Version 0.1.1 from FTP

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 319 insertions, 0 deletions

diff --git a/codecs/mp3/src/cupL1.c b/codecs/mp3/src/cupL1.c
new file mode 100755
index 000000000..84b46d24d
--- /dev/null
+++ b/codecs/mp3/src/cupL1.c
@@ -0,0 +1,319 @@
+/*____________________________________________________________________________
+	
+	FreeAmp - The Free MP3 Player
+
+        MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
+        Corp.  http://www.xingtech.com
+
+	Portions Copyright (C) 1998-1999 EMusic.com
+
+	This program is free software; you can redistribute it and/or modify
+	it under the terms of the GNU General Public License as published by
+	the Free Software Foundation; either version 2 of the License, or
+	(at your option) any later version.
+
+	This program 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 General Public License for more details.
+
+	You should have received a copy of the GNU General Public License
+	along with this program; if not, write to the Free Software
+	Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  cupL1.c  ***************************************************
+
+MPEG audio decoder Layer I mpeg1 and mpeg2
+
+include to clup.c
+
+
+******************************************************************/
+/*======================================================================*/
+
+
+/* Read Only */
+static int bat_bit_masterL1[] =
+{
+   0, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16};
+
+/*======================================================================*/
+static void unpack_baL1(MPEG *m)
+{
+   int j;
+   int nstereo;
+
+   m->cup.bit_skip = 0;
+   nstereo = m->cup.stereo_sb;
+
+   for (j = 0; j < m->cup.nbatL1; j++)
+   {
+      mac_load_check(4);
+      m->cup.ballo[j] = m->cup.samp_dispatch[j] = mac_load(4);
+      if (j >= m->cup.nsb_limit)
+	 m->cup.bit_skip += bat_bit_masterL1[m->cup.samp_dispatch[j]];
+      m->cup.c_value[j] = m->cup.look_c_valueL1[m->cup.samp_dispatch[j]];
+      if (--nstereo < 0)
+      {
+	 m->cup.ballo[j + 1] = m->cup.ballo[j];
+	 m->cup.samp_dispatch[j] += 15;	/* flag as joint */
+	 m->cup.samp_dispatch[j + 1] = m->cup.samp_dispatch[j];	/* flag for sf */
+	 m->cup.c_value[j + 1] = m->cup.c_value[j];
+	 j++;
+      }
+   }
+/*-- terminate with bit skip and end --*/
+   m->cup.samp_dispatch[m->cup.nsb_limit] = 31;
+   m->cup.samp_dispatch[j] = 30;
+}
+/*-------------------------------------------------------------------------*/
+static void unpack_sfL1(MPEG *m)	/* unpack scale factor */
+{				/* combine dequant and scale factors */
+   int i;
+
+   for (i = 0; i < m->cup.nbatL1; i++)
+   {
+      if (m->cup.ballo[i])
+      {
+	 mac_load_check(6);
+	 m->cup.cs_factorL1[i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
+      }
+   }
+/*-- done --*/
+}
+/*-------------------------------------------------------------------------*/
+#define UNPACKL1_N(n) s[k]     =  m->cup.cs_factorL1[k]*(load(m,n)-((1 << (n-1)) -1));  \
+    goto dispatch;
+#define UNPACKL1J_N(n) tmp        =  (load(m,n)-((1 << (n-1)) -1));                 \
+    s[k]       =  m->cup.cs_factorL1[k]*tmp;                        \
+    s[k+1]     =  m->cup.cs_factorL1[k+1]*tmp;                      \
+    k++;                                                     \
+    goto dispatch;
+/*-------------------------------------------------------------------------*/
+static void unpack_sampL1(MPEG *m)	/* unpack samples */
+{
+   int j, k;
+   float *s;
+   long tmp;
+
+   s = m->cup.sample;
+   for (j = 0; j < 12; j++)
+   {
+      k = -1;
+    dispatch:switch (m->cup.samp_dispatch[++k])
+      {
+	 case 0:
+	    s[k] = 0.0F;
+	    goto dispatch;
+	 case 1:
+	    UNPACKL1_N(2)	/*  3 levels */
+	 case 2:
+	    UNPACKL1_N(3)	/*  7 levels */
+	 case 3:
+	    UNPACKL1_N(4)	/* 15 levels */
+	 case 4:
+	    UNPACKL1_N(5)	/* 31 levels */
+	 case 5:
+	    UNPACKL1_N(6)	/* 63 levels */
+	 case 6:
+	    UNPACKL1_N(7)	/* 127 levels */
+	 case 7:
+	    UNPACKL1_N(8)	/* 255 levels */
+	 case 8:
+	    UNPACKL1_N(9)	/* 511 levels */
+	 case 9:
+	    UNPACKL1_N(10)	/* 1023 levels */
+	 case 10:
+	    UNPACKL1_N(11)	/* 2047 levels */
+	 case 11:
+	    UNPACKL1_N(12)	/* 4095 levels */
+	 case 12:
+	    UNPACKL1_N(13)	/* 8191 levels */
+	 case 13:
+	    UNPACKL1_N(14)	/* 16383 levels */
+	 case 14:
+	    UNPACKL1_N(15)	/* 32767 levels */
+/* -- joint ---- */
+	 case 15 + 0:
+	    s[k + 1] = s[k] = 0.0F;
+	    k++;		/* skip right chan dispatch */
+	    goto dispatch;
+/* -- joint ---- */
+	 case 15 + 1:
+	    UNPACKL1J_N(2)	/*  3 levels */
+	 case 15 + 2:
+	    UNPACKL1J_N(3)	/*  7 levels */
+	 case 15 + 3:
+	    UNPACKL1J_N(4)	/* 15 levels */
+	 case 15 + 4:
+	    UNPACKL1J_N(5)	/* 31 levels */
+	 case 15 + 5:
+	    UNPACKL1J_N(6)	/* 63 levels */
+	 case 15 + 6:
+	    UNPACKL1J_N(7)	/* 127 levels */
+	 case 15 + 7:
+	    UNPACKL1J_N(8)	/* 255 levels */
+	 case 15 + 8:
+	    UNPACKL1J_N(9)	/* 511 levels */
+	 case 15 + 9:
+	    UNPACKL1J_N(10)	/* 1023 levels */
+	 case 15 + 10:
+	    UNPACKL1J_N(11)	/* 2047 levels */
+	 case 15 + 11:
+	    UNPACKL1J_N(12)	/* 4095 levels */
+	 case 15 + 12:
+	    UNPACKL1J_N(13)	/* 8191 levels */
+	 case 15 + 13:
+	    UNPACKL1J_N(14)	/* 16383 levels */
+	 case 15 + 14:
+	    UNPACKL1J_N(15)	/* 32767 levels */
+
+/* -- end of dispatch -- */
+	 case 31:
+	    skip(m,m->cup.bit_skip);
+	 case 30:
+	    s += 64;
+      }				/* end switch */
+   }				/* end j loop */
+
+/*-- done --*/
+}
+/*-------------------------------------------------------------------*/
+IN_OUT L1audio_decode(void *mv, unsigned char *bs, signed short *pcm)
+{
+   MPEG *m = mv;
+   int sync, prot;
+   IN_OUT in_out;
+
+   load_init(m, bs);		/* initialize bit getter */
+/* test sync */
+   in_out.in_bytes = 0;		/* assume fail */
+   in_out.out_bytes = 0;
+   sync = load(m,12);
+   if (sync != 0xFFF)
+      return in_out;		/* sync fail */
+
+
+   load(m,3);			/* skip id and option (checked by init) */
+   prot = load(m,1);		/* load prot bit */
+   load(m,6);			/* skip to pad */
+   m->cup.pad = (load(m,1)) << 2;
+   load(m,1);			/* skip to mode */
+   m->cup.stereo_sb = look_joint[load(m,4)];
+   if (prot)
+      load(m,4);			/* skip to data */
+   else
+      load(m,20);			/* skip crc */
+
+   unpack_baL1(m);		/* unpack bit allocation */
+   unpack_sfL1(m);		/* unpack scale factor */
+   unpack_sampL1(m);		/* unpack samples */
+
+   m->cup.sbt(m, m->cup.sample, pcm, 12);
+/*-----------*/
+   in_out.in_bytes = m->cup.framebytes + m->cup.pad;
+   in_out.out_bytes = m->cup.outbytes;
+
+   return in_out;
+}
+/*-------------------------------------------------------------------------*/
+int L1audio_decode_init(MPEG *m, MPEG_HEAD * h, int framebytes_arg,
+		   int reduction_code, int transform_code, int convert_code,
+			int freq_limit)
+{
+   int i, k;
+   long samprate;
+   int limit;
+   long step;
+   int bit_code;
+
+/*--- sf init done by layer II init ---*/
+   if (m->cup.first_pass_L1)
+   {
+      for (step = 4, i = 1; i < 16; i++, step <<= 1)
+	 m->cup.look_c_valueL1[i] = (float) (2.0 / (step - 1));
+      m->cup.first_pass_L1 = 0;
+   }
+
+   transform_code = transform_code;	/* not used, asm compatability */
+
+   bit_code = 0;
+   if (convert_code & 8)
+      bit_code = 1;
+   convert_code = convert_code & 3;	/* higher bits used by dec8 freq cvt */
+   if (reduction_code < 0)
+      reduction_code = 0;
+   if (reduction_code > 2)
+      reduction_code = 2;
+   if (freq_limit < 1000)
+      freq_limit = 1000;
+
+
+   m->cup.framebytes = framebytes_arg;
+/* check if code handles */
+   if (h->option != 3)
+      return 0;			/* layer I only */
+
+   m->cup.nbatL1 = 32;
+   m->cup.max_sb = m->cup.nbatL1;
+/*----- compute nsb_limit --------*/
+   samprate = sr_table[4 * h->id + h->sr_index];
+   m->cup.nsb_limit = (freq_limit * 64L + samprate / 2) / samprate;
+/*- caller limit -*/
+/*---- limit = 0.94*(32>>reduction_code);  ----*/
+   limit = (32 >> reduction_code);
+   if (limit > 8)
+      limit--;
+   if (m->cup.nsb_limit > limit)
+      m->cup.nsb_limit = limit;
+   if (m->cup.nsb_limit > m->cup.max_sb)
+      m->cup.nsb_limit = m->cup.max_sb;
+
+   m->cup.outvalues = 384 >> reduction_code;
+   if (h->mode != 3)
+   {				/* adjust for 2 channel modes */
+      m->cup.nbatL1 *= 2;
+      m->cup.max_sb *= 2;
+      m->cup.nsb_limit *= 2;
+   }
+
+/* set sbt function */
+   k = 1 + convert_code;
+   if (h->mode == 3)
+   {
+      k = 0;
+   }
+   m->cup.sbt = sbt_table[bit_code][reduction_code][k];
+   m->cup.outvalues *= out_chans[k];
+
+   if (bit_code)
+      m->cup.outbytes = m->cup.outvalues;
+   else
+      m->cup.outbytes = sizeof(short) * m->cup.outvalues;
+
+   m->cup.decinfo.channels = out_chans[k];
+   m->cup.decinfo.outvalues = m->cup.outvalues;
+   m->cup.decinfo.samprate = samprate >> reduction_code;
+   if (bit_code)
+      m->cup.decinfo.bits = 8;
+   else
+      m->cup.decinfo.bits = sizeof(short) * 8;
+
+   m->cup.decinfo.framebytes = m->cup.framebytes;
+   m->cup.decinfo.type = 0;
+
+
+/* clear sample buffer, unused sub bands must be 0 */
+   for (i = 0; i < 768; i++)
+      m->cup.sample[i] = 0.0F;
+
+
+/* init sub-band transform */
+   sbt_init();
+
+   return 1;
+}
+/*---------------------------------------------------------*/