(stdout,"4");
	smr_dump(smr,nch);
	break;
      case 8:
	/* Compare 0 and 4 */	
	psycho_n1 (smr, nch);
	fprintf(stdout,"0");
	smr_dump(smr,nch);

	for (ch = 0; ch < nch; ch++) 
	  psycho_4 (&buffer[ch][0], &sam[ch][0], ch, &smr[ch][0], // snr32,
		     (FLOAT) s_freq[header.version][header.sampling_frequency] *
		     1000, &glopts);
	fprintf(stdout,"4");
	smr_dump(smr,nch);
	break;
      default:
	fprintf (stderr, "Invalid psy model specification: %i\n", model);
	exit (0);
      }

      if (glopts.quickmode == TRUE)
	/* copy the smr values and reuse them later */
	for (ch = 0; ch < nch; ch++) {
	  for (sb = 0; sb < SBLIMIT; sb++)
	    smrdef[ch][sb] = smr[ch][sb];
	}

      if (glopts.verbosity > 4) 
	smr_dump(smr, nch);
     
      


    }

#ifdef NEWENCODE
    sf_transmission_pattern (scalar, scfsi, &frame);
    main_bit_allocation_new (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
    //main_bit_allocation (smr, scfsi, bit_alloc, &adb, &frame, &glopts);

    if (error_protection)
      CRC_calc (&frame, bit_alloc, scfsi, &crc);

    write_header (&frame, &bs);
    //encode_info (&frame, &bs);
    if (error_protection)
      putbits (&bs, crc, 16);
    write_bit_alloc (bit_alloc, &frame, &bs);
    //encode_bit_alloc (bit_alloc, &frame, &bs);
    write_scalefactors(bit_alloc, scfsi, scalar, &frame, &bs);
    //encode_scale (bit_alloc, scfsi, scalar, &frame, &bs);
    subband_quantization_new (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
    			  *subband, &frame);
    //subband_quantization (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
    //	  *subband, &frame);
    write_samples_new(*subband, bit_alloc, &frame, &bs);
    //sample_encoding (*subband, bit_alloc, &frame, &bs);
#else
    transmission_pattern (scalar, scfsi, &frame);
    main_bit_allocation (smr, scfsi, bit_alloc, &adb, &frame, &glopts);
    if (error_protection)
      CRC_calc (&frame, bit_alloc, scfsi, &crc);
    encode_info (&frame, &bs);
    if (error_protection)
      encode_CRC (crc, &bs);
    encode_bit_alloc (bit_alloc, &frame, &bs);
    encode_scale (bit_alloc, scfsi, scalar, &frame, &bs);
    subband_quantization (scalar, *sb_sample, j_scale, *j_sample, bit_alloc,
			  *subband, &frame);
    sample_encoding (*subband, bit_alloc, &frame, &bs);
#endif


    /* If not all the bits were used, write out a stack of zeros */
    for (i = 0; i < adb; i++)
      put1bit (&bs, 0);
    if (header.dab_extension) {
      /* Reserve some bytes for X-PAD in DAB mode */
      putbits (&bs, 0, header.dab_length * 8);
      
      for (i = header.dab_extension - 1; i >= 0; i--) {
	CRC_calcDAB (&frame, bit_alloc, scfsi, scalar, &crc, i);
	/* this crc is for the previous frame in DAB mode  */
	if (bs.buf_byte_idx + lg_frame < bs.buf_size)
	  bs.buf[bs.buf_byte_idx + lg_frame] = crc;
	/* reserved 2 bytes for F-PAD in DAB mode  */
	putbits (&bs, crc, 8);
      }
      putbits (&bs, 0, 16);
    }

    frameBits = sstell (&bs) - sentBits;

    if (frameBits % 8) {	/* a program failure */
      fprintf (stderr, "Sent %ld bits = %ld slots plus %ld\n", frameBits,
	       frameBits / 8, frameBits % 8);
      fprintf (stderr, "If you are reading this, the program is broken\n");
      fprintf (stderr, "email [mfc at NOTplanckenerg.com] without the NOT\n");
      fprintf (stderr, "with the command line arguments and other info\n");
      exit (0);
    }

    sentBits += frameBits;
  }

  close_bit_stream_w (&bs);

  if ((glopts.verbosity > 1) && (glopts.vbr == TRUE)) {
    int i;
#ifdef NEWENCODE
    extern int vbrstats_new[15];
#else
    extern int vbrstats[15];
#endif
    fprintf (stdout, "VBR stats:\n");
    for (i = 1; i < 15; i++)
      fprintf (stdout, "%4i ", bitrate[header.version][i]);
    fprintf (stdout, "\n");
    for (i = 1; i < 15; i++)
#ifdef NEWENCODE
      fprintf (stdout,"%4i ",vbrstats_new[i]);
#else
      fprintf (stdout, "%4i ", vbrstats[i]);
#endif
    fprintf (stdout, "\n");
  }

  fprintf (stderr,
	   "Avg slots/frame = %.3f; b/smp = %.2f; bitrate = %.3f kbps\n",
	   (FLOAT) sentBits / (frameNum * 8),
	   (FLOAT) sentBits / (frameNum * 1152),
	   (FLOAT) sentBits / (frameNum * 1152) *
	   s_freq[header.version][header.sampling_frequency]);

  if (fclose (musicin) != 0) {
    fprintf (stderr, "Could not close \"%s\".\n", original_file_name);
    exit (2);
  }

  fprintf (stderr, "\nDone\n");
  exit (0);
}

/************************************************************************
*
* print_config
*
* PURPOSE:  Prints the encoding parameters used
*
************************************************************************/

void print_config (frame_info * frame, int *psy, char *inPath,
		   char *outPath)
{
  frame_header *header = frame->header;

  if (glopts.verbosity == 0)
    return;

  fprintf (stderr, "--------------------------------------------\n");
  fprintf (stderr, "Input File : '%s'   %.1f kHz\n",
	   (strcmp (inPath, "-") ? inPath : "stdin"),
	   s_freq[header->version][header->sampling_frequency]);
  fprintf (stderr, "Output File: '%s'\n",
	   (strcmp (outPath, "-") ? outPath : "stdout"));
  fprintf (stderr, "%d kbps ", bitrate[header->version][header->bitrate_index]);
  fprintf (stderr, "%s ", version_names[header->version]);
  if (header->mode != MPG_MD_JOINT_STEREO)
    fprintf (stderr, "Layer II %s Psycho model=%d  (Mode_Extension=%d)\n",
	     mode_names[header->mode], *psy, header->mode_ext);
  else
    fprintf (stderr, "Layer II %s Psy model %d \n", mode_names[header->mode],
	     *psy);

  fprintf (stderr, "[De-emph:%s\tCopyright:%s\tOriginal:%s\tCRC:%s]\n",
	   ((header->emphasis) ? "On" : "Off"),
	   ((header->copyright) ? "Yes" : "No"),
	   ((header->original) ? "Yes" : "No"),
	   ((header->error_protection) ? "On" : "Off"));

  fprintf (stderr, "[Padding:%s\tByte-swap:%s\tChanswap:%s\tDAB:%s]\n",
	   ((glopts.usepadbit) ? "Normal" : "Off"),
	   ((glopts.byteswap) ? "On" : "Off"),
	   ((glopts.channelswap) ? "On" : "Off"),
	   ((glopts.dab) ? "On" : "Off"));

  if (glopts.vbr == TRUE)
    fprintf (stderr, "VBR Enabled. Using MNR boost of %f\n", glopts.vbrlevel);
  fprintf(stderr,"ATH adjustment %f\n",glopts.athlevel);

  fprintf (stderr, "--------------------------------------------\n");
}


/************************************************************************
*
* usage
*
* PURPOSE:  Writes command line syntax to the file specified by #stderr#
*
************************************************************************/

void usage (void)
{				/* print syntax & exit */
  /* FIXME: maybe have an option to display better definitions of help codes, and
     long equivalents of the flags */
  fprintf (stdout, "\ntooLAME version %s (http://toolame.sourceforge.net)\n",
	   toolameversion);
  fprintf (stdout, "MPEG Audio Layer II encoder\n\n");
  fprintf (stdout, "usage: \n");
  fprintf (stdout, "\t%s [options] <input> <output>\n\n", programName);

  fprintf (stdout, "Options:\n");
  fprintf (stdout, "Input\n");
  fprintf (stdout, "\t-s sfrq  input smpl rate in kHz   (dflt %4.1f)\n",
	   DFLT_SFQ);
  fprintf (stdout, "\t-a       downmix from stereo to mono\n");
  fprintf (stdout, "\t-x       force byte-swapping of input\n");
  fprintf (stdout, "\t-g       swap channels of input file\n");
  fprintf (stdout, "Output\n");
  fprintf (stdout, "\t-m mode  channel mode : s/d/j/m   (dflt %4c)\n",
	   DFLT_MOD);
  fprintf (stdout, "\t-p psy   psychoacoustic model 0/1/2/3 (dflt %4u)\n",
	   DFLT_PSY);
  fprintf (stdout, "\t-b br    total bitrate in kbps    (dflt 192)\n");
  fprintf (stdout, "\t-v lev   vbr mode\n");
  fprintf (stdout, "\t-l lev   ATH level (dflt 0)\n");
  fprintf (stdout, "\t-W       output BWF-compatible wav file, rather than raw MPEG frames\n");
  fprintf (stdout, "Operation\n");
  // fprintf (stdout, "\t-f       fast mode (turns off psy model)\n");
  // deprecate the -f switch. use "-p 0" instead.
  fprintf (stdout,
	   "\t-q num   quick mode. only calculate psy model every num frames\n");
  fprintf (stdout, "Misc\n");
  fprintf (stdout, "\t-d emp   de-emphasis n/5/c        (dflt %4c)\n",
	   DFLT_EMP);
  fprintf (stdout, "\t-c       mark as copyright\n");
  fprintf (stdout, "\t-o       mark as original\n");
  fprintf (stdout, "\t-e       add error protection\n");
  fprintf (stdout, "\t-r       force padding bit/frame off\n");
  fprintf (stdout, "\t-D len   add DAB extensions of length [len]\n");
  fprintf (stdout, "\t-t       talkativity 0=no messages (dflt 2)");
  fprintf (stdout, "Files\n");
  fprintf (stdout,
	   "\tinput    input sound file. (WAV,AIFF,PCM or use '/dev/stdin')\n");
  fprintf (stdout, "\toutput   output bit stream of encoded audio\n");
  fprintf (stdout,
	   "\n\tAllowable bitrates for 16, 22.05 and 24kHz sample input\n");
  fprintf (stdout,
	   "\t8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160\n");
  fprintf (stdout,
	   "\n\tAllowable bitrates for 32, 44.1 and 48kHz sample input\n");
  fprintf (stdout,
	   "\t32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384\n");
  exit (1);
}

/*********************************************
 * void short_usage(void)
 ********************************************/
void short_usage (void)
{
  /* print a bit of info about the program */
  fprintf (stderr, "tooLAME version %s\n (http://toolame.sourceforge.net)\n",
	   toolameversion);
  fprintf (stderr, "MPEG Audio Layer II encoder\n\n");
  fprintf (stderr, "USAGE: %s [options] <infile> [outfile]\n\n", programName);
  fprintf (stderr, "Try \"%s -h\" for more information.\n", programName);
  exit (0);
}

/*********************************************
 * void proginfo(void)
 ********************************************/
void proginfo (void)
{
  /* print a bit of info about the program */
  fprintf (stderr,
	   "\ntooLAME version 0.2g (http://toolame.sourceforge.net)\n");
  fprintf (stderr, "MPEG Audio Layer II encoder\n\n");
}

/************************************************************************
*
* parse_args
*
* PURPOSE:  Sets encoding parameters to the specifications of the
* command line.  Default settings are used for parameters
* not specified in the command line.
*
* SEMANTICS:  The command line is parsed according to the following
* syntax:
*
* -m  is followed by the mode
* -p  is followed by the psychoacoustic model number
* -s  is followed by the sampling rate
* -b  is followed by the total bitrate, irrespective of the mode
* -d  is followed by the emphasis flag
* -c  is followed by the copyright/no_copyright flag
* -o  is followed by the original/not_original flag
* -e  is followed by the error_protection on/off flag
* -f  turns off psy model (fast mode)
* -q <i>  only calculate psy model every ith frame
* -a  downmix from stereo to mono 
* -r  turn off padding bits in frames.
* -x  force byte swapping of input
* -g  swap the channels on an input file
* -t  talkativity. how verbose should the program be. 0 = no messages. 
* -W  output a bwf wav file, rather than raw MPEG frames.
*
* If the input file is in AIFF format, the sampling frequency is read
* from the AIFF header.
*
* The input and output filenames are read into #inpath# and #outpath#.
*
************************************************************************/

void parse_args (int argc, char **argv, frame_info * frame, int *psy,
		 unsigned long *num_samples, char inPath[MAX_NAME_SIZE],
		 char outPath[MAX_NAME_SIZE])
{
  FLOAT srate;
  int brate;
  frame_header *header = frame->header;
  int err = 0, i = 0;
  long samplerate;

  /* preset defaults */
  inPath[0] = '\0';
  outPath[0] = '\0';
  header->lay = DFLT_LAY;
  switch (DFLT_MOD) {
  case 's':
    header->mode = MPG_MD_STEREO;
    header->mode_ext = 0;
    break;
  case 'd':
    header->mode = MPG_MD_DUAL_CHANNEL;
    header->mode_ext = 0;
    break;
    /* in j-stereo mode, no default header->mode_ext was defined, gave error..
       now  default = 2   added by MFC 14 Dec 1999.  */
  case 'j':
    header->mode = MPG_MD_JOINT_STEREO;
    header->mode_ext = 2;
    break;
  case 'm':
    header->mode = MPG_MD_MONO;
    header->mode_ext = 0;
    break;
  default:
    fprintf (stderr, "%s: Bad mode dflt %c\n", programName, DFLT_MOD);
    abort ();
  }
  *psy = DFLT_PSY;
  if ((header->sampling_frequency =
       SmpFrqIndex ((long) (1000 * DFLT_SFQ), &header->version)) < 0) {
    fprintf (stderr, "%s: bad sfrq default %.2f\n", programName, DFLT_SFQ);
    abort ();
  }
  header->bitrate_index = 14;
  brate = 0;
  switch (DFLT_EMP) {
  case 'n':
    header->emphasis = 0;
    break;
  case '5':
    header->emphasis = 1;
    break;
  case 'c':
    header->emphasis = 3;
    break;
  default:
    fprintf (stderr, "%s: Bad emph dflt %c\n", programName, DFLT_EMP);
    abort ();
  }
  header->copyright = 0;
  header->original = 0;
  header->error_protection = FALSE;
  header->dab_extension = 0;

  /* process args */
  while (++i < argc && err == 0) {
    char c, *token, *arg, *nextArg;
    int argUsed;

    token = argv[i];
    if (*token++ == '-') {
      if (i + 1 < argc)
	nextArg = argv[i + 1];
      else
	nextArg = "";
      argUsed = 0;
      if (!*token) {
	/* The user wants to use stdin and/or stdout. */
	if (inPath[0] == '\0')
	  strncpy (inPath, argv[i], MAX_NAME_SIZE);
	else if (outPath[0] == '\0')
	  strncpy (outPath, argv[i], MAX_NAME_SIZE);
      }
      while ((c = *token++)) {
	if (*token /* NumericQ(token) */ )
	  arg = token;
	else
	  arg = nextArg;
	switch (c) {
	case 'm':
	  argUsed = 1;
	  if (*arg == 's') {
	    header->mode = MPG_MD_STEREO;
	    header->mode_ext = 0;
	  } else if (*arg == 'd') {
	    header->mode = MPG_MD_DUAL_CHANNEL;
	    header->mode_ext = 0;
	  } else if (*arg == 'j') {
	    header->mode = MPG_MD_JOINT_STEREO;
	  } else if (*arg == 'm') {
	    header->mode = MPG_MD_MONO;
	    header->mode_ext = 0;
	  } else {
	    fprintf (stderr, "%s: -m mode must be s/d/j/m not %s\n",
		     programName, arg);
	    err = 1;
	  }
	  break;
	case 'p':
	  *psy = atoi (arg);
	  argUsed = 1;
	  break;

	case 's':
	  argUsed = 1;
	  srate = atof (arg);
	  /* samplerate = rint( 1000.0 * srate ); $A  */
	  samplerate = (long) ((1000.0 * srate) + 0.5);
	  if ((header->sampling_frequency =
	       SmpFrqIndex ((long) samplerate, &header->version)) < 0)
	    err = 1;
	  break;

	case 'b':
	  argUsed = 1;
	  brate = atoi (arg);
	  break;
	case 'd':
	  argUsed = 1;
	  if (*arg == 'n')
	    header->emphasis = 0;
	  else if (*arg == '5')
	    header->emphasis = 1;
	  else if (*arg == 'c')
	    header->emphasis = 3;
	  else {
	    fprintf (stderr, "%s: -d emp must be n/5/c not %s\n", programName,
		     arg);
	    err = 1;
	  }
	  break;
	case 'D':
	  argUsed = 1;
	  header->dab_length = atoi (arg);
	  header->error_protection = TRUE;
	  header->dab_extension = 2;
	  glopts.dab = TRUE;
	  break;
	case 'c':
	  header->copyright = 1;
	  break;
	case 'o':
	  header->original = 1;
	  break;
	case 'e':
	  header->error_protection = TRUE;
	  break;
	case 'f':
	  *psy = 0;
	  /* this switch is deprecated? FIXME get rid of glopts.usepsy
	     instead us psymodel 0, i.e. "-p 0" */
	  glopts.usepsy = FALSE;
	  break;
	case 'r':
	  glopts.usepadbit = FALSE;
	  header->padding = 0;
	  break;
	case 'q':
	  argUsed = 1;
	  glopts.quickmode = TRUE;
	  glopts.usepsy = TRUE;
	  glopts.quickcount = atoi (arg);
	  if (glopts.quickcount == 0) {
	    /* just don't use psy model */
	    glopts.usepsy = FALSE;
	    glopts.quickcount = FALSE;
	  }
	  break;
	case 'a':
	  glopts.downmix = TRUE;
	  header->mode = MPG_MD_MONO;
	  header->mode_ext = 0;
	  break;
	case 'x':
	  glopts.byteswap = TRUE;
	  break;
	case 'v':
	  argUsed = 1;
	  glopts.vbr = TRUE;
	  glopts.vbrlevel = atof (arg);
	  glopts.usepadbit = FALSE;	/* don't use padding for VBR */
	  header->padding = 0;
	  /* MFC Feb 2003: in VBR mode, joint stereo doesn't make
	     any sense at the moment, as there are no noisy subbands 
	     according to bits_for_nonoise in vbr mode */
	  header->mode = MPG_MD_STEREO; /* force stereo mode */
	  header->mode_ext = 0;
	  break;
	case 'l':
	  argUsed = 1;
	  glopts.athlevel = atof(arg);
	  break;
	case 'h':
	  usage ();
	  break;
	case 'g':
	  glopts.channelswap = TRUE;
	  break;
	case 't':
	  argUsed = 1;
	  glopts.verbosity = atoi (arg);
	  break;
	case 'W':
	  glopts.outputformat = 1;
	  break;
	default:
	  fprintf (stderr, "%s: unrec option %c\n", programName, c);
	  err = 1;
	  break;
	}
	if (argUsed) {
	  if (arg == token)
	    token = "";		/* no more from token */
	  else
	    ++i;		/* skip arg we used */
	  arg = "";
	  argUsed = 0;
	}
      }
    } else {
      if (inPath[0] == '\0')
	strcpy (inPath, argv[i]);
      else if (outPath[0] == '\0')
	strcpy (outPath, argv[i]);
      else {
	fprintf (stderr, "%s: excess arg %s\n", programName, argv[i]);
	err = 1;
      }
    }
  }

  if (header->dab_extension) {
    /* in 48 kHz */
    /* if the bit rate per channel is less then 56 kbit/s, we have 2 scf-crc */
    /* else we have 4 scf-crc */
    /* in 24 kHz, we have 4 scf-crc, see main loop */
    if (brate / (header->mode == MPG_MD_MONO ? 1 : 2) >= 56)
      header->dab_extension = 4;
  }


  if (err || inPath[0] == '\0')
    usage ();			/* If no infile defined, or err has occured, then call usage() */

  if (outPath[0] == '\0') {
    /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */
    if(glopts.outputformat==0) {
      new_ext (inPath, DFLT_EXT, outPath);
    }
    else {
      new_ext (inPath, DFLT_WAV_EXT, outPath);
    }
  }

  if (!strcmp (inPath, "-")) {
    musicin = stdin;		/* read from stdin */
    *num_samples = MAX_U_32_NUM;
  } else {
    if ((musicin = fopen (inPath, "rb")) == NULL) {
      fprintf (stderr, "Could not find \"%s\".\n", inPath);
      exit (1);
    }
    parse_input_file (musicin, inPath, header, num_samples);
  }

  /* check for a valid bitrate */
  if (brate == 0)
    brate = bitrate[header->version][10];

  /* Check to see we have a sane value for the bitrate for this version */
  if ((header->bitrate_index = BitrateIndex (brate, header->version)) < 0)
    err = 1;

  /* All options are hunky dory, open the input audio file and
     return to the main drag */
  open_bit_stream_w (&bs, outPath, BUFFER_SIZE);
  if(glopts.outputformat==1) {
    bwf_open(&bs,header);
  }
}


void smr_dump(double smr[2][SBLIMIT], int nch) {
  int ch, sb;

  fprintf(stdout,"SMR:");
  for (ch = 0;ch<nch; ch++) {
    if (ch==1)
      fprintf(stdout,"    ");
    for (sb=0;sb<SBLIMIT;sb++)
      fprintf(stdout,"%3.0f ",smr[ch][sb]);
    fprintf(stdout,"\n");
  }
}
