reindent toolame.c

1 files changed, 816 insertions, 815 deletions

diff --git a/toolame.c b/toolame.c
index d8f581a..b1c7f20 100644
--- a/toolame.c
+++ b/toolame.c
@@ -30,545 +30,545 @@ 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.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;
 }
 
 /************************************************************************
-*
-* 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)
-*
-************************************************************************/
+ *
+ * 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;
-    }
+    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;
 
-    {
-      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] );
-    }
+    /* 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]);
-	  }
-    }
+        {
+            /* 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);
-    }
+        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);
-    }
+        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);
-     
-      
-
-
-    }
+        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);
+        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);
+        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);
+        /* 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;
     }
 
-    sentBits += frameBits;
-  }
+    close_bit_stream_w (&bs);
 
-  close_bit_stream_w (&bs);
-
-  if ((glopts.verbosity > 1) && (glopts.vbr == TRUE)) {
-    int i;
+    if ((glopts.verbosity > 1) && (glopts.vbr == TRUE)) {
+        int i;
 #ifdef NEWENCODE
-    extern int vbrstats_new[15];
+        extern int vbrstats_new[15];
 #else
-    extern int vbrstats[15];
+        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++)
+        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]);
+            fprintf (stdout,"%4i ",vbrstats_new[i]);
 #else
-      fprintf (stdout, "%4i ", vbrstats[i]);
+        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);
+        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
-*
-************************************************************************/
+ *
+ * print_config
+ *
+ * PURPOSE:  Prints the encoding parameters used
+ *
+ ************************************************************************/
 
 void print_config (frame_info * frame, int *psy, char *inPath,
-		   char *outPath)
+        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");
+    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#
-*
-************************************************************************/
+ *
+ * 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, "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);
+    /* 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, "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);
 }
 
 /*********************************************
@@ -576,13 +576,13 @@ void 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);
+    /* 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);
 }
 
 /*********************************************
@@ -590,336 +590,337 @@ void short_usage (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");
+    /* 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. 
-*
-* 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#.
-*
-************************************************************************/
+ *
+ * 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. 
+ *
+ * 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])
+        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;
-	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;
-      }
+    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 ();
     }
-  }
-
-  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 */
-    new_ext (inPath, DFLT_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);
+    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;
+                    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;
     }
-    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;
+    if (err || inPath[0] == '\0')
+        usage ();			/* If no infile defined, or err has occured, then call usage() */
 
-  /* 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 (outPath[0] == '\0') {
+        /* replace old extension with new one, 1992-08-19, 1995-06-12 shn */
+        new_ext (inPath, DFLT_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);
 }
 
 
 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");
-  }
+    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");
+    }
 }
+